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[PATCH] ARM: OMAP: Hack add some OMAP3 McBSP muxing
by Jarkko Nikula 09 Oct '08

09 Oct '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- arch/arm/mach-omap2/mux.c | 52 +++++++++++++++++++++++++++++++++ arch/arm/plat-omap/include/mach/mux.h | 29 ++++++++++++++++++ 2 files changed, 81 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-omap2/mux.c b/arch/arm/mach-omap2/mux.c index dacb41f..1dff09e 100644 --- a/arch/arm/mach-omap2/mux.c +++ b/arch/arm/mach-omap2/mux.c @@ -261,6 +261,58 @@ static struct pin_config __initdata_or_module omap34xx_pins[] = { * mux-mode | [active-mode | off-mode] */ +/* Primary McBSP multiplexing */ +MUX_CFG_34XX("Y21_3430_MCBSP1_CLKR", 0x18c, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AA21_3430_MCBSP1_FSR", 0x18e, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("Y21_3430_MCBSP1_DX", 0x190, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("U21_3430_MCBSP1_DR", 0x192, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("T21_3430_MCBSP1_CLKS", 0x194, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("K26_3430_MCBSP1_FSX", 0x196, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("W21_3430_MCBSP1_CLKX", 0x198, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("P21_3430_MCBSP2_FSX", 0x13c, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("N21_3430_MCBSP2_CLKX", 0x13e, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("R21_3430_MCBSP2_DR", 0x140, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("M21_3430_MCBSP2_DX", 0x142, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("AF6_3430_MCBSP3_DX", 0x16c, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AE6_3430_MCBSP3_DR", 0x16e, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AF5_3430_MCBSP3_CLKX", 0x170, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AE5_3430_MCBSP3_FSX", 0x172, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("AE1_3430_MCBSP4_CLKX", 0x184, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AD1_3430_MCBSP4_DR", 0x186, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AD2_3430_MCBSP4_DX", 0x188, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AC1_3430_MCBSP4_FSX", 0x18a, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("AF10_3430_MCBSP5_CLKX", 0x5d8, + OMAP34XX_MUX_MODE1) +MUX_CFG_34XX("AE11_3430_MCBSP5_DR", 0x5e4, + OMAP34XX_MUX_MODE1) +MUX_CFG_34XX("AH9_3430_MCBSP5_FSX", 0x5e6, + OMAP34XX_MUX_MODE1) +MUX_CFG_34XX("AF13_3430_MCBSP5_DX", 0x5e8, + OMAP34XX_MUX_MODE1) + /* 34xx I2C */ MUX_CFG_34XX("K21_34XX_I2C1_SCL", 0x1ba, OMAP34XX_MUX_MODE0 | OMAP34XX_PIN_INPUT_PULLUP) diff --git a/arch/arm/plat-omap/include/mach/mux.h b/arch/arm/plat-omap/include/mach/mux.h index f4362b8..c0cfa73 100644 --- a/arch/arm/plat-omap/include/mach/mux.h +++ b/arch/arm/plat-omap/include/mach/mux.h @@ -677,6 +677,35 @@ enum omap24xx_index { }; enum omap34xx_index { + /* Primary McBSP multiplexing */ + Y21_3430_MCBSP1_CLKR, + AA21_3430_MCBSP1_FSR, + Y21_3430_MCBSP1_DX, + U21_3430_MCBSP1_DR, + T21_3430_MCBSP1_CLKS, + K26_3430_MCBSP1_FSX, + W21_3430_MCBSP1_CLKX, + + P21_3430_MCBSP2_FSX, + N21_3430_MCBSP2_CLKX, + R21_3430_MCBSP2_DR, + M21_3430_MCBSP2_DX, + + AF6_3430_MCBSP3_DX, + AE6_3430_MCBSP3_DR, + AF5_3430_MCBSP3_CLKX, + AE5_3430_MCBSP3_FSX, + + AE1_3430_MCBSP4_CLKX, + AD1_3430_MCBSP4_DR, + AD2_3430_MCBSP4_DX, + AC1_3430_MCBSP4_FSX, + + AF10_3430_MCBSP5_CLKX, + AE11_3430_MCBSP5_DR, + AH9_3430_MCBSP5_FSX, + AF13_3430_MCBSP5_DX, + /* 34xx I2C */ K21_34XX_I2C1_SCL, J21_34XX_I2C1_SDA, -- 1.5.6.5 --Multipart=_Thu__2_Apr_2009_19_56_53_+0300_OCGZiXCSMdXq4.P8--

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[alsa-devel] programming AC'97
by joan vicient 09 Oct '08

09 Oct '08

Hi, I'm an electronic engineering student and I'm having some troubles with my project wich I have to develop to become an engineer. The project itself is based on my laptop (ICH6 with the AC97 soundcard). I'm trying to develop a sound generator with the AC97. I've been working hard but there is something wrong and I'm not sure what is it. I think is the "Buffer Descriptor List Base Addres" or maybe the codec initialization, not sure... I am looking at the "AC'97 program model" and alsa-driver to understand how does the driver configure the soundcard so I can do the same but I don't understand the source code at all... So... somebody out there with some skills can help me? Thank you very much!!

1 0

[alsa-devel] [PATCH 0/9] ASoC Blackfin supporting
by Bryan Wu 09 Oct '08

09 Oct '08

Hi Mark, After your last review, Cliff and me updated this patch series: - fix coding style issue - use latest ASoC API - split the whole patch into this 9 patches in a patchset Thanks a lot -Bryan

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Re: [alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by The Source 09 Oct '08

09 Oct '08

So, should anyone start to port OSS driver or should we wait for specs?

1 0

[alsa-devel] dv6627el & audio problems
by yelo_3 09 Oct '08

09 Oct '08

Hello, I want to share my problems with you. I'm running ubuntu intrepid, with these packages versions: linux 2.6.27-6-generic alsa 1.0.17.dfsg-2ubuntu1 I have an hp dv6627el with this audio card model: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) and this is my alsa page: http://www.alsa-project.org/db/?f=457f3992bcf0b34b51772a730071ce18deea75b2 I don't use any additional module option other that the one I will specify. I have 2 problems: 1) mute led does not work. if I use the option power_save (for isntance=3) the mute led lights up (red) after 3 seconds of audio inactivity (and after having closed the audio software), even if the volume is not muted. If I keep the audio software open, the led remains "off" (blue). if instead I don't use the option power_save (so power_save=0) the led never lights up. 2) when using headphones, the speakers are normally muted, but if I hit the mute button twice (or use the volume control applet to mute and unmute), the speakers are activated. Thank you very much for you help. Nicolò

1 0

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with debugfs filesystem
by Troy Kisky 08 Oct '08

08 Oct '08

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> Mark, I didn't base it on dapm's pop_time directory because that hasn't made it to my testing kernel yet. I cannot test directly from the alsa git tree. Thanks for the feedback. diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f616b0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -516,6 +516,9 @@ struct snd_soc_device { struct delayed_work delayed_work; struct work_struct deferred_resume_work; void *codec_data; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_root; +#endif }; /* runtime channel data */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..156d11a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -962,10 +962,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +1000,112 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_DEBUG_FS +static int code_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t code_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(devdata, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t code_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = code_reg_open_file, + .read = code_reg_read_file, + .write = code_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *debugfs_codec_reg; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + debugfs_codec_reg = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!debugfs_codec_reg) + goto exit2; + + socdev->debugfs_root = root; + return; +exit2: + debugfs_remove(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove_recursive(socdev->debugfs_root); + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1217,6 +1319,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_codec_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1240,6 +1343,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #endif mutex_lock(&codec->mutex); + soc_cleanup_codec_debugfs(socdev); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; -- 1.5.4.3

2 1

[alsa-devel] [PATCH] ASoC: Check for machine type in GTA01 machine driver
by Mark Brown 08 Oct '08

08 Oct '08

Since there are now multiple OpenMoko platforms it is more important to check that the machine driver is running on the correct system. This was orgininally generated as part of the initial GTA02 machine port. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/neo1973_wm8753.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c602674..87ddfef 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -24,6 +24,7 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include <asm/mach-types.h> #include <asm/hardware/scoop.h> #include <mach/regs-clock.h> #include <mach/regs-gpio.h> @@ -677,6 +678,12 @@ static int __init neo1973_init(void) DBG("Entered %s\n", __func__); + if (!machine_is_neo1973_gta01()) { + printk(KERN_INFO + "Only GTA01 hardware supported by ASoC driver\n"); + return -ENODEV; + } + neo1973_snd_device = platform_device_alloc("soc-audio", -1); if (!neo1973_snd_device) return -ENOMEM; -- 1.5.6.5

2 1

Re: [alsa-devel] AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 08 Oct '08

08 Oct '08

Not yet. Would please share what you have done to make it work ? Also when I tried to strace aplay , it keeps looking for a </share/alsa/cards/tlv320aic23.conf> file which is not there in the alsa-lib package. Any idea what goes in that file ? Regards Jesslyn On Tue, Oct 7, 2008 at 7:16 PM, Arun KS <arunks(a)mistralsolutions.com> wrote: > Jesslyn Abdul Salam wrote: > >> I got it working. I2C_BOARD_INFO macro was wrongly intialized. Instead of >> tlv320aic23 i had written tlvaic23. Thanks to Arun for pointing this error >> >> > Can you hear audio playing? > Because i had done some hack to make it working in 5912osk. > I want to know you are not facing it or not. > > Arun > > Jesslyn Abdul Salam >> >> >> On Tue, Oct 7, 2008 at 5:36 PM, Jesslyn Abdul Salam < >> jesslyn.abdulsalam(a)gmail.com> wrote: >> >> >> >>> ---------- Forwarded message ---------- >>> From: Jesslyn Abdul Salam <jesslyn.abdulsalam(a)gmail.com> >>> Date: Tue, Oct 7, 2008 at 5:36 PM >>> Subject: Re: [alsa-devel] AIC23 SOC Driver on OSK5912 >>> To: Arun KS <arunks(a)mistralsolutions.com> >>> >>> >>> Yes, it is there. but it shows as "tlv320aic23b" instead of "tlv320aic" >>> Will that be the reason ???? >>> >>> >>> >>> >>> On Tue, Oct 7, 2008 at 5:32 PM, Arun KS <arunks(a)mistralsolutions.com >>> >wrote: >>> >>> >>> >>>> On Tue, Oct 7, 2008 at 5:24 PM, Mark Brown <broonie(a)sirena.org.uk> >>>> wrote: >>>> >>>> >>>>> On Tue, Oct 07, 2008 at 05:18:55PM +0530, Jesslyn Abdul Salam wrote: >>>>> >>>>> >>>>> >>>>>> I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for >>>>>> the >>>>>> >>>>>> >>>>> past >>>> >>>> >>>>> few days. I got the driver patch from the linux-omap/alsa-devel mailing >>>>>> lists. The driver got compiled along just fine. But when the kernel >>>>>> >>>>>> >>>>> boots, >>>> >>>> >>>>> no sound cards have been listed. >>>>>> >>>>>> >>>>> CCing in Arun KS <arunks(a)mistralsolutions.com> who contributed that >>>>> support. >>>>> >>>>> >>>>> >>>>>> Advanced Linux Sound Architecture Driver Version 1.0.17. >>>>>> ASoC version >>>>>> 0.13.2 >>>>>> AIC23 Audio Codec >>>>>> 0.1 >>>>>> MCLK = 12000000 [12000000], usecount = >>>>>> 0 >>>>>> ALSA device >>>>>> list: >>>>>> No soundcards >>>>>> found. >>>>>> Any one have tested this before ? >>>>>> >>>>>> >>>>> Can you check i2c? >>>> >>>> # cat /sys/bus/i2c/devices/1-001b/name >>>> tlv320aic23 >>>> >>>> >>>> >>>>> _______________________________________________ >>>>> Alsa-devel mailing list >>>>> Alsa-devel(a)alsa-project.org >>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>> >>>>> >>>>> >>>> >>> >>> >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >> >> >> > > > Please do not print this email unless it is absolutely necessary. Spread > environmental awareness. > > > -------------------------------------------------------DISCLAIMER------------------------------------------------------ > The information transmitted herewith is confidential and proprietary > information intended only for use by the individual or entity to which it is > addressed. If the reader of this message is not the intended recipient, you > are hereby notified that any review, retransmission, dissemination, > distribution, copying or other use of, or taking of any action in reliance > upon this information is strictly prohibited. If you have received this > communication in error, please contact the sender and delete the material > from your computer. > > -------------------------------------------------------------------------------------------------------------------------------- > > > >

3 3

[alsa-devel] [PATCH 0/8] ASoC updates
by Mark Brown 08 Oct '08

08 Oct '08

The following changes since commit 691ac84a77b1aebb7e09b4924b4b9c07a0f4f66a: Takashi Iwai (1): Merge branch 'asoc-fixes' into topic/asoc are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai This is all 2.6.28ish material. Arun KS (1): ASoC: Add custom SOC_SINGLE_TLV for tlv320aic23 codec Jarkko Nikula (2): ASoC: tlv320aic3x: Use uniform tlv320aic naming ASoC: Remove unused AUDIO_NAME define from codec drivers Mark Brown (4): ASoC: Add WM8753 SPI support ASoC: Add WM8510 SPI support ASoC: Implement WM8510 bias level control ASoC: Make WM8510 microphone input a DAPM mixer Richard Zhao (1): ALSA: ASoC: add new param mux to dapm_mux_update_power sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320aic23.c | 54 ++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.c | 5 +- sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8510.c | 109 ++++++++++++++++++++++++++++++++++----- sound/soc/codecs/wm8510.h | 1 + sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 72 +++++++++++++++++++++++++- sound/soc/codecs/wm8753.h | 1 + sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/soc-dapm.c | 8 ++-- 15 files changed, 221 insertions(+), 37 deletions(-)

2 9

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: dynamic detection of MIDI interfaces in uaxx-quirk
by pedro.lopez.cabanillas＠gmail.com 08 Oct '08

08 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> The MIDI interfaces have to be detected dynamically for Edirol devices ua-700, ua-25 and ua4-fx. This patch reverses the wrong changes made by my other patch in uaxx-quirk. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6e70ba4..bbd70d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3023,6 +3023,31 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); + if (altsd->bNumEndpoints == 2) { + static const struct snd_usb_midi_endpoint_info ua700_ep = { + .out_cables = 0x0003, + .in_cables = 0x0003 + }; + static const struct snd_usb_audio_quirk ua700_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &ua700_ep + }; + static const struct snd_usb_midi_endpoint_info uaxx_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk uaxx_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &uaxx_ep + }; + if (chip->usb_id == USB_ID(0x0582, 0x002b)) + return snd_usb_create_midi_interface(chip, iface, + &ua700_quirk); + else + return snd_usb_create_midi_interface(chip, iface, + &uaxx_quirk); + } + if (altsd->bNumEndpoints != 1) return -ENXIO; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 85532cf..69689e7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -870,11 +870,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1216,11 +1212,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1375,11 +1367,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 -- 1.5.2.4

2 1

[alsa-devel] ASoC WM8510 Mic PGA problem
by ext-jukka.hynninen＠vaisala.com 08 Oct '08

08 Oct '08

Hi all, I'm facing trouble with programmable mic gain stage and WM8510 ASoC driver. First of all, the DAPM control "Mic PGA Switch" seems to be operating just the opposite way the codec datasheet suggests. I inverted the control with expected result, PGA now almost works. When I start capturing, the "Mic PGA" control is instantly turned off again. If I manually enable the control while the capture is running, it works just fine. Any ideas? Am I missing some DAPM routings? -Jukka

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Re: [alsa-devel] [PATCH] Sigmatel stac92hd71b7 low HP volume
by Takashi Iwai 08 Oct '08

08 Oct '08

[please don't drop Cc to alsa-devel ML] At Mon, 6 Oct 2008 12:48:43 +0200, Klaas De Craemer wrote: > > Hello, > > I'm sure that the audioquality will degrade somewhat because of the additional amplification, > but before, the HP-out was not usable at all. I had to use an external ampli just to use my > headphones, even after turning all volume sliders to 100%. Then again, when removing the HP, > the volume though the internal speakers was way too loud... > > I'm not sure what you mean with the output pin and headphone control, but I have uploaded the > output of the codegraph tool before and after my modification here: > http://users.vtk.be/~s0159120/codec_original.svg > http://users.vtk.be/~s0159120/codec_final.svg > > Below is the output of alsa-info with the patched module > Klaas Thanks. So, if I understand correctly, when the pin 0x0a is connected directly from DAC0 0x10, the output level is too low even if you set the all volumes maximal. In the alsa-info.sh output, I see the following: > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] which means that you decrease the output level intentionally. Is the HP output still too low even if you set to the higher value? Takashi > > ----- > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.51 > !!################################ > > !!Script ran on: Mon Oct 6 12:42:48 CEST 2008 > > > !!Linux Distribution > !!------------------ > > Debian GNU/Linux lenny/sid \n \l > > > !!Kernel Information > !!------------------ > > Kernel release: 2.6.26-5.slh.6-sidux-686 > Operating System: GNU/Linux > Architecture: i686 > Processor: unknown > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.16 > Library version: > Utilities version: 1.0.16 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xf6fdc000 irq 21 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > > > !!Advanced information - PCI Vendor/Device/Susbsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:293e (rev 03) > Subsystem: 1028:024f > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-pcsp: index=-2 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : 0 > id : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > single_cmd : N > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: IDT 92HD71B7X > Address: 0 > Vendor Id: 0x111d76b2 > Subsystem Id: 0x1028024f > Revision Id: 0x100302 > No Modem Function Group found > Default PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 > GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 > Node 0x0a [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x081c: OUT HP Detect > Pin Default 0x0421101f: [Jack] HP Out at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=30, enabled=1 > Connection: 3 > 0x10 0x11 0x17* > Node 0x0b [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x04a11221: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0c [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x0e [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x23a1902e: [Jack] Mic at Sep Left > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0xe > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x23014250: [Jack] Line Out at Sep Left > Conn = 1/8, Color = Green > DefAssociation = 0x5, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x3d 0x3d] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x12 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x1c > Processing caps: benign=0, ncoeff=0 > Node 0x13 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x1d > Processing caps: benign=0, ncoeff=0 > Node 0x14 [Pin Complex] wcaps 0x400100: Mono > Pincap 0x0810: OUT > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x16 > Node 0x15 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x10* 0x11 0x17 > Node 0x16 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x15 > Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x1f 0x1f] [0x1f 0x1f] [0x97 0x97] [0x97 0x97] [0x97 0x97] > Connection: 5 > 0x10 0x11 0x27 0x1a 0x1b > Node 0x18 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x90a000f0: [Fixed] Mic at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x19 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x01 0x01] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x00 0x00] > Connection: 4 > 0x1a* 0x17 0x18 0x19 > Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x80 0x80] > Connection: 4 > 0x1b* 0x17 0x18 0x19 > Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x24 > Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital > Pincap 0x0810010: OUT EAPD > EAPD 0x0: > Pin Default 0x40f000f6: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x6 > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Connection: 2 > 0x24* 0x25 > Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x40f000f7: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x7 > Pin-ctls: 0x00: > Connection: 1 > 0x25 > Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x22 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x24 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x21* 0x1c 0x1d > Node 0x25 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x22* 0x1c 0x1d > Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > Node 0x27 [Pin Complex] wcaps 0x400000: Mono > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Node 0x28 [Volume Knob Widget] wcaps 0x600000: Mono > Volume-Knob: delta=1, steps=127, direct=1, val=127 > Connection: 2 > 0x10* 0x11 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw---- 1 root audio 116, 7 okt 6 12:40 /dev/snd/controlC0 > crw-rw---- 1 root audio 116, 6 okt 6 12:40 /dev/snd/hwC0D0 > crw-rw---- 1 root audio 116, 5 okt 6 12:40 /dev/snd/pcmC0D0c > crw-rw---- 1 root audio 116, 4 okt 6 12:41 /dev/snd/pcmC0D0p > crw-rw---- 1 root audio 116, 3 okt 6 11:41 /dev/snd/seq > crw-rw---- 1 root audio 116, 2 okt 6 11:41 /dev/snd/timer > > > !!Aplay/Arecord output > !!------------ > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 2/2 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xf6fdc000 irq 21' > Mixer name : 'IDT 92HD71B7X' > Components : 'HDA:111d76b2' > Controls : 19 > Simple ctrls : 14 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 127 > Mono: Playback 94 [74%] [-24.75dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Front',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [on] > Front Right: Capture 0 [0%] [0.00dB] [on] > Simple mixer control 'Capture Mux',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 1 [33%] [10.00dB] > Front Right: 1 [33%] [10.00dB] > Simple mixer control 'Capture Mux',1 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'Capture',1 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [off] > Front Right: Capture 0 [0%] [0.00dB] [off] > Simple mixer control 'Analog Loopback',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Analog Loopback',1 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Digital Input Source',0 > Capabilities: enum > Items: 'Analog Inputs' 'Digital Mic 1' > Item0: 'Analog Inputs' > Simple mixer control 'Digital Mic 1',0 > Capabilities: cvolume > Capture channels: Front Left - Front Right > Limits: Capture 0 - 3 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > Simple mixer control 'Input Source',1 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.2 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.3 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > value false > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > index 1 > value false > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > value.0 0 > value.1 0 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > index 1 > value.0 0 > value.1 0 > } > control.9 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.10 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > index 1 > value.0 0 > value.1 0 > } > control.11 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Front Playback Volume' > value.0 127 > value.1 127 > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.13 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Headphone Playback Volume' > value.0 127 > value.1 127 > } > control.14 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.15 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Digital Mic 1 Capture Volume' > value.0 0 > value.1 0 > } > control.16 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 'Analog Inputs' > comment.item.1 'Digital Mic 1' > iface MIXER > name 'Digital Input Source' > value 'Analog Inputs' > } > control.17 { > comment.access 'read write' > comment.type INTEGER > comment.count 1 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Master Playback Volume' > value 94 > } > control.18 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Master Playback Switch' > value true > } > control.19 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > comment.dbmin -5100 > comment.dbmax 0 > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > snd_hda_intel > snd_hwdep > nls_cp437 > vfat > fat > loop > isofs > udf > nvidia > ppdev > lp > parport > acpi_cpufreq > cpufreq_performance > cpufreq_stats > cpufreq_ondemand > cpufreq_powersave > cpufreq_conservative > freq_table > ipv6 > af_packet > hfsplus > nls_utf8 > ntfs > nls_base > fuse > snd_pcm_oss > snd_pcm > snd_page_alloc > snd_mixer_oss > pcmcia > firmware_class > snd_seq_dummy > rtc_cmos > snd_seq_oss > rtc_core > rtc_lib > dcdbas > snd_seq_midi_event > snd_seq > snd_timer > snd_seq_device > serio_raw > psmouse > i2c_i801 > snd > pcspkr > evdev > i2c_core > yenta_socket > rsrc_nonstatic > pcmcia_core > soundcore > video > output > wmi > ac > button > battery > intel_agp > ext3 > jbd > mbcache > dm_mirror > dm_log > dm_snapshot > dm_mod > firewire_sbp2 > sg > sr_mod > cdrom > sd_mod > usbhid > hid > ff_memless > ahci > libata > scsi_mod > dock > ricoh_mmc > sdhci > mmc_core > firewire_ohci > firewire_core > crc_itu_t > ehci_hcd > uhci_hcd > usbcore > e1000e > thermal > processor > fan > > > On Monday 06 October 2008 11:44:33 you wrote: > > At Sun, 5 Oct 2008 22:10:46 +0200, > > > > Klaas De Craemer wrote: > > > Hello, > > > > > > A few days ago, I posted a patch to the linux kernel list, but afterwards > > > I realised that it is more appropriate for the alsa list. So I repost it > > > here. > > > > > > Below is my patch for an issue on Dell E6500 notebooks with the Sigmatel > > > stac92hd71b7 codec. In the default snd-hda-intel configuration, the sound > > > going to the headphone jack goes comes directly from the DAC, resulting > > > in very low volume output. The patch changes the codec's widget > > > connections so that the internal mixer is put in between and sets its > > > input amplifiers to the maximum value. The output to the internal speaker > > > port has been left untouched. > > > I have tested this and it gives a much better output. > > > > Thanks for the patch. > > > > I wonder whether this really improves the quality. > > Basically what you does is to set the volume of the input mixer over > > 0dB, and this is usually bad when really high signals come in. > > > > Just to be sure, is the output pin set to the headphone pin control? > > You can give alsa-info.sh output (run with --no-upload option) for > > further analysis. The script is found in > > http://www.alsa-project.org/alsa-info.sh > > > > > > thanks, > > > > Takashi > >

2 6

[alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by Sebastian Schneider 08 Oct '08

08 Oct '08

What about we start porting the driver on our own? -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

1 0

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with debugfs filesystem
by Troy Kisky 08 Oct '08

08 Oct '08

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f5a6dbb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -516,6 +516,10 @@ struct snd_soc_device { struct delayed_work delayed_work; struct work_struct deferred_resume_work; void *codec_data; +#ifdef CONFIG_SND_SOC_DEBUG_FS + struct dentry *debugfs_root; + struct dentry *debugfs_codec_reg; +#endif }; /* runtime channel data */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..aad7e0b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,14 @@ if SND_SOC config SND_SOC_AC97_BUS bool +config SND_SOC_DEBUG_FS + boolean "changing codec registers through debugfs (DEVELOPMENT)" + depends on DEBUG_FS + help + The ability to directly write codec registers through + /sys/kernel/debug/xx/codec_reg is useful when developing + a new codec. + # All the supported Soc's source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..46827cb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -962,10 +962,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +1000,113 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_SND_SOC_DEBUG_FS +static int code_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t code_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t res; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + + res = soc_codec_reg_show(devdata, buf); + res = simple_read_from_buffer(user_buf, count, ppos, buf, res); + kfree(buf); + return res; +} + +static ssize_t code_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = code_reg_open_file, + .read = code_reg_read_file, + .write = code_reg_write_file, +}; + +static void soc_init_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *debugfs_codec_reg; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + debugfs_codec_reg = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!debugfs_codec_reg) + goto exit2; + + socdev->debugfs_root = root; + socdev->debugfs_codec_reg = debugfs_codec_reg; + return; +exit2: + debugfs_remove(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove(socdev->debugfs_codec_reg); + debugfs_remove(socdev->debugfs_root); + socdev->debugfs_codec_reg = NULL; + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1217,6 +1320,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1254,6 +1358,7 @@ free_card: if (codec->card) snd_card_free(codec->card); device_remove_file(socdev->dev, &dev_attr_codec_reg); + soc_cleanup_debugfs(socdev); mutex_unlock(&codec->mutex); } EXPORT_SYMBOL_GPL(snd_soc_free_pcms); -- 1.5.4.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 07 Oct '08

07 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..1077e9d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,8 +3016,8 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); @@ -3031,20 +3031,20 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &ua700_ep }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { + static const struct snd_usb_midi_endpoint_info uaxx_ep = { .out_cables = 0x0001, .in_cables = 0x0001 }; - static const struct snd_usb_audio_quirk ua25_quirk = { + static const struct snd_usb_audio_quirk uaxx_quirk = { .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep + .data = &uaxx_ep }; if (chip->usb_id == USB_ID(0x0582, 0x002b)) return snd_usb_create_midi_interface(chip, iface, &ua700_quirk); else return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); + &uaxx_quirk); } if (altsd->bNumEndpoints != 1) @@ -3376,9 +3376,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..815bf36 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1197,15 +1197,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1338,6 +1338,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

3 2

Re: [alsa-devel] 2 Au1xxx (Alchemy) AC97 drivers
by Manuel Lauss 07 Oct '08

07 Oct '08

On Tue, Oct 07, 2008 at 08:28:26AM -0500, Kevin Hickey wrote: > There appear to be 2 AC97 drivers for Alchemy in the latest Linux tree. > One is in sound/mips and the other is in sound/soc. Are both going to > be maintained going forward? If not, which one will be? The one in sound/soc is for Au1200/Au1550; I think the sound/mips one only works on Au1000/Au1500. Manuel Lauss -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [topic/asoc][PATCH 0/2] minor clean-ups and aic3x naming change
by Jarkko Nikula 07 Oct '08

07 Oct '08

These two patches are generated agains Takashi's topic/asoc branch. First one will remove unused AUDIO_NAME define from codec drivers. IMO it didn't look so readable to use this define so I end up removing it. Second one will use uniform naming like other tlv320aic drivers for tlv320aic3x. -- Jarkko

3 4

[alsa-devel] size of snd_hda_intel
by Jon Smirl 07 Oct '08

07 Oct '08

>ismod snd_hda_intel 435,772 435,772 bytes seems awfully large for an audio driver. The large size seems to be due to having support for multiple codecs in the driver. Could a generic hda driver load the correct codec dynamically instead of building them all in statically? I have a Realtek codec. -- Jon Smirl jonsmirl(a)gmail.com

5 8

[alsa-devel] Fwd: AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 07 Oct '08

07 Oct '08

---------- Forwarded message ---------- From: Jesslyn Abdul Salam <jesslyn.abdulsalam(a)gmail.com> Date: Tue, Oct 7, 2008 at 5:36 PM Subject: Re: [alsa-devel] AIC23 SOC Driver on OSK5912 To: Arun KS <arunks(a)mistralsolutions.com> Yes, it is there. but it shows as "tlv320aic23b" instead of "tlv320aic" Will that be the reason ???? On Tue, Oct 7, 2008 at 5:32 PM, Arun KS <arunks(a)mistralsolutions.com> wrote: > On Tue, Oct 7, 2008 at 5:24 PM, Mark Brown <broonie(a)sirena.org.uk> wrote: > > On Tue, Oct 07, 2008 at 05:18:55PM +0530, Jesslyn Abdul Salam wrote: > > > >> I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for the > past > >> few days. I got the driver patch from the linux-omap/alsa-devel mailing > >> lists. The driver got compiled along just fine. But when the kernel > boots, > >> no sound cards have been listed. > > > > CCing in Arun KS <arunks(a)mistralsolutions.com> who contributed that > > support. > > > >> Advanced Linux Sound Architecture Driver Version 1.0.17. > > > >> ASoC version > >> 0.13.2 > >> AIC23 Audio Codec > >> 0.1 > >> MCLK = 12000000 [12000000], usecount = > >> 0 > >> ALSA device > >> list: > >> No soundcards > >> found. > > > >> Any one have tested this before ? > > Can you check i2c? > > # cat /sys/bus/i2c/devices/1-001b/name > tlv320aic23 > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

2 2

[alsa-devel] AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 07 Oct '08

07 Oct '08

Hi all, I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for the past few days. I got the driver patch from the linux-omap/alsa-devel mailing lists. The driver got compiled along just fine. But when the kernel boots, no sound cards have been listed. Advanced Linux Sound Architecture Driver Version 1.0.17. ASoC version 0.13.2 AIC23 Audio Codec 0.1 MCLK = 12000000 [12000000], usecount = 0 ALSA device list: No soundcards found. Any one have tested this before ? - Jesslyn Abdul Salam

3 2

[alsa-devel] [PATCH 2.6.27?] ASoC: Correct inverted Mic PGA Switch control in wm8510 driver
by Mark Brown 07 Oct '08

07 Oct '08

Mic PGA Switch should be inverted in the WM8510 driver but isn't. Reported-by: ext-jukka.hynninen(a)vaisala.com Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 75ed041..9a37c8d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -199,7 +199,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; static const struct snd_kcontrol_new wm8510_boost_controls[] = { -SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 1), SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), }; -- 1.5.6.5

2 2

[alsa-devel] [PATCH] snd-core: enlarge snd_card.components for up to 4 codecs
by Wu Fengguang 07 Oct '08

07 Oct '08

Enlarge snd_card.components[80] to 128 bytes, with space for 4 codecs. The previous size 80 cannot support HP 2230s which has 3 codecs. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- diff --git a/include/sound/asound.h b/include/sound/asound.h index ca2f358..b3801b0 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -708,7 +708,7 @@ struct snd_ctl_card_info { unsigned char longname[80]; /* name + info text about soundcard */ unsigned char reserved_[16]; /* reserved for future (was ID of mixer) */ unsigned char mixername[80]; /* visual mixer identification */ - unsigned char components[80]; /* card components / fine identification, delimited with one space (AC97 etc..) */ + unsigned char components[128]; /* card components / fine identification, delimited with one space (AC97 etc..) */ unsigned char reserved[48]; /* reserved for future */ }; diff --git a/include/sound/core.h b/include/sound/core.h index f52ab6f..dcbc424 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -116,7 +116,7 @@ struct snd_card { char shortname[32]; /* short name of this soundcard */ char longname[80]; /* name of this soundcard */ char mixername[80]; /* mixer name */ - char components[80]; /* card components delimited with + char components[128]; /* card components delimited with space */ struct module *module; /* top-level module */

3 7

[alsa-devel] [PATCH] hda: comment typo fix
by Wu Fengguang 07 Oct '08

07 Oct '08

fix a typo in comment for process_unsol_events(). Signed-off-by: Wu Fengguang<wfg(a)linux.intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c742e10..7717a92 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -314,7 +314,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) } /* - * process queueud unsolicited events + * process queued unsolicited events */ static void process_unsol_events(struct work_struct *work) {

2 2

[alsa-devel] [PATCH] hda: comment typo fix
by Wu Fengguang 07 Oct '08

07 Oct '08

fix a typo in comment for is_in_nid_list(). Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c742e10..6268667 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2807,7 +2807,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, } /* - * Helper for automatic ping configuration + * Helper for automatic pin configuration */ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)

1 0

[alsa-devel] [PATCH 1/1] ALSA: ASoC: add new param mux to dapm_mux_update_power
by Richard Zhao 07 Oct '08

07 Oct '08

Function dapm_mux_update_power needs enum index mux and register mask value val as parameters, but it only has a parameter val, and uses it as both val and mux. snd_soc_test_bits(widget->codec, e->reg, mask, val) val is register mask here, e->texts[val] but val should be enum index mux here. This patch adds a new param mux to fix it. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

1 0

[alsa-devel] [PATCH 1/1] ALSA: ASoC: Fix wrong param when call snd_soc_test_bits
by Richard Zhao 07 Oct '08

07 Oct '08

Fix wrong param when call snd_soc_test_bits in function dapm_mux_update_power. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

3 4

Re: [alsa-devel] [Alsa-user] [SOLVED] Re: snd_hda_intel on ALC662 (Classmate 2nd gen) does not mute and does not offer mic boost
by Vedran Miletić 07 Oct '08

07 Oct '08

http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=ac… See diff of that commit for some hints. Or don't. I got it wrong. :-( Basically, you need to find out subsystem ID for your soundcard. "lspci -vnn" will be your friend here. You should find out something like this for your soundcard: 00:07.0 Audio device [0403]: nVidia Corporation Device [10de:0774] (rev a1) Subsystem: ASRock Incorporation Device [1849:3662] <------ THIS IS THE IMPORTANT ROW Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 21 Memory at f9e78000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel So I should have put 0x1849, 0x3662 for my souncard instead of 0x1849, 0x0774. I probably wouldn't notice, but now I will patch it. Thanks for asking! 2008/10/4 Pedro Pessoa <pessoa(a)angulosolido.pt>: > On Saturday 04 October 2008 17:54, Vedran Miletić wrote: >> Can you create a patch to add this device to quirks list? > > Sure! Let me know what I have to do. > > The quirks list is something that will make 'auto' load ECS? > > -- > Angulo Sólido - Tecnologias de Informação > http://angulosolido.pt > -- Vedran Miletić

2 1

[alsa-devel] [PATCH] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 06 Oct '08

06 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic, moving the MIDI interfaces to the quirk data header. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..05c7849 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,37 +3016,12 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); - if (altsd->bNumEndpoints == 2) { - static const struct snd_usb_midi_endpoint_info ua700_ep = { - .out_cables = 0x0003, - .in_cables = 0x0003 - }; - static const struct snd_usb_audio_quirk ua700_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua700_ep - }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { - .out_cables = 0x0001, - .in_cables = 0x0001 - }; - static const struct snd_usb_audio_quirk ua25_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep - }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); - } - if (altsd->bNumEndpoints != 1) return -ENXIO; @@ -3376,9 +3351,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..ad7a318 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } }, { .ifnum = -1 @@ -1197,15 +1201,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } }, { .ifnum = -1 @@ -1338,6 +1346,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

4 3

Re: [alsa-devel] ICH9 Sound
by Ronald Horn 06 Oct '08

06 Oct '08

Hello, all mixer levels are ok. I have several times checked all settings. I deinstalled and reinstalled the drivers in yast2, too. The phenomenon is that then a sound is playing, once an incredible noice is playing over the original sound. And this sound is stopping first about ten seconds after I closed the player. It's like an feedback loop. Greetings Ronald > ICH9 sound should be supported. And, as you see, your card is > detected. Perhaps the codec or that specific notebook isn't supported. > Are you sure mixer levels are adjusted properly? > > I'm using a HP Pavilion dv7-1045eg. The Problem is I have no sound, > > because there is no driver witch works. > > Here are some reports from my system: > > / > > lspci -nv/ > > 00:1b.0 0403: 8086:293e (rev 03) > > Subsystem: 103c:30f4 > > Flags: bus master, fast devsel, latency 0, IRQ 22 > > Memory at df300000 (64-bit, non-prefetchable) [size=16K] > > Capabilities: [50] Power Management version 2 > > Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ > > Queue=0/0 Enable+ > > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > > Capabilities: [100] Virtual Channel <?> > > Capabilities: [130] Root Complex Link <?> > > Kernel driver in use: HDA Intel > > Kernel modules: snd-hda-intel > > > > / > > cat /proc/asound/cards/ > > 0 [Intel ]: HDA-Intel - HDA Intel > > HDA Intel at 0xdf300000 irq 22 > > > > > > /head -n 1 /proc/asound/card0/codec*/ > > ==> /proc/asound/card0/codec#0 <== > > Codec: IDT 92HD71B7X > > ==> /proc/asound/card0/codec#2 <== > > Codec: Generic 10de ID 6 > > > > > > I'm using openSuse 11.0 with Kernel 2.6.25.16-0.1-pae. > > Maybe your project could develop some drivers.

1 0

[alsa-devel] [PATCH]ASoC: Create a common directory for AVR32 and ARM9 atmel boards.
by Sedji Gaouaou 06 Oct '08

06 Oct '08

Create a common directory for AVR32 and ARM9 atmel boards to unify atmel audio DAI. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 2 +- sound/soc/atmel/Kconfig | 43 ++ sound/soc/atmel/Makefile | 15 + sound/soc/atmel/atmel-pcm.c | 484 +++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.h | 86 ++++ sound/soc/atmel/atmel_ssc_dai.c | 803 ++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 ++++++ sound/soc/atmel/playpaq_wm8510.c | 523 +++++++++++++++++++++++++ sound/soc/atmel/sam9g20_wm8731.c | 329 ++++++++++++++++ 10 files changed, 2406 insertions(+), 3 deletions(-) create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile create mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/atmel/atmel-pcm.h create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h create mode 100644 sound/soc/atmel/playpaq_wm8510.c create mode 100644 sound/soc/atmel/sam9g20_wm8731.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..615ebf0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349..4d475c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..170d9da --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the AT91 SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..f54a7cc --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..6d317d2 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,484 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..2ed1575 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,803 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} +#if 0 +/* + * Record the SSC system clock rate. + */ +static int atmel_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* + * The only clock supplied to the SSC is the AT91 master clock, + * which is only used if the SSC is generating BCLK and/or + * LRC clocks. + */ + switch (clk_id) { + case ATMEL_SYSCLK_MCK: + atmel_ssc_sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} +#endif +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..a828746 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..7e6560b --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,523 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..4e191df --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,329 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); -- 1.5.3.7

4 16

[alsa-devel] TASCAM US122L
by Fernando Perfumo 06 Oct '08

06 Oct '08

Hello all, following is an extract of kern.log when connecting and disconecting a TASCAM US122L : ( the pcm part of the us122L does not appear in /proc/asound/... but only the midi part. kernel : 2.6.27-rc5-git11, alsa-driver-1.0.18rc3 compiled separately of the kernel sources, alsa-lib-1.0.18rc3 and alsa-utils-1.0.18rc3 ) I'd appreciate some ideas about the segfault in alsactl and the ausence of the pcm functionality.. Thanks many, Fernando. Oct 6 14:18:17 64studio kernel: usb 2-1: new high speed USB device using ehci_hcd and address 10 Oct 6 14:18:17 64studio kernel: usb 2-1: config 1 interface 1 altsetting 1 bulk endpoint 0x83 has invalid maxpacket 9 Oct 6 14:18:17 64studio kernel: usb 2-1: configuration #1 chosen from 1 choice Oct 6 14:18:17 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:538: ffff880037942000:0 Oct 6 14:18:17 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:538: ffff88005e018000:1 Oct 6 14:18:17 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:103: 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:103: 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=-18 ul=1518 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=0 ul=1554 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:714: 23 106 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 1 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=-12 ul=1524 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=-30 ul=1518 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 6 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 11 46 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 88 46 -6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 176 46 -12 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 253 46 -18 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 1518 6 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 1518 11 46 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 1518 242 46 -6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 3036 6 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 4590 5 46 0 Oct 6 14:18:18 .... .............................................................. Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 44538 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 3 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 46056 5 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 50688 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-ker.................................. ......................................................................................................................................................................... Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 93696 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 95220 5 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 3 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 99846 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb.............................................................. ...................................................................................................................................................................... Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 153576 5 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 158208 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 165882 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 173556 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 3 Oct 6 14:18:19 64studio last message repeated 2 times Oct 6 14:18:19 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usbmidi.c:1368: created 1 output and 1 input ports Oct 6 14:18:19 64studio kernel: alsactl[25592]: segfault at 0 ip 00007fb5c9e2f5a0 sp 00007fffd26840b8 error 4 in libc-2.3.6.so[7fb5c9dbd000+121000] Oct 6 14:18:19 64studio kernel: alsactl[25594]: segfault at 0 ip 00007fc0e9fc45a0 sp 00007ffff2819238 error 4 in libc-2.3.6.so[7fc0e9f52000+121000] Oct 6 14:18:19 64studio kernel: alsactl[25670]: segfault at 0 ip 00007fd7550975a0 sp 00007fff5d8ee318 error 4 in libc-2.3.6.so[7fd755025000+121000] Oct 6 14:18:19 64studio kernel: alsactl[25671]: segfault at 0 ip 00007f2ab8cc65a0 sp 00007fffc151cf38 error 4 in libc-2.3.6.so[7f2ab8c54000+121000] Oct 6 14:18:47 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:456: underrun, status=4294967225 Oct 6 14:18:47 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:456: underrun, status=4294967225 Oct 6 14:18:47 64studio kernel: usb 2-1: USB disconnect, address 10

2 1

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with sys filesystem
by Troy Kisky 06 Oct '08

06 Oct '08

i.e. echo 6 59 >/sys/devices/platform/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..df23b23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1002,7 +1002,33 @@ static ssize_t codec_reg_show(struct device *dev, return count; } -static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); + +static ssize_t codec_reg_write(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + char *start = (char *)buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = dev_get_drvdata(dev); + struct snd_soc_codec *codec = devdata->codec; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return start - buf; +} + +static DEVICE_ATTR(codec_reg, 0644, codec_reg_show, codec_reg_write); /** * snd_soc_new_ac97_codec - initailise AC97 device -- 1.5.4.3

3 3

[alsa-devel] [PATCH] Sigmatel stac92hd71b7 low HP volume
by Klaas De Craemer 06 Oct '08

06 Oct '08

Hello, A few days ago, I posted a patch to the linux kernel list, but afterwards I realised that it is more appropriate for the alsa list. So I repost it here. Below is my patch for an issue on Dell E6500 notebooks with the Sigmatel stac92hd71b7 codec. In the default snd-hda-intel configuration, the sound going to the headphone jack goes comes directly from the DAC, resulting in very low volume output. The patch changes the codec's widget connections so that the internal mixer is put in between and sets its input amplifiers to the maximum value. The output to the internal speaker port has been left untouched. I have tested this and it gives a much better output. Klaas De Craemer diff -uNr linux-2.6.26.5/sound/pci/hda/patch_sigmatel.c linux-2.6.26.5.new/sound/pci/hda/patch_sigmatel.c --- linux-2.6.26.5/sound/pci/hda/patch_sigmatel.c 2008-09-08 19:40:20.000000000 +0200 +++ linux-2.6.26.5.new/sound/pci/hda/patch_sigmatel.c 2008-10-02 20:08:51.000000000 +0200 @@ -624,6 +624,22 @@ {} }; +static struct hda_verb stac92hd71b7_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* connect headphone jack to mixer */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Increase mixer volume for the two dac's */ + { 0x17, 0x370, 0x7f }, + { 0x17, 0x371, 0x7f }, + {} +}; + static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -3642,6 +3658,10 @@ spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; + case 0x111d76b2: /* Dell E6500 stac92hd71b7 configuration */ + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71b7_core_init; + break; default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init;

2 1

[alsa-devel] [patch] bugfix: SOUND: SOC: OMAP: Add support for Gumstix Overo
by David Brownell 06 Oct '08

06 Oct '08

From: David Brownell <dbrownell(a)users.sourceforge.net> Get rid of bogus ASOC boot messages on non-Overo boards, such as with a kernel built for both Overo and Beagle. Signed-off-by: David Brownell <dbrownell(a)users.sourceforge.net> Acked-By: Steve Sakoman <steve(a)sakoman.com> --- I'm not sure where the four patches for Overo support are in the ALSA queue (that's how they'll merge, yes?) but I'll suggest this one merge into the initial "overo.c" submit... sound/soc/omap/overo.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -107,11 +107,11 @@ static int __init overo_soc_init(void) { int ret; - printk(KERN_INFO "overo SoC init\n"); if (!machine_is_overo()) { - printk(KERN_ERR "Not Overo!\n"); + pr_debug("Not Overo!\n"); return -ENODEV; } + printk(KERN_INFO "overo SoC init\n"); overo_snd_device = platform_device_alloc("soc-audio", -1); if (!overo_snd_device) { @@ -135,13 +135,12 @@ err1: return ret; } +module_init(overo_soc_init); static void __exit overo_soc_exit(void) { platform_device_unregister(overo_snd_device); } - -module_init(overo_soc_init); module_exit(overo_soc_exit); MODULE_AUTHOR("Steve Sakoman <steve(a)sakoman.com>");

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[alsa-devel] [PATCH] ALSA: hda: add more board-specific information for Realtek ALC662 rev1
by Vedran Miletić 05 Oct '08

05 Oct '08

I recently got a chance to play with two boards with ALC662 rev1: * BIOSTAR TA780G M2+ * ASROCK K10N78FullHD-hSLI R3.0 Both use 3 stack, 6ch mode with digital out. Since autodetection isn't able to figure that out from BIOS, we need to specify that manually. BTW, thanks for fixing NVIDIA MCP78S [GeForce 8200] HD audio in 1.0.18, it's much appreciated :-) -- Vedran Miletić

2 3

[alsa-devel] [PATCH 2.6.27?] ASoC: Correct inverted Mic PGA Switch control in wm8510 driver
by Mark Brown 04 Oct '08

04 Oct '08

Mic PGA Switch should be inverted in the WM8510 driver but isn't. Reported-by: ext-jukka.hynninen(a)vaisala.com Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This bugfix applies to 2.6.27 but the issue is not very severe - it'd be good to get it in but it's not critical. sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 75ed041..f644da6 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -199,7 +199,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; static const struct snd_kcontrol_new wm8510_boost_controls[] = { -SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Mic PGA Switch", WM8509_INPPGA, 6, 1, 1), SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), }; -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 04 Oct '08

04 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic, moving the MIDI interfaces to the quirk data header. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..05c7849 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,37 +3016,12 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); - if (altsd->bNumEndpoints == 2) { - static const struct snd_usb_midi_endpoint_info ua700_ep = { - .out_cables = 0x0003, - .in_cables = 0x0003 - }; - static const struct snd_usb_audio_quirk ua700_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua700_ep - }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { - .out_cables = 0x0001, - .in_cables = 0x0001 - }; - static const struct snd_usb_audio_quirk ua25_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep - }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); - } - if (altsd->bNumEndpoints != 1) return -ENXIO; @@ -3376,9 +3351,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..ad7a318 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } }, { .ifnum = -1 @@ -1197,15 +1201,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } }, { .ifnum = -1 @@ -1338,6 +1346,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

2 1

[alsa-devel] snd-hda-intel overly loud when using OSS interface
by Jan Engelhardt 04 Oct '08

04 Oct '08

Hi, [don't forget to Cc me] I've got a Dell Precision 390 workstation, onboard sound running with snd-hda-intel module. 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) 00:1b.0 0403: 8086:27d8 (rev 01) When audio is output using the OSS interface (mpg123 or mplayer -ao oss), output is overly loud and the PCM mixer control has no effect. Master tuned in to 13% (alsamixer says dB gain=-40.50) is about the level as when using the ALSA interface running with Master 58% (dB gain=-19.50) with PCM 63% (dB gain=-19.50). $ uname -a Linux nuqneh 2.6.24-19-generic #1 SMP Thu Jul 10 22:38:03 PDT 2008 x86_64 GNU/Linux /proc/asound: $ cat cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xefffc000 irq 16 $ cat devices 0: [ 0] : control 1: : sequencer 4: [ 0- 0]: hardware dependent 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer $ cat hwdep 00-00: HDA Codec 0 $ cat modules 0 snd_hda_intel $ cat pcm 00-00: STAC92xx Analog : STAC92xx Analog : playback 1 : capture 1 $ cat timers G0: system timer : 4000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE Client application 22482 : running P0-0-1: PCM capture 0-0-1 : SLAVE $ cat version Advanced Linux Sound Architecture Driver Version 1.0.16. Compiled on Jul 11 2008 for kernel 2.6.24-19-generic (SMP). $ cat /proc/asound/card0/codec#0 Codec: SigmaTel STAC9200 Address: 0 Vendor Id: 0x83847690 Subsystem Id: 0x102801de Revision Id: 0x102201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=4, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x0a Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x08 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Delay: 3 samples Node 0x07 [Audio Selector] wcaps 0x300901: Stereo R/L Connection: 3 0x02* 0x08 0x0a Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect EAPD 0x0: Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x400001f1: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 2 0x05* 0x0a Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x12 0x12] Connection: 1 0x07 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 5 0x10* 0x0f 0x0e 0x0d 0x12 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02214030: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=30, enabled=1 Connection: 1 0x0b Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Trigger ImpSense Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19021: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x90100140: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x13 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x01813122: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00]

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[alsa-devel] [PATCH 7/7] ASoC: Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731).
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731). It is based on the former eti_b1_wm8731.c file, using the atmel scc API. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/sam9g20_wm8731.c | 329 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 329 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/sam9g20_wm8731.c diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..4e191df --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,329 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); -- 1.5.3.7 >From 0f63f310868d8135d091c9ac5eff679f7b34b894 Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Fri, 3 Oct 2008 11:53:59 +0200 Subject: [PATCH] Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731). Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/sam9g20_wm8731.c | 329 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 329 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/sam9g20_wm8731.c diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..4e191df --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,329 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); -- 1.5.3.7

1 0

[alsa-devel] [PATCH 6/7] ASoC: Change include files at32-pcm.h and at32-ssc.h to new atmel_ssc_dai.h and atmel-pcm.h.
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Change include files at32-pcm.h and at32-ssc.h(in playpaq_wm8510.c) to new atmel_ssc_dai.h and atmel-pcm.h. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/playpaq_wm8510.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 7e6560b..ffad18f 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -40,8 +40,8 @@ #include <mach/portmux.h> #include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" /*-------------------------------------------------------------------------*\ -- 1.5.3.7 >From 35c348266d0ddf8e47c3cf25a5db8ada2ff0a639 Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Fri, 3 Oct 2008 16:10:47 +0200 Subject: [PATCH] Change include files at32-pcm.h and at32-ssc.h to new atmel_ssc_dai.h and atmel-pcm.h. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/playpaq_wm8510.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 7e6560b..ffad18f 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -40,8 +40,8 @@ #include <mach/portmux.h> #include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" /*-------------------------------------------------------------------------*\ -- 1.5.3.7

1 0

[alsa-devel] [PATCH 5/7] ASoC: Move the file playpaq_wm8510.c from sound/soc/at32 directory to the new sound/soc/atmel directory.
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Move the file playpaq_wm8510.c from sound/soc/at32 directory to the new sound/soc/atmel directory. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/playpaq_wm8510.c | 523 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 523 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/playpaq_wm8510.c diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..ffad18f --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,523 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); -- 1.5.3.7 >From fd41c4ea4989e64238c45170d54fd06c4551f865 Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Fri, 3 Oct 2008 11:52:39 +0200 Subject: [PATCH] Move the file playpaq_wm8510.c from sound/soc/at32 directory to the new sound/soc/atmel directory. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/playpaq_wm8510.c | 523 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 523 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/playpaq_wm8510.c diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..ffad18f --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,523 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); -- 1.5.3.7

1 0

[alsa-devel] [PATCH 4/7] ASoC: Add files to manage the pcm interface for all atmel boards.
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Add files to manage the pcm interface for all atmel boards(merge between sound/soc/at91 sound/soc/at32). As well as the atmel_ssc_dai file, it will use the common atmel_ssc API. This file is based on at32-pcm. and at91-pcm.c. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/atmel-pcm.c | 484 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.h | 86 ++++++++ 2 files changed, 570 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/atmel/atmel-pcm.h diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..6d317d2 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,484 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ -- 1.5.3.7 >From 37a7a3d9f7c652f1912e448949b8173118144a20 Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Fri, 3 Oct 2008 11:39:03 +0200 Subject: [PATCH] Add files to manage the pcm interface for all atmel boards(merge between sound/soc/at91 sound/soc/at32) Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/atmel-pcm.c | 484 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.h | 86 ++++++++ 2 files changed, 570 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/atmel/atmel-pcm.h diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..6d317d2 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,484 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ -- 1.5.3.7

1 0

[alsa-devel] [PATCH 3/7] ASoC: Add files in the new sound/soc/atmel directory to manage the DAI ssc for all atmel boards.
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Add files in the new sound/soc/atmel directory to manage the DAI ssc for all atmel boards. It is based on at91-ssc and at32-ssc files. It will use the same ssc API across all the atmel platforms. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 782 +++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 ++++++ 2 files changed, 903 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..df02f2b --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,782 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..a828746 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ -- 1.5.3.7 >From eeb6c1281d60b5bdebcc7327d5c758f29139ac5e Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Fri, 3 Oct 2008 17:47:26 +0200 Subject: [PATCH] Add files in the new sound/soc/atmel directory to manage the DAI ssc for all atmel boards. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 782 +++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 ++++++ 2 files changed, 903 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..df02f2b --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,782 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..a828746 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ -- 1.5.3.7

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[alsa-devel] [PATCH 2/7] ASoC: Add new directory: sound/soc/atmel
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Add new directory: sound/soc/atmel which is a merge between sound/soc/at91 and sound/soc/at32. Add Makefile and Kconfig for this directory. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/Kconfig | 43 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/Makefile | 15 +++++++++++++++ 2 files changed, 58 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..170d9da --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the ATMEL SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..f54a7cc --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o -- 1.5.3.7 >From 2443ad5a23c54948ed86f774538b65ad69ff37ee Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Fri, 3 Oct 2008 11:25:36 +0200 Subject: [PATCH] Add new directry: sound/soc/atmel which is a merge between sound/soc/at91 and sound/soc/at32. Add Makefile and Kconfig for this directory. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/atmel/Kconfig | 43 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/Makefile | 15 +++++++++++++++ 2 files changed, 58 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..170d9da --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the ATMEL SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..f54a7cc --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o -- 1.5.3.7

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[alsa-devel] [PATCH 1/7] ASoC: Change Makefile and Kconfig to support atmel avr32 and at91 merging into a single atmel directory.
by Sedji Gaouaou 03 Oct '08

03 Oct '08

Change Makefile and Kconfig to support atmel avr32 and at91 merging into a single atmel directory. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/Kconfig | 3 +-- sound/soc/Makefile | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..615ebf0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349..4d475c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ -- 1.5.3.7 >From ec6e9878520679b27381cad11dd07f10fb9ce426 Mon Sep 17 00:00:00 2001 From: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> Date: Thu, 2 Oct 2008 19:51:31 +0200 Subject: [PATCH] Change Makefile and Kconfig to support atmel avr32 and at91 merging into a single atmel directory. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/Kconfig | 3 +-- sound/soc/Makefile | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..615ebf0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349..4d475c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ -- 1.5.3.7

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Re: [alsa-devel] [Alsa-user] em28xx_alsa: disagrees about version of symbol xxx after upgrading CentOS 5.2 ALSA .14 for ALSA .17
by Robert Vincent Krakora 03 Oct '08

03 Oct '08

All: I have searched on the web and it seems that at least one person has overcome the problem I am having by rebuilding soundcore.ko (See here: http://redmonk.com/sogrady/2008/04/29/sog-1-busted-sound-0-getting-audio-wo…). There is a sound_core.c in the ALSA tar ball. Why doesn't ALSA rebuild the soundcore.ko module? The functions that em28xx_alsa is complaining about reside in soundcore.ko according to my research. Basically, I downloaded .18rc3 driver, lib and util and configured, built and installed. I rebooted and em28xx_alsa complained that symbols that resided down in soundcore.ko did not jive with the versions of the same symbols for which it was trying to resolve linkages. I rebuilt the video4linux code, reinstalled and rebooted and the problem was still present. It looks as though one should be able to build an new soundcore.ko with a ALSA package but there are no instructions on how to do so. Please help!!! Best Regards, -- Rob Krakora Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

1 0

[alsa-devel] [PATCH] Adds custom SOC_SINGLE_TLV for tlv320aic23 codec
by Arun KS 03 Oct '08

03 Oct '08

Replaces SOC_ENUM with custom SOC_SINGLE_TLV for Sidetone volume Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/tlv320aic23.c | 53 +++++++++++++++++++++++++++++++++++++--- 1 files changed, 49 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c2d35e9..bb7cfb8 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -113,7 +113,6 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"}; static const struct soc_enum rec_src_enum = SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); @@ -125,11 +124,56 @@ static const struct soc_enum tlv320aic23_rec_src = SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); static const struct soc_enum tlv320aic23_deemph = SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); -static const struct soc_enum tlv320aic23_sidetone = - SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_text); static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); + +static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val, reg; + + val = (ucontrol->value.integer.value[0] & 0x07); + + /* linear conversion to userspace + * 000 = -6db + * 001 = -9db + * 010 = -12db + * 011 = -18db (Min) + * 100 = 0db (Max) + */ + val = (val >= 4) ? 4 : (3 - val); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); + tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + + return 0; +} + +static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val; + + val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = val >> 6; + val = (val >= 4) ? 4 : (3 - val); + ucontrol->value.integer.value[0] = val; + return 0; + +} + +#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ + .put = snd_soc_tlv320aic23_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, @@ -141,7 +185,8 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), - SOC_ENUM("Sidetone Gain", tlv320aic23_sidetone), + SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, + 6, 4, 0, sidetone_vol_tlv), SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; -- 1.5.3.4

2 1

[alsa-devel] More ASoC updates
by Mark Brown 03 Oct '08

03 Oct '08

The following changes since commit ddaea5119fd2efde7b3936d4ccfa34c4201be0d3: Takashi Iwai (1): Merge branch 'asoc-fixes' into topic/asoc are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai There's more to come on the TLV320AIC23 but at this point it's all cleanups and it's much easier to review incremental patches. Arun KS (3): ASoC: Add TLV320AIC23 codec driver ASoC: Add support for osk5912 ASoC: Add DSP DAI format support to the OMAP McBSP driver sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 670 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 122 ++++++++ sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-mcbsp.c | 5 + sound/soc/omap/osk5912.c | 232 ++++++++++++++ 8 files changed, 1046 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h create mode 100644 sound/soc/omap/osk5912.c

2 4

[alsa-devel] [PATCH 1/5 v2] ASoC Add TLV320AIC23 codec driver
by Arun KS 03 Oct '08

03 Oct '08

ASoC codec driver for TLV320AIC23 device Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 670 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 122 ++++++++ 4 files changed, 799 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0507fcf..bdead2d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" depends on SND_SOC && SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0731844..90f0a58 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..c2d35e9 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,670 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "tlv320aic23" +#define AIC23_VERSION "0.1" + +struct tlv320aic23_srate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, + (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); +static const struct soc_enum tlv320aic23_sidetone = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_ENUM("Sidetone Gain", tlv320aic23_sidetone), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((srate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(srate_reg_info))) { + count++; + } + + data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | + (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | + TLV320AIC23_USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (freq) { + case 12000000: + return 0; + } + return -EINVAL; +} + +static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); + tlv320aic23_write(codec, TLV320AIC23_LINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + (TLV320AIC23_LRS_ENABLED)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + TLV320AIC23_LRS_ENABLED); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); + tlv320aic23_write(codec, TLV320AIC23_ANLG, + (reg) & (~TLV320AIC23_BYPASS_ON) & + (~TLV320AIC23_MICM_MUTED)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..79d1faf --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ -- 1.5.3.4

3 2

[alsa-devel] snd_hda_intel shutdown noise
by Howard Chu 03 Oct '08

03 Oct '08

On my HP dv5z whenever shutting down or rebooting there's a loud pop on the speakers just before the screen shuts off. It seems to me that it might be worthwhile to zero the volume or set the mute bit on the output channels just before closing the device. I started looking into this but haven't found the right place to insert the code yet. Anyone else have any ideas? Will this even make a difference? -- -- Howard Chu CTO, Symas Corp. http://www.symas.com Director, Highland Sun http://highlandsun.com/hyc/ Chief Architect, OpenLDAP http://www.openldap.org/project/

2 1

[alsa-devel] ASoC updates
by Mark Brown 02 Oct '08

02 Oct '08

The following changes since commit 2e2ed79f64e935540b0d148f7e2c3ebefafe133e: Vladimir Barinov (1): ALSA: Correct Vladimir Barinov's e-mail address are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Of these only the patch from Rob is 2.6.27 material. Arun KS (1): ASoC: Add destination and source port for DMA on OMAP1 Frank Mandarino (1): ASoC: Remove references to Endrelia ETI-B1 board Jonas Bonn (2): ASoC: Add widgets before setting endpoints on GTA01 ASoC: Drop device registration from GTA01 lm4857 driver Rob Sims (1): ASoC: Set correct name for WM8753 rec mixer output sound/soc/at91/Kconfig | 17 -- sound/soc/at91/Makefile | 5 - sound/soc/at91/eti_b1_wm8731.c | 349 ------------------------------------ sound/soc/codecs/wm8753.c | 4 +- sound/soc/omap/omap-pcm.c | 4 +- sound/soc/s3c24xx/neo1973_wm8753.c | 57 +----- 6 files changed, 14 insertions(+), 422 deletions(-) delete mode 100644 sound/soc/at91/eti_b1_wm8731.c

2 6

[alsa-devel] [PATCH 1/5 v2] ASoC Add TLV320AIC23 codec driver
by Arun KS 02 Oct '08

02 Oct '08

ASoC codec driver for TLV320AIC23 device Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 670 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 122 ++++++++ 4 files changed, 799 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0507fcf..bdead2d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" depends on SND_SOC && SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0731844..90f0a58 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..c2d35e9 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,670 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "tlv320aic23" +#define AIC23_VERSION "0.1" + +struct tlv320aic23_srate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, + (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); +static const struct soc_enum tlv320aic23_sidetone = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_ENUM("Sidetone Gain", tlv320aic23_sidetone), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((srate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(srate_reg_info))) { + count++; + } + + data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | + (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | + TLV320AIC23_USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (freq) { + case 12000000: + return 0; + } + return -EINVAL; +} + +static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); + tlv320aic23_write(codec, TLV320AIC23_LINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + (TLV320AIC23_LRS_ENABLED)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + TLV320AIC23_LRS_ENABLED); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); + tlv320aic23_write(codec, TLV320AIC23_ANLG, + (reg) & (~TLV320AIC23_BYPASS_ON) & + (~TLV320AIC23_MICM_MUTED)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..79d1faf --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ -- 1.5.3.4

2 1

[alsa-devel] [PATCH] external plugin for adding delay to channels
by Alex 02 Oct '08

02 Oct '08

This should be interesting for home theater installations, as I was personnally missing a "per channel delay configuration". Feel free to tell me what should be modified, I am not really used to proposing a patch. Thx

2 1

[alsa-devel] [PATCH 5/5 v2] ASoC Adds DSP DAI format support for platform driver
by Arun KS 02 Oct '08

02 Oct '08

Enables DSP DAI format for mcbsp in omap platform driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e1..fb920e1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -245,6 +245,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; -- 1.5.3.4

1 0

[alsa-devel] [PATCH 4/5 v2] ASOC adds destination and source port for dma
by Arun KS 02 Oct '08

02 Oct '08

Adds destination and source port for dma in platform driver as required by OMAP1 Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-pcm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfea..e9084fd 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!cpu_is_omap1510()) { + if (!err & !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame -- 1.5.3.4

1 0

[alsa-devel] [PATCH 3/5 v2] Adds I2C board info for tlvaic23
by Arun KS 02 Oct '08

02 Oct '08

Adding I2C board info required for tlvaic23 i2c chip driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- arch/arm/mach-omap1/board-osk.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 3e766e4..e506aef 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -258,8 +258,10 @@ static struct i2c_board_info __initdata osk_i2c_board_info[] = { .platform_data = &tps_board, }, + { + I2C_BOARD_INFO("tlv320aic23", 0x1B), + }, /* TODO when driver support is ready: - * - aic23 audio chip at 0x1a * - on Mistral, 24c04 eeprom at 0x50 * - optionally on Mistral, ov9640 camera sensor at 0x30 */ -- 1.5.3.4

1 0

[alsa-devel] [PATCH 2/5 v2] Adds ASOC Support for osk5912
by Arun KS 02 Oct '08

02 Oct '08

Adding ASOC machine driver for osk5912 Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/osk5912.c | 232 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 242 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/osk5912.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e7..8b7766b 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on osk5912. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58..e09d1f2 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-osk5912-objs := osk5912.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c new file mode 100644 index 0000000..0fe7337 --- /dev/null +++ b/sound/soc/omap/osk5912.c @@ -0,0 +1,232 @@ +/* + * osk5912.c -- SoC audio for OSK 5912 + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks(a)mistralsolutions.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "OSK5912", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC OSK 5912"); +MODULE_LICENSE("GPL"); -- 1.5.3.4

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[alsa-devel] How to start with a driver for custom hardware?
by javier.martinc＠alumnos.unican.es 02 Oct '08

02 Oct '08

I am trying to develop an ALSA compatible driver for a custom hardware. The situation is the following: We are using a Cyclone II FPGA with a nommu NiosII processor and uClinux. We have an external TLV320AIC110 codec with I2C control interface and PCM audio interface. Into the FPGA we have a hardware module which controls both interfaces and basically copies x amount of data into a given memory position and vice versa. I have been looking at the documentation (which in my honest opinion is scarce) and many doubts have arisen. For example, should I used alsa or asoc? In my opinion, alsa documentation is more clear than asoc, but it is highlt oriented to pci drivers. As you can see it is a simple hardware which control registers are mapped into memory. Which part of the alsa driver takes care of accesing these hardware dependent registers? Maybe we should use some other standard hardware module into the FPGA to get better alsa compliance? I know these are very generic and begginer questions but, anyway, any help will be welcome. Thank you. Javier Martín

2 3

[alsa-devel] [PATCH 0/5 v2] ASoC Add TLV320AIC23 codec driver
by Arun KS 02 Oct '08

02 Oct '08

Hi all, This is the next version of patches for tlv320aic23 codec. Only patch 1 and 2 are different from their version 1 counterparts. Changes: ------------ Patch1 - Incorporated the comments from Mark Brown. Patch2- File name is changed to osk5912.c from 5912osk.c. Thanks Troy Kisky for your help and time. Since my mailer is word wrapping the patches, I am attaching them too. Thanks Arun

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[alsa-devel] [PATCH]ASoC: Remove at91 and at32 directories
by Sedji Gaouaou 02 Oct '08

02 Oct '08

First patch to unify atmel audio dai. Remove at91 and at32 directories. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/at32/Kconfig | 34 -- sound/soc/at32/Makefile | 11 - sound/soc/at32/at32-pcm.c | 492 ----------------------- sound/soc/at32/at32-pcm.h | 79 ---- sound/soc/at32/at32-ssc.c | 849 --------------------------------------- sound/soc/at32/at32-ssc.h | 59 --- sound/soc/at32/playpaq_wm8510.c | 523 ------------------------ sound/soc/at91/Kconfig | 10 - sound/soc/at91/Makefile | 6 - sound/soc/at91/at91-pcm.c | 434 -------------------- sound/soc/at91/at91-pcm.h | 72 ---- sound/soc/at91/at91-ssc.c | 791 ------------------------------------ sound/soc/at91/at91-ssc.h | 27 -- 13 files changed, 0 insertions(+), 3387 deletions(-) delete mode 100644 sound/soc/at32/Kconfig delete mode 100644 sound/soc/at32/Makefile delete mode 100644 sound/soc/at32/at32-pcm.c delete mode 100644 sound/soc/at32/at32-pcm.h delete mode 100644 sound/soc/at32/at32-ssc.c delete mode 100644 sound/soc/at32/at32-ssc.h delete mode 100644 sound/soc/at32/playpaq_wm8510.c delete mode 100644 sound/soc/at91/Kconfig delete mode 100644 sound/soc/at91/Makefile delete mode 100644 sound/soc/at91/at91-pcm.c delete mode 100644 sound/soc/at91/at91-pcm.h delete mode 100644 sound/soc/at91/at91-ssc.c delete mode 100644 sound/soc/at91/at91-ssc.h diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig deleted file mode 100644 index b0765e8..0000000 --- a/sound/soc/at32/Kconfig +++ /dev/null @@ -1,34 +0,0 @@ -config SND_AT32_SOC - tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" - depends on AVR32 && SND_SOC - help - Say Y or M if you want to add support for codecs attached to - the AT32 SSC interface. You will also need to - to select the audio interfaces to support below. - - -config SND_AT32_SOC_SSC - tristate - - - -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_AT32_SOC && BOARD_PLAYPAQ - select SND_AT32_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - - - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile deleted file mode 100644 index c03e55e..0000000 --- a/sound/soc/at32/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT32 Platform Support -snd-soc-at32-objs := at32-pcm.o -snd-soc-at32-ssc-objs := at32-ssc.o - -obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o -obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o - -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c deleted file mode 100644 index c83584f..0000000 --- a/sound/soc/at32/at32-pcm.c +++ /dev/null @@ -1,492 +0,0 @@ -/* sound/soc/at32/at32-pcm.c - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-pcm.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "at32-pcm.h" - - - -/*--------------------------------------------------------------------------*\ - * Hardware definition -\*--------------------------------------------------------------------------*/ -/* TODO: These values were taken from the AT91 platform driver, check - * them against real values for AT32 - */ -static const struct snd_pcm_hardware at32_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - - .formats = SNDRV_PCM_FMTBIT_S16, - .period_bytes_min = 32, - .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - - - -/*--------------------------------------------------------------------------*\ - * Data types -\*--------------------------------------------------------------------------*/ -struct at32_runtime_data { - struct at32_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of DMA buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - - dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ - - /* Save PDC registers (for power management) */ - u32 pdc_xpr_save; - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - - - -/*--------------------------------------------------------------------------*\ - * Helper functions -\*--------------------------------------------------------------------------*/ -static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *dmabuf = &substream->dma_buffer; - size_t size = at32_pcm_hardware.buffer_bytes_max; - - dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; - dmabuf->dev.dev = pcm->card->dev; - dmabuf->private_data = NULL; - dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, - &dmabuf->addr, GFP_KERNEL); - pr_debug("at32_pcm: preallocate_dma_buffer: " - "area=%p, addr=%p, size=%ld\n", - (void *)dmabuf->area, (void *)dmabuf->addr, size); - - if (!dmabuf->area) - return -ENOMEM; - - dmabuf->bytes = size; - return 0; -} - - - -/*--------------------------------------------------------------------------*\ - * ISR -\*--------------------------------------------------------------------------*/ -static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - static int count; - - count++; - if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "underrun" : "overrun", params->name, ssc_sr, count); - - /* re-start the PDC */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - } - - - if (ssc_sr & params->mask->ssc_endx) { - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - - snd_pcm_period_elapsed(substream); -} - - - -/*--------------------------------------------------------------------------*\ - * PCM operations -\*--------------------------------------------------------------------------*/ -static int at32_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params - */ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at32_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - pr_debug("hw_params: DMA for %s initialized " - "(dma_bytes=%ld, period_size=%ld)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - - return 0; -} - - - -static int at32_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - - - -static int at32_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - ssc_writex(params->ssc->regs, SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - return 0; -} - - -static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - int ret = 0; - - pr_debug("at32_pcm_trigger: buffer_size = %ld, " - "dma_area = %p, dma_bytes = %ld\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - pr_debug("trigger: period_ptr=%lx, xpr=%x, " - "xcr=%d, xnpr=%x, xncr=%d\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); - - ssc_writex(params->ssc->regs, SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_enable); - - pr_debug("sr=%x, imr=%x\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); - break; /* SNDRV_PCM_TRIGGER_START */ - - - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - break; - - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - - - -static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - - return x; -} - - - -static int at32_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - -out: - return ret; -} - - - -static int at32_pcm_close(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - - -static int at32_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - - - -static struct snd_pcm_ops at32_pcm_ops = { - .open = at32_pcm_open, - .close = at32_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at32_pcm_hw_params, - .hw_free = at32_pcm_hw_free, - .prepare = at32_pcm_prepare, - .trigger = at32_pcm_trigger, - .pointer = at32_pcm_pointer, - .mmap = at32_pcm_mmap, -}; - - - -/*--------------------------------------------------------------------------*\ - * ASoC platform driver -\*--------------------------------------------------------------------------*/ -static u64 at32_pcm_dmamask = 0xffffffff; - -static int at32_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at32_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - - -out: - return ret; -} - - - -static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream == NULL) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - - - -#ifdef CONFIG_PM -static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Disable the PDC and save the PDC registers */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); - - return 0; -} - - - -static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else /* CONFIG_PM */ -# define at32_pcm_suspend NULL -# define at32_pcm_resume NULL -#endif /* CONFIG_PM */ - - - -struct snd_soc_platform at32_soc_platform = { - .name = "at32-audio", - .pcm_ops = &at32_pcm_ops, - .pcm_new = at32_pcm_new, - .pcm_free = at32_pcm_free_dma_buffers, - .suspend = at32_pcm_suspend, - .resume = at32_pcm_resume, -}; -EXPORT_SYMBOL_GPL(at32_soc_platform); - - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("Atmel AT32 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h deleted file mode 100644 index 2a52430..0000000 --- a/sound/soc/at32/at32-pcm.h +++ /dev/null @@ -1,79 +0,0 @@ -/* sound/soc/at32/at32-pcm.h - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_PCM_H -#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ - -#include <linux/atmel-ssc.h> - - -/* - * Registers and status bits that are required by the PCM driver - * TODO: Is ptcr really used? - */ -struct at32_pdc_regs { - u32 xpr; /* PDC RX/TX pointer */ - u32 xcr; /* PDC RX/TX counter */ - u32 xnpr; /* PDC next RX/TX pointer */ - u32 xncr; /* PDC next RX/TX counter */ - u32 ptcr; /* PDC transfer control */ -}; - - - -/* - * SSC mask info - */ -struct at32_ssc_mask { - u32 ssc_enable; /* SSC RX/TX enable */ - u32 ssc_disable; /* SSC RX/TX disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ - u32 pdc_enable; /* PDC RX/TX enable */ - u32 pdc_disable; /* PDC RX/TX disable */ -}; - - - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at32_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - struct ssc_device *ssc; /* SSC device for stream */ - struct at32_pdc_regs *pdc; /* PDC register info */ - struct at32_ssc_mask *mask; /* SSC mask info */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler) (u32, struct snd_pcm_substream *); -}; - - - -/* - * The AT32 ASoC platform driver - */ -extern struct snd_soc_platform at32_soc_platform; - - - -/* - * SSC register access (since ssc_writel() / ssc_readl() require literal name) - */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) - -#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c deleted file mode 100644 index 4ef6492..0000000 --- a/sound/soc/at32/at32-ssc.c +++ /dev/null @@ -1,849 +0,0 @@ -/* sound/soc/at32/at32-ssc.c - * ASoC platform driver for AT32 using SSC as DAI - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-ssc.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -/* #define DEBUG */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "at32-pcm.h" -#include "at32-ssc.h" - - - -/*-------------------------------------------------------------------------*\ - * Constants -\*-------------------------------------------------------------------------*/ -#define NUM_SSC_DEVICES 3 - -/* - * SSC direction masks - */ -#define SSC_DIR_MASK_UNUSED 0 -#define SSC_DIR_MASK_PLAYBACK 1 -#define SSC_DIR_MASK_CAPTURE 2 - -/* - * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These - * are expected to be used with SSC_BF - */ -/* START bit field values */ -#define SSC_START_CONTINUOUS 0 -#define SSC_START_TX_RX 1 -#define SSC_START_LOW_RF 2 -#define SSC_START_HIGH_RF 3 -#define SSC_START_FALLING_RF 4 -#define SSC_START_RISING_RF 5 -#define SSC_START_LEVEL_RF 6 -#define SSC_START_EDGE_RF 7 -#define SSS_START_COMPARE_0 8 - -/* CKI bit field values */ -#define SSC_CKI_FALLING 0 -#define SSC_CKI_RISING 1 - -/* CKO bit field values */ -#define SSC_CKO_NONE 0 -#define SSC_CKO_CONTINUOUS 1 -#define SSC_CKO_TRANSFER 2 - -/* CKS bit field values */ -#define SSC_CKS_DIV 0 -#define SSC_CKS_CLOCK 1 -#define SSC_CKS_PIN 2 - -/* FSEDGE bit field values */ -#define SSC_FSEDGE_POSITIVE 0 -#define SSC_FSEDGE_NEGATIVE 1 - -/* FSOS bit field values */ -#define SSC_FSOS_NONE 0 -#define SSC_FSOS_NEGATIVE 1 -#define SSC_FSOS_POSITIVE 2 -#define SSC_FSOS_LOW 3 -#define SSC_FSOS_HIGH 4 -#define SSC_FSOS_TOGGLE 5 - -#define START_DELAY 1 - - - -/*-------------------------------------------------------------------------*\ - * Module data -\*-------------------------------------------------------------------------*/ -/* - * SSC PDC registered required by the PCM DMA engine - */ -static struct at32_pdc_regs pdc_tx_reg = { - .xpr = SSC_PDC_TPR, - .xcr = SSC_PDC_TCR, - .xnpr = SSC_PDC_TNPR, - .xncr = SSC_PDC_TNCR, -}; - - - -static struct at32_pdc_regs pdc_rx_reg = { - .xpr = SSC_PDC_RPR, - .xcr = SSC_PDC_RCR, - .xnpr = SSC_PDC_RNPR, - .xncr = SSC_PDC_RNCR, -}; - - - -/* - * SSC and PDC status bits for transmit and receive - */ -static struct at32_ssc_mask ssc_tx_mask = { - .ssc_enable = SSC_BIT(CR_TXEN), - .ssc_disable = SSC_BIT(CR_TXDIS), - .ssc_endx = SSC_BIT(SR_ENDTX), - .ssc_endbuf = SSC_BIT(SR_TXBUFE), - .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), -}; - - - -static struct at32_ssc_mask ssc_rx_mask = { - .ssc_enable = SSC_BIT(CR_RXEN), - .ssc_disable = SSC_BIT(CR_RXDIS), - .ssc_endx = SSC_BIT(SR_ENDRX), - .ssc_endbuf = SSC_BIT(SR_RXBUFF), - .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), -}; - - - -/* - * DMA parameters for each SSC - */ -static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - { - { - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, -}; - - - -static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, -}; - - - - -/*-------------------------------------------------------------------------*\ - * ISR -\*-------------------------------------------------------------------------*/ -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt - * handler in the PCM driver. - */ -static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) -{ - struct at32_ssc_info *ssc_p = dev_id; - struct at32_pcm_dma_params *dma_params; - u32 ssc_sr; - u32 ssc_substream_mask; - int i; - - ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & - ssc_readl(ssc_p->ssc->regs, IMR)); - - /* - * Loop through substreams attached to this SSC. If a DMA-related - * interrupt occured on that substream, call the DMA interrupt - * handler function, if one has been registered in the dma_param - * structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if ((dma_params != NULL) && - (dma_params->dma_intr_handler != NULL)) { - ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); - if (ssc_sr & ssc_substream_mask) { - dma_params->dma_intr_handler(ssc_sr, - dma_params-> - substream); - } - } - } - - - return IRQ_HANDLED; -} - -/*-------------------------------------------------------------------------*\ - * DAI functions -\*-------------------------------------------------------------------------*/ -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at32_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - - - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at32_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - int dir_mask; - - dma_params = ssc_p->dma_params[substream->stream]; - - if (dma_params != NULL) { - ssc_writel(dma_params->ssc->regs, CR, - dma_params->mask->ssc_disable); - pr_debug("%s disabled SSC_SR=0x%08x\n", - (substream->stream ? "receiver" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - - dma_params->ssc = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[substream->stream] = NULL; - } - - - dir_mask = 1 << substream->stream; - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock */ - pr_debug("at32-ssc: Stopping user %d clock\n", - ssc_p->ssc->user); - clk_disable(ssc_p->ssc->clk); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc->irq, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - /* clear the SSC dividers */ - ssc_p->cmr_div = 0; - ssc_p->tcmr_period = 0; - ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - - - -/* - * Set the SSC system clock rate - */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* TODO: What the heck do I do here? */ - return 0; -} - - - -/* - * Record DAI format for use by hw_params() - */ -static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - - - -/* - * Record SSC clock dividers for use in hw_params() - */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT32_SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT32_SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT32_SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - - - -/* - * Configure the SSC - */ -static int at32_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at32_ssc_info *ssc_p = &ssc_info[id]; - struct at32_pcm_dma_params *dma_params; - int channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - - /* - * Currently, there is only one set of dma_params for each direction. - * If more are added, this code will have to be changed to select - * the proper set - */ - dma_params = &ssc_dma_params[id][substream->stream]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[substream->stream] = dma_params; - - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the PCM driver's hw_params() - * function. It should not be used for other purposes as it - * is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - - /* - * Determine sample size in bits and the PDC increment - */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - - case SNDRV_PCM_FORMAT_S16: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - - case SNDRV_PCM_FORMAT_S24: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - - case SNDRV_PCM_FORMAT_S32: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - - default: - pr_warning("at32-ssc: Unsupported PCM format %d", - params_format(params)); - return -EINVAL; - } - pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", - bits, dma_params->pdc_xfer_size, channels); - - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) - if (bits > 16) { - pr_warning("at32-ssc: " - "sample size %d is too large for I2S\n", - bits); - return -EINVAL; - } - - - /* - * Compute the SSC register settings - */ - switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRS clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, SSC_START_FALLING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(RFMR_FSLEN, bits - 1) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, SSC_START_FALLING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(TFMR_FSLEN, bits - 1) | - SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | - SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clock. - * - * The SSC transmit clock is obtained from the BCLK signal - * on the TK line, and the SSC receive clock is generated from - * the transmit clock. - * - * For single channel data, one sample is transferred on the - * falling edge of the LRC clock. For two channel data, one - * sample is transferred on both edges of the LRC clock. - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); - start_event = ((channels == 1) ? - SSC_START_FALLING_RF : SSC_START_EDGE_RF); - - rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, start_event) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, start_event) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_NONE) | - SSC_BF(TCMR_CKS, SSC_CKS_PIN)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, 1) | - SSC_BF(RCMR_START, SSC_START_RISING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, 1) | - SSC_BF(TCMR_START, SSC_START_RISING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_RISING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(TFMR_DATNB, channels - 1) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - pr_warning("at32-ssc: unsupported DAI format 0x%x\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", - rcmr, rfmr, tcmr, tfmr); - - - if (!ssc_p->initialized) { - /* enable peripheral clock */ - pr_debug("at32-ssc: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); - - ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, - ssc_p->name, ssc_p); - if (ret < 0) { - pr_warning("at32-ssc: request irq failed (%d)\n", ret); - pr_debug("at32-ssc: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - return ret; - } - - ssc_p->initialized = 1; - } - - /* Set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); - - /* set transmit clock mode and format */ - ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); - ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); - - pr_debug("at32-ssc: SSC initialized\n"); - return 0; -} - - - -static int at32_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - - dma_params = ssc_p->dma_params[substream->stream]; - - ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); - - return 0; -} - - - -#ifdef CONFIG_PM -static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive */ - ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); - - /* Save the current interrupt mask, then disable unmasked interrupts */ - ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); - ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); - ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); - ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); - ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); - ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); - - return 0; -} - - - -static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - u32 cr; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* restore SSC register settings */ - ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); - ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); - ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); - - /* re-enable interrupts */ - ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - - /* Re-enable recieve and transmit as appropriate */ - cr = 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; - ssc_writel(ssc_p->ssc->regs, CR, cr); - - return 0; -} -#else /* CONFIG_PM */ -# define at32_ssc_suspend NULL -# define at32_ssc_resume NULL -#endif /* CONFIG_PM */ - - -#define AT32_SSC_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - - -#define AT32_SSC_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) - - -struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { - { - .name = "at32-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[0], - }, - { - .name = "at32-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[1], - }, - { - .name = "at32-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[2], - }, -}; -EXPORT_SYMBOL_GPL(at32_ssc_dai); - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h deleted file mode 100644 index 3c052db..0000000 --- a/sound/soc/at32/at32-ssc.h +++ /dev/null @@ -1,59 +0,0 @@ -/* sound/soc/at32/at32-ssc.h - * ASoC SSC interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_SSC_H -#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ - -#include <linux/types.h> -#include <linux/atmel-ssc.h> - -#include "at32-pcm.h" - - - -struct at32_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - - - -struct at32_ssc_info { - char *name; - struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* true if SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at32_pcm_dma_params *dma_params[2]; - struct at32_ssc_state ssc_state; -}; - - -/* SSC divider ids */ -#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - - -extern struct snd_soc_dai at32_ssc_dai[]; - - - -#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c deleted file mode 100644 index 7e6560b..0000000 --- a/sound/soc/at32/playpaq_wm8510.c +++ /dev/null @@ -1,523 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/version.h> -#include <linux/kernel.h> -#include <linux/errno.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/at32ap700x.h> -#include <mach/portmux.h> - -#include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_3; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const char *intercon[][3] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_codec *codec) -{ - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } - - - /* always connected pins */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync(codec); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai = &at32_ssc_dai[0], - .codec_dai = &wm8510_dai, - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_machine snd_soc_machine_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - - - -static struct wm8510_setup_data playpaq_wm8510_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - - - -static struct snd_soc_device playpaq_wm8510_snd_devdata = { - .machine = &snd_soc_machine_playpaq, - .platform = &at32_soc_platform, - .codec_dev = &soc_codec_dev_wm8510, - .codec_data = &playpaq_wm8510_setup, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc = NULL; - - - /* - * Request SSC device - */ - ssc = ssc_request(0); - if (IS_ERR(ssc)) { - ret = PTR_ERR(ssc); - ssc = NULL; - goto err_ssc; - } - ssc_p->ssc = ssc; - - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - goto err_pll0; - } - if (clk_set_parent(_gclk0, _pll0)) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); - playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } -err_gclk0: - if (ssc != NULL) { - ssc_free(ssc); - ssc = NULL; - } -err_ssc: - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc; - - if (ssc_p != NULL) { - ssc = ssc_p->ssc; - if (ssc != NULL) - ssc_free(ssc); - ssc_p->ssc = NULL; - } - - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig deleted file mode 100644 index 85a8832..0000000 --- a/sound/soc/at91/Kconfig +++ /dev/null @@ -1,10 +0,0 @@ -config SND_AT91_SOC - tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 - help - Say Y or M if you want to add support for codecs attached to - the AT91 SSC interface. You will also need - to select the audio interfaces to support below. - -config SND_AT91_SOC_SSC - tristate diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile deleted file mode 100644 index b817f11..0000000 --- a/sound/soc/at91/Makefile +++ /dev/null @@ -1,6 +0,0 @@ -# AT91 Platform Support -snd-soc-at91-objs := at91-pcm.o -snd-soc-at91-ssc-objs := at91-ssc.o - -obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c deleted file mode 100644 index 7ab48bd..0000000 --- a/sound/soc/at91/at91-pcm.c +++ /dev/null @@ -1,434 +0,0 @@ -/* - * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "at91-pcm: " x) -#else -#define DBG(x...) -#endif - -static const struct snd_pcm_hardware at91_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - -struct at91_runtime_data { - struct at91_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - u32 pdc_xpr_save; /* PDC register save */ - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - -static void at91_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - static int count = 0; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - - printk(KERN_WARNING - "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, - prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - -static int at91_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at91_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - return 0; -} - -static int at91_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int at91_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - at91_ssc_write(params->ssc_base + AT91_SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; -} - -static int at91_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", - (unsigned long) prtd->period_ptr, - at91_ssc_read(params->ssc_base + params->pdc->xpr), - at91_ssc_read(params->ssc_base + params->pdc->xcr), - at91_ssc_read(params->ssc_base + params->pdc->xnpr), - at91_ssc_read(params->ssc_base + params->pdc->xncr)); - - at91_ssc_write(params->ssc_base + AT91_SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, - params->mask->pdc_enable); - - DBG("sr=%lx imr=%lx\n", - at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t at91_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - return x; -} - -static int at91_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int at91_pcm_close(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int at91_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -struct snd_pcm_ops at91_pcm_ops = { - .open = at91_pcm_open, - .close = at91_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at91_pcm_hw_params, - .hw_free = at91_pcm_hw_free, - .prepare = at91_pcm_prepare, - .trigger = at91_pcm_trigger, - .pointer = at91_pcm_pointer, - .mmap = at91_pcm_mmap, -}; - -static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = at91_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static u64 at91_pcm_dmamask = 0xffffffff; - -static int at91_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at91_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -#ifdef CONFIG_PM -static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr); - prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr); - prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr); - prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr); - - return 0; -} - -static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save); - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define at91_pcm_suspend NULL -#define at91_pcm_resume NULL -#endif - -struct snd_soc_platform at91_soc_platform = { - .name = "at91-audio", - .pcm_ops = &at91_pcm_ops, - .pcm_new = at91_pcm_new, - .pcm_free = at91_pcm_free_dma_buffers, - .suspend = at91_pcm_suspend, - .resume = at91_pcm_resume, -}; - -EXPORT_SYMBOL_GPL(at91_soc_platform); - -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("Atmel AT91 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h deleted file mode 100644 index e5aada2..0000000 --- a/sound/soc/at91/at91-pcm.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.h by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_PCM_H -#define _AT91_PCM_H - -#include <mach/hardware.h> - -struct at91_ssc_periph { - void __iomem *base; - u32 pid; -}; - -/* - * Registers and status bits that are required by the PCM driver. - */ -struct at91_pdc_regs { - unsigned int xpr; /* PDC recv/trans pointer */ - unsigned int xcr; /* PDC recv/trans counter */ - unsigned int xnpr; /* PDC next recv/trans pointer */ - unsigned int xncr; /* PDC next recv/trans counter */ - unsigned int ptcr; /* PDC transfer control */ -}; - -struct at91_ssc_mask { - u32 ssc_enable; /* SSC recv/trans enable */ - u32 ssc_disable; /* SSC recv/trans disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ - u32 pdc_enable; /* PDC recv/trans enable */ - u32 pdc_disable; /* PDC recv/trans disable */ -}; - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at91_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - void __iomem *ssc_base; /* SSC base address */ - struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */ - struct at91_ssc_mask *mask;/* SSC & PDC status bits */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler)(u32, struct snd_pcm_substream *); -}; - -extern struct snd_soc_platform at91_soc_platform; - -#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -#define at91_ssc_write(a,v) __raw_writel((v),(a)) - -#endif /* _AT91_PCM_H */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c deleted file mode 100644 index a5b1a79..0000000 --- a/sound/soc/at91/at91-ssc.c +++ /dev/null @@ -1,791 +0,0 @@ -/* - * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * - * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_pmc.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x) -#else -#define DBG(x...) -#endif - -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) -#define NUM_SSC_DEVICES 1 -#else -#define NUM_SSC_DEVICES 3 -#endif - - -/* - * SSC PDC registers required by the PCM DMA engine. - */ -static struct at91_pdc_regs pdc_tx_reg = { - .xpr = ATMEL_PDC_TPR, - .xcr = ATMEL_PDC_TCR, - .xnpr = ATMEL_PDC_TNPR, - .xncr = ATMEL_PDC_TNCR, -}; - -static struct at91_pdc_regs pdc_rx_reg = { - .xpr = ATMEL_PDC_RPR, - .xcr = ATMEL_PDC_RCR, - .xnpr = ATMEL_PDC_RNPR, - .xncr = ATMEL_PDC_RNCR, -}; - -/* - * SSC & PDC status bits for transmit and receive. - */ -static struct at91_ssc_mask ssc_tx_mask = { - .ssc_enable = AT91_SSC_TXEN, - .ssc_disable = AT91_SSC_TXDIS, - .ssc_endx = AT91_SSC_ENDTX, - .ssc_endbuf = AT91_SSC_TXBUFE, - .pdc_enable = ATMEL_PDC_TXTEN, - .pdc_disable = ATMEL_PDC_TXTDIS, -}; - -static struct at91_ssc_mask ssc_rx_mask = { - .ssc_enable = AT91_SSC_RXEN, - .ssc_disable = AT91_SSC_RXDIS, - .ssc_endx = AT91_SSC_ENDRX, - .ssc_endbuf = AT91_SSC_RXBUFF, - .pdc_enable = ATMEL_PDC_RXTEN, - .pdc_disable = ATMEL_PDC_RXTDIS, -}; - - -/* - * DMA parameters. - */ -static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - {{ - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#if NUM_SSC_DEVICES == 3 - {{ - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, - {{ - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#endif -}; - -struct at91_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - -static struct at91_ssc_info { - char *name; - struct at91_ssc_periph ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* 1=SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at91_pcm_dma_params *dma_params[2]; - struct at91_ssc_state ssc_state; - -} ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = 0, - .initialized = 0, - }, -#if NUM_SSC_DEVICES == 3 - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = 0, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = 0, - .initialized = 0, - }, -#endif -}; - -static unsigned int at91_ssc_sysclk; - -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA - * interrupt handler in the PCM driver. - */ -static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id) -{ - struct at91_ssc_info *ssc_p = dev_id; - struct at91_pcm_dma_params *dma_params; - u32 ssc_sr; - int i; - - ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR) - & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - - /* - * Loop through the substreams attached to this SSC. If - * a DMA-related interrupt occurred on that substream, call - * the DMA interrupt handler function, if one has been - * registered in the dma_params structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if (dma_params != NULL && dma_params->dma_intr_handler != NULL && - (ssc_sr & - (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) - - dma_params->dma_intr_handler(ssc_sr, dma_params->substream); - } - - return IRQ_HANDLED; -} - -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at91_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - DBG("ssc_startup: SSC_SR=0x%08lx\n", - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at91_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir, dir_mask; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - if (dma_params != NULL) { - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_disable); - DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - - dma_params->ssc_base = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[dir] = NULL; - } - - dir_mask = 1 << dir; - - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc.pid, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - -/* - * Record the SSC system clock rate. - */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* - * The only clock supplied to the SSC is the AT91 master clock, - * which is only used if the SSC is generating BCLK and/or - * LRC clocks. - */ - switch (clk_id) { - case AT91_SYSCLK_MCK: - at91_ssc_sysclk = freq; - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * Record the DAI format for use in hw_params(). - */ -static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - -/* - * Record SSC clock dividers for use in hw_params(). - */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT91SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value. - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT91SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT91SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -/* - * Configure the SSC. - */ -static int at91_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at91_ssc_info *ssc_p = &ssc_info[id]; - struct at91_pcm_dma_params *dma_params; - int dir, channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - /* - * Currently, there is only one set of dma params for - * each direction. If more are added, this code will - * have to be changed to select the proper set. - */ - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc_base = ssc_p->ssc.base; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - /* - * Determine sample size in bits and the PDC increment. - */ - switch(params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format\n"); - return -EINVAL; - } - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "at91-ssc: sample size %d is too large for I2S\n", bits); - return -EINVAL; - } - - /* - * Compute SSC register settings. - */ - switch (ssc_p->daifmt - & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is generated from the - * transmit clock. - * - * For single channel data, one sample is transferred on the falling - * edge of the LRC clock. For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = channels == 1 - ? AT91_SSC_START_FALLING_RF - : AT91_SSC_START_EDGE_RF; - - rcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - - - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); - - if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - DBG("Starting pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); - - /* Reset the SSC and its PDC registers */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0); - - if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt, - 0, ssc_p->name, ssc_p)) < 0) { - printk(KERN_WARNING "at91-ssc: request_irq failure\n"); - - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - return ret; - } - - ssc_p->initialized = 1; - } - - /* set SSC clock mode register */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr); - - /* set transmit clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr); - - DBG("hw_params: SSC initialized\n"); - return 0; -} - - -static int at91_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_enable); - - DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", - at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR)); - return 0; -} - - -#ifdef CONFIG_PM -static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive. */ - ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - AT91_SSC_TXDIS | AT91_SSC_RXDIS); - - /* Save the current interrupt mask, then disable unmasked interrupts. */ - ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR); - ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR); - ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR); - ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR); - ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR); - - return 0; -} - -static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); - - return 0; -} - -#else -#define at91_ssc_suspend NULL -#define at91_ssc_resume NULL -#endif - -#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - -#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { - { .name = "at91-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[0].ssc, - }, -#if NUM_SSC_DEVICES == 3 - { .name = "at91-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[1].ssc, - }, - { .name = "at91-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[2].ssc, - }, -#endif -}; - -EXPORT_SYMBOL_GPL(at91_ssc_dai); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino(a)endrelia.com, www.endrelia.com"); -MODULE_DESCRIPTION("AT91 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h deleted file mode 100644 index 6b7bf38..0000000 --- a/sound/soc/at91/at91-ssc.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Jan 9, 2007 - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_SSC_H -#define _AT91_SSC_H - -/* SSC system clock ids */ -#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ - -/* SSC divider ids */ -#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - -extern struct snd_soc_dai at91_ssc_dai[]; - -#endif /* _AT91_SSC_H */ - -- 1.5.3.7

1 0

[alsa-devel] [PATCH] ASoC: Remove at91 and at32 directories
by Sedji Gaouaou 01 Oct '08

01 Oct '08

First patch to unify atmel audio dai. Remove at91 and at32 directories. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/at32/Kconfig | 34 -- sound/soc/at32/Makefile | 11 - sound/soc/at32/at32-pcm.c | 492 ----------------------- sound/soc/at32/at32-pcm.h | 79 ---- sound/soc/at32/at32-ssc.c | 849 --------------------------------------- sound/soc/at32/at32-ssc.h | 59 --- sound/soc/at32/playpaq_wm8510.c | 523 ------------------------ sound/soc/at91/Kconfig | 10 - sound/soc/at91/Makefile | 6 - sound/soc/at91/at91-pcm.c | 434 -------------------- sound/soc/at91/at91-pcm.h | 72 ---- sound/soc/at91/at91-ssc.c | 791 ------------------------------------ sound/soc/at91/at91-ssc.h | 27 -- 13 files changed, 0 insertions(+), 3387 deletions(-) delete mode 100644 sound/soc/at32/Kconfig delete mode 100644 sound/soc/at32/Makefile delete mode 100644 sound/soc/at32/at32-pcm.c delete mode 100644 sound/soc/at32/at32-pcm.h delete mode 100644 sound/soc/at32/at32-ssc.c delete mode 100644 sound/soc/at32/at32-ssc.h delete mode 100644 sound/soc/at32/playpaq_wm8510.c delete mode 100644 sound/soc/at91/Kconfig delete mode 100644 sound/soc/at91/Makefile delete mode 100644 sound/soc/at91/at91-pcm.c delete mode 100644 sound/soc/at91/at91-pcm.h delete mode 100644 sound/soc/at91/at91-ssc.c delete mode 100644 sound/soc/at91/at91-ssc.h diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig deleted file mode 100644 index b0765e8..0000000 --- a/sound/soc/at32/Kconfig +++ /dev/null @@ -1,34 +0,0 @@ -config SND_AT32_SOC - tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" - depends on AVR32 && SND_SOC - help - Say Y or M if you want to add support for codecs attached to - the AT32 SSC interface. You will also need to - to select the audio interfaces to support below. - - -config SND_AT32_SOC_SSC - tristate - - - -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_AT32_SOC && BOARD_PLAYPAQ - select SND_AT32_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - - - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile deleted file mode 100644 index c03e55e..0000000 --- a/sound/soc/at32/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT32 Platform Support -snd-soc-at32-objs := at32-pcm.o -snd-soc-at32-ssc-objs := at32-ssc.o - -obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o -obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o - -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c deleted file mode 100644 index c83584f..0000000 --- a/sound/soc/at32/at32-pcm.c +++ /dev/null @@ -1,492 +0,0 @@ -/* sound/soc/at32/at32-pcm.c - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-pcm.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "at32-pcm.h" - - - -/*--------------------------------------------------------------------------*\ - * Hardware definition -\*--------------------------------------------------------------------------*/ -/* TODO: These values were taken from the AT91 platform driver, check - * them against real values for AT32 - */ -static const struct snd_pcm_hardware at32_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - - .formats = SNDRV_PCM_FMTBIT_S16, - .period_bytes_min = 32, - .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - - - -/*--------------------------------------------------------------------------*\ - * Data types -\*--------------------------------------------------------------------------*/ -struct at32_runtime_data { - struct at32_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of DMA buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - - dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ - - /* Save PDC registers (for power management) */ - u32 pdc_xpr_save; - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - - - -/*--------------------------------------------------------------------------*\ - * Helper functions -\*--------------------------------------------------------------------------*/ -static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *dmabuf = &substream->dma_buffer; - size_t size = at32_pcm_hardware.buffer_bytes_max; - - dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; - dmabuf->dev.dev = pcm->card->dev; - dmabuf->private_data = NULL; - dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, - &dmabuf->addr, GFP_KERNEL); - pr_debug("at32_pcm: preallocate_dma_buffer: " - "area=%p, addr=%p, size=%ld\n", - (void *)dmabuf->area, (void *)dmabuf->addr, size); - - if (!dmabuf->area) - return -ENOMEM; - - dmabuf->bytes = size; - return 0; -} - - - -/*--------------------------------------------------------------------------*\ - * ISR -\*--------------------------------------------------------------------------*/ -static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - static int count; - - count++; - if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "underrun" : "overrun", params->name, ssc_sr, count); - - /* re-start the PDC */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - } - - - if (ssc_sr & params->mask->ssc_endx) { - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - - snd_pcm_period_elapsed(substream); -} - - - -/*--------------------------------------------------------------------------*\ - * PCM operations -\*--------------------------------------------------------------------------*/ -static int at32_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params - */ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at32_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - pr_debug("hw_params: DMA for %s initialized " - "(dma_bytes=%ld, period_size=%ld)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - - return 0; -} - - - -static int at32_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - - - -static int at32_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - ssc_writex(params->ssc->regs, SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - return 0; -} - - -static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - int ret = 0; - - pr_debug("at32_pcm_trigger: buffer_size = %ld, " - "dma_area = %p, dma_bytes = %ld\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - pr_debug("trigger: period_ptr=%lx, xpr=%x, " - "xcr=%d, xnpr=%x, xncr=%d\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); - - ssc_writex(params->ssc->regs, SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_enable); - - pr_debug("sr=%x, imr=%x\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); - break; /* SNDRV_PCM_TRIGGER_START */ - - - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - break; - - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - - - -static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - - return x; -} - - - -static int at32_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - -out: - return ret; -} - - - -static int at32_pcm_close(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - - -static int at32_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - - - -static struct snd_pcm_ops at32_pcm_ops = { - .open = at32_pcm_open, - .close = at32_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at32_pcm_hw_params, - .hw_free = at32_pcm_hw_free, - .prepare = at32_pcm_prepare, - .trigger = at32_pcm_trigger, - .pointer = at32_pcm_pointer, - .mmap = at32_pcm_mmap, -}; - - - -/*--------------------------------------------------------------------------*\ - * ASoC platform driver -\*--------------------------------------------------------------------------*/ -static u64 at32_pcm_dmamask = 0xffffffff; - -static int at32_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at32_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - - -out: - return ret; -} - - - -static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream == NULL) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - - - -#ifdef CONFIG_PM -static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Disable the PDC and save the PDC registers */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); - - return 0; -} - - - -static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else /* CONFIG_PM */ -# define at32_pcm_suspend NULL -# define at32_pcm_resume NULL -#endif /* CONFIG_PM */ - - - -struct snd_soc_platform at32_soc_platform = { - .name = "at32-audio", - .pcm_ops = &at32_pcm_ops, - .pcm_new = at32_pcm_new, - .pcm_free = at32_pcm_free_dma_buffers, - .suspend = at32_pcm_suspend, - .resume = at32_pcm_resume, -}; -EXPORT_SYMBOL_GPL(at32_soc_platform); - - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("Atmel AT32 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h deleted file mode 100644 index 2a52430..0000000 --- a/sound/soc/at32/at32-pcm.h +++ /dev/null @@ -1,79 +0,0 @@ -/* sound/soc/at32/at32-pcm.h - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_PCM_H -#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ - -#include <linux/atmel-ssc.h> - - -/* - * Registers and status bits that are required by the PCM driver - * TODO: Is ptcr really used? - */ -struct at32_pdc_regs { - u32 xpr; /* PDC RX/TX pointer */ - u32 xcr; /* PDC RX/TX counter */ - u32 xnpr; /* PDC next RX/TX pointer */ - u32 xncr; /* PDC next RX/TX counter */ - u32 ptcr; /* PDC transfer control */ -}; - - - -/* - * SSC mask info - */ -struct at32_ssc_mask { - u32 ssc_enable; /* SSC RX/TX enable */ - u32 ssc_disable; /* SSC RX/TX disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ - u32 pdc_enable; /* PDC RX/TX enable */ - u32 pdc_disable; /* PDC RX/TX disable */ -}; - - - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at32_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - struct ssc_device *ssc; /* SSC device for stream */ - struct at32_pdc_regs *pdc; /* PDC register info */ - struct at32_ssc_mask *mask; /* SSC mask info */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler) (u32, struct snd_pcm_substream *); -}; - - - -/* - * The AT32 ASoC platform driver - */ -extern struct snd_soc_platform at32_soc_platform; - - - -/* - * SSC register access (since ssc_writel() / ssc_readl() require literal name) - */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) - -#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c deleted file mode 100644 index 4ef6492..0000000 --- a/sound/soc/at32/at32-ssc.c +++ /dev/null @@ -1,849 +0,0 @@ -/* sound/soc/at32/at32-ssc.c - * ASoC platform driver for AT32 using SSC as DAI - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-ssc.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -/* #define DEBUG */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "at32-pcm.h" -#include "at32-ssc.h" - - - -/*-------------------------------------------------------------------------*\ - * Constants -\*-------------------------------------------------------------------------*/ -#define NUM_SSC_DEVICES 3 - -/* - * SSC direction masks - */ -#define SSC_DIR_MASK_UNUSED 0 -#define SSC_DIR_MASK_PLAYBACK 1 -#define SSC_DIR_MASK_CAPTURE 2 - -/* - * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These - * are expected to be used with SSC_BF - */ -/* START bit field values */ -#define SSC_START_CONTINUOUS 0 -#define SSC_START_TX_RX 1 -#define SSC_START_LOW_RF 2 -#define SSC_START_HIGH_RF 3 -#define SSC_START_FALLING_RF 4 -#define SSC_START_RISING_RF 5 -#define SSC_START_LEVEL_RF 6 -#define SSC_START_EDGE_RF 7 -#define SSS_START_COMPARE_0 8 - -/* CKI bit field values */ -#define SSC_CKI_FALLING 0 -#define SSC_CKI_RISING 1 - -/* CKO bit field values */ -#define SSC_CKO_NONE 0 -#define SSC_CKO_CONTINUOUS 1 -#define SSC_CKO_TRANSFER 2 - -/* CKS bit field values */ -#define SSC_CKS_DIV 0 -#define SSC_CKS_CLOCK 1 -#define SSC_CKS_PIN 2 - -/* FSEDGE bit field values */ -#define SSC_FSEDGE_POSITIVE 0 -#define SSC_FSEDGE_NEGATIVE 1 - -/* FSOS bit field values */ -#define SSC_FSOS_NONE 0 -#define SSC_FSOS_NEGATIVE 1 -#define SSC_FSOS_POSITIVE 2 -#define SSC_FSOS_LOW 3 -#define SSC_FSOS_HIGH 4 -#define SSC_FSOS_TOGGLE 5 - -#define START_DELAY 1 - - - -/*-------------------------------------------------------------------------*\ - * Module data -\*-------------------------------------------------------------------------*/ -/* - * SSC PDC registered required by the PCM DMA engine - */ -static struct at32_pdc_regs pdc_tx_reg = { - .xpr = SSC_PDC_TPR, - .xcr = SSC_PDC_TCR, - .xnpr = SSC_PDC_TNPR, - .xncr = SSC_PDC_TNCR, -}; - - - -static struct at32_pdc_regs pdc_rx_reg = { - .xpr = SSC_PDC_RPR, - .xcr = SSC_PDC_RCR, - .xnpr = SSC_PDC_RNPR, - .xncr = SSC_PDC_RNCR, -}; - - - -/* - * SSC and PDC status bits for transmit and receive - */ -static struct at32_ssc_mask ssc_tx_mask = { - .ssc_enable = SSC_BIT(CR_TXEN), - .ssc_disable = SSC_BIT(CR_TXDIS), - .ssc_endx = SSC_BIT(SR_ENDTX), - .ssc_endbuf = SSC_BIT(SR_TXBUFE), - .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), -}; - - - -static struct at32_ssc_mask ssc_rx_mask = { - .ssc_enable = SSC_BIT(CR_RXEN), - .ssc_disable = SSC_BIT(CR_RXDIS), - .ssc_endx = SSC_BIT(SR_ENDRX), - .ssc_endbuf = SSC_BIT(SR_RXBUFF), - .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), -}; - - - -/* - * DMA parameters for each SSC - */ -static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - { - { - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, -}; - - - -static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, -}; - - - - -/*-------------------------------------------------------------------------*\ - * ISR -\*-------------------------------------------------------------------------*/ -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt - * handler in the PCM driver. - */ -static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) -{ - struct at32_ssc_info *ssc_p = dev_id; - struct at32_pcm_dma_params *dma_params; - u32 ssc_sr; - u32 ssc_substream_mask; - int i; - - ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & - ssc_readl(ssc_p->ssc->regs, IMR)); - - /* - * Loop through substreams attached to this SSC. If a DMA-related - * interrupt occured on that substream, call the DMA interrupt - * handler function, if one has been registered in the dma_param - * structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if ((dma_params != NULL) && - (dma_params->dma_intr_handler != NULL)) { - ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); - if (ssc_sr & ssc_substream_mask) { - dma_params->dma_intr_handler(ssc_sr, - dma_params-> - substream); - } - } - } - - - return IRQ_HANDLED; -} - -/*-------------------------------------------------------------------------*\ - * DAI functions -\*-------------------------------------------------------------------------*/ -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at32_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - - - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at32_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - int dir_mask; - - dma_params = ssc_p->dma_params[substream->stream]; - - if (dma_params != NULL) { - ssc_writel(dma_params->ssc->regs, CR, - dma_params->mask->ssc_disable); - pr_debug("%s disabled SSC_SR=0x%08x\n", - (substream->stream ? "receiver" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - - dma_params->ssc = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[substream->stream] = NULL; - } - - - dir_mask = 1 << substream->stream; - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock */ - pr_debug("at32-ssc: Stopping user %d clock\n", - ssc_p->ssc->user); - clk_disable(ssc_p->ssc->clk); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc->irq, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - /* clear the SSC dividers */ - ssc_p->cmr_div = 0; - ssc_p->tcmr_period = 0; - ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - - - -/* - * Set the SSC system clock rate - */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* TODO: What the heck do I do here? */ - return 0; -} - - - -/* - * Record DAI format for use by hw_params() - */ -static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - - - -/* - * Record SSC clock dividers for use in hw_params() - */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT32_SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT32_SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT32_SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - - - -/* - * Configure the SSC - */ -static int at32_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at32_ssc_info *ssc_p = &ssc_info[id]; - struct at32_pcm_dma_params *dma_params; - int channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - - /* - * Currently, there is only one set of dma_params for each direction. - * If more are added, this code will have to be changed to select - * the proper set - */ - dma_params = &ssc_dma_params[id][substream->stream]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[substream->stream] = dma_params; - - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the PCM driver's hw_params() - * function. It should not be used for other purposes as it - * is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - - /* - * Determine sample size in bits and the PDC increment - */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - - case SNDRV_PCM_FORMAT_S16: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - - case SNDRV_PCM_FORMAT_S24: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - - case SNDRV_PCM_FORMAT_S32: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - - default: - pr_warning("at32-ssc: Unsupported PCM format %d", - params_format(params)); - return -EINVAL; - } - pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", - bits, dma_params->pdc_xfer_size, channels); - - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) - if (bits > 16) { - pr_warning("at32-ssc: " - "sample size %d is too large for I2S\n", - bits); - return -EINVAL; - } - - - /* - * Compute the SSC register settings - */ - switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRS clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, SSC_START_FALLING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(RFMR_FSLEN, bits - 1) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, SSC_START_FALLING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(TFMR_FSLEN, bits - 1) | - SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | - SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clock. - * - * The SSC transmit clock is obtained from the BCLK signal - * on the TK line, and the SSC receive clock is generated from - * the transmit clock. - * - * For single channel data, one sample is transferred on the - * falling edge of the LRC clock. For two channel data, one - * sample is transferred on both edges of the LRC clock. - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); - start_event = ((channels == 1) ? - SSC_START_FALLING_RF : SSC_START_EDGE_RF); - - rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, start_event) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, start_event) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_NONE) | - SSC_BF(TCMR_CKS, SSC_CKS_PIN)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, 1) | - SSC_BF(RCMR_START, SSC_START_RISING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, 1) | - SSC_BF(TCMR_START, SSC_START_RISING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_RISING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(TFMR_DATNB, channels - 1) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - pr_warning("at32-ssc: unsupported DAI format 0x%x\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", - rcmr, rfmr, tcmr, tfmr); - - - if (!ssc_p->initialized) { - /* enable peripheral clock */ - pr_debug("at32-ssc: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); - - ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, - ssc_p->name, ssc_p); - if (ret < 0) { - pr_warning("at32-ssc: request irq failed (%d)\n", ret); - pr_debug("at32-ssc: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - return ret; - } - - ssc_p->initialized = 1; - } - - /* Set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); - - /* set transmit clock mode and format */ - ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); - ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); - - pr_debug("at32-ssc: SSC initialized\n"); - return 0; -} - - - -static int at32_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - - dma_params = ssc_p->dma_params[substream->stream]; - - ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); - - return 0; -} - - - -#ifdef CONFIG_PM -static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive */ - ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); - - /* Save the current interrupt mask, then disable unmasked interrupts */ - ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); - ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); - ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); - ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); - ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); - ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); - - return 0; -} - - - -static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - u32 cr; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* restore SSC register settings */ - ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); - ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); - ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); - - /* re-enable interrupts */ - ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - - /* Re-enable recieve and transmit as appropriate */ - cr = 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; - ssc_writel(ssc_p->ssc->regs, CR, cr); - - return 0; -} -#else /* CONFIG_PM */ -# define at32_ssc_suspend NULL -# define at32_ssc_resume NULL -#endif /* CONFIG_PM */ - - -#define AT32_SSC_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - - -#define AT32_SSC_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) - - -struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { - { - .name = "at32-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[0], - }, - { - .name = "at32-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[1], - }, - { - .name = "at32-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[2], - }, -}; -EXPORT_SYMBOL_GPL(at32_ssc_dai); - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h deleted file mode 100644 index 3c052db..0000000 --- a/sound/soc/at32/at32-ssc.h +++ /dev/null @@ -1,59 +0,0 @@ -/* sound/soc/at32/at32-ssc.h - * ASoC SSC interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_SSC_H -#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ - -#include <linux/types.h> -#include <linux/atmel-ssc.h> - -#include "at32-pcm.h" - - - -struct at32_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - - - -struct at32_ssc_info { - char *name; - struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* true if SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at32_pcm_dma_params *dma_params[2]; - struct at32_ssc_state ssc_state; -}; - - -/* SSC divider ids */ -#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - - -extern struct snd_soc_dai at32_ssc_dai[]; - - - -#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c deleted file mode 100644 index 7e6560b..0000000 --- a/sound/soc/at32/playpaq_wm8510.c +++ /dev/null @@ -1,523 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/version.h> -#include <linux/kernel.h> -#include <linux/errno.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/at32ap700x.h> -#include <mach/portmux.h> - -#include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_3; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const char *intercon[][3] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_codec *codec) -{ - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } - - - /* always connected pins */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync(codec); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai = &at32_ssc_dai[0], - .codec_dai = &wm8510_dai, - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_machine snd_soc_machine_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - - - -static struct wm8510_setup_data playpaq_wm8510_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - - - -static struct snd_soc_device playpaq_wm8510_snd_devdata = { - .machine = &snd_soc_machine_playpaq, - .platform = &at32_soc_platform, - .codec_dev = &soc_codec_dev_wm8510, - .codec_data = &playpaq_wm8510_setup, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc = NULL; - - - /* - * Request SSC device - */ - ssc = ssc_request(0); - if (IS_ERR(ssc)) { - ret = PTR_ERR(ssc); - ssc = NULL; - goto err_ssc; - } - ssc_p->ssc = ssc; - - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - goto err_pll0; - } - if (clk_set_parent(_gclk0, _pll0)) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); - playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } -err_gclk0: - if (ssc != NULL) { - ssc_free(ssc); - ssc = NULL; - } -err_ssc: - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc; - - if (ssc_p != NULL) { - ssc = ssc_p->ssc; - if (ssc != NULL) - ssc_free(ssc); - ssc_p->ssc = NULL; - } - - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig deleted file mode 100644 index 85a8832..0000000 --- a/sound/soc/at91/Kconfig +++ /dev/null @@ -1,10 +0,0 @@ -config SND_AT91_SOC - tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 - help - Say Y or M if you want to add support for codecs attached to - the AT91 SSC interface. You will also need - to select the audio interfaces to support below. - -config SND_AT91_SOC_SSC - tristate diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile deleted file mode 100644 index b817f11..0000000 --- a/sound/soc/at91/Makefile +++ /dev/null @@ -1,6 +0,0 @@ -# AT91 Platform Support -snd-soc-at91-objs := at91-pcm.o -snd-soc-at91-ssc-objs := at91-ssc.o - -obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c deleted file mode 100644 index 7ab48bd..0000000 --- a/sound/soc/at91/at91-pcm.c +++ /dev/null @@ -1,434 +0,0 @@ -/* - * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "at91-pcm: " x) -#else -#define DBG(x...) -#endif - -static const struct snd_pcm_hardware at91_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - -struct at91_runtime_data { - struct at91_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - u32 pdc_xpr_save; /* PDC register save */ - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - -static void at91_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - static int count = 0; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - - printk(KERN_WARNING - "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, - prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - -static int at91_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at91_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - return 0; -} - -static int at91_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int at91_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - at91_ssc_write(params->ssc_base + AT91_SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; -} - -static int at91_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", - (unsigned long) prtd->period_ptr, - at91_ssc_read(params->ssc_base + params->pdc->xpr), - at91_ssc_read(params->ssc_base + params->pdc->xcr), - at91_ssc_read(params->ssc_base + params->pdc->xnpr), - at91_ssc_read(params->ssc_base + params->pdc->xncr)); - - at91_ssc_write(params->ssc_base + AT91_SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, - params->mask->pdc_enable); - - DBG("sr=%lx imr=%lx\n", - at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t at91_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - return x; -} - -static int at91_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int at91_pcm_close(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int at91_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -struct snd_pcm_ops at91_pcm_ops = { - .open = at91_pcm_open, - .close = at91_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at91_pcm_hw_params, - .hw_free = at91_pcm_hw_free, - .prepare = at91_pcm_prepare, - .trigger = at91_pcm_trigger, - .pointer = at91_pcm_pointer, - .mmap = at91_pcm_mmap, -}; - -static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = at91_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static u64 at91_pcm_dmamask = 0xffffffff; - -static int at91_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at91_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -#ifdef CONFIG_PM -static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr); - prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr); - prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr); - prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr); - - return 0; -} - -static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save); - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define at91_pcm_suspend NULL -#define at91_pcm_resume NULL -#endif - -struct snd_soc_platform at91_soc_platform = { - .name = "at91-audio", - .pcm_ops = &at91_pcm_ops, - .pcm_new = at91_pcm_new, - .pcm_free = at91_pcm_free_dma_buffers, - .suspend = at91_pcm_suspend, - .resume = at91_pcm_resume, -}; - -EXPORT_SYMBOL_GPL(at91_soc_platform); - -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("Atmel AT91 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h deleted file mode 100644 index e5aada2..0000000 --- a/sound/soc/at91/at91-pcm.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.h by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_PCM_H -#define _AT91_PCM_H - -#include <mach/hardware.h> - -struct at91_ssc_periph { - void __iomem *base; - u32 pid; -}; - -/* - * Registers and status bits that are required by the PCM driver. - */ -struct at91_pdc_regs { - unsigned int xpr; /* PDC recv/trans pointer */ - unsigned int xcr; /* PDC recv/trans counter */ - unsigned int xnpr; /* PDC next recv/trans pointer */ - unsigned int xncr; /* PDC next recv/trans counter */ - unsigned int ptcr; /* PDC transfer control */ -}; - -struct at91_ssc_mask { - u32 ssc_enable; /* SSC recv/trans enable */ - u32 ssc_disable; /* SSC recv/trans disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ - u32 pdc_enable; /* PDC recv/trans enable */ - u32 pdc_disable; /* PDC recv/trans disable */ -}; - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at91_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - void __iomem *ssc_base; /* SSC base address */ - struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */ - struct at91_ssc_mask *mask;/* SSC & PDC status bits */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler)(u32, struct snd_pcm_substream *); -}; - -extern struct snd_soc_platform at91_soc_platform; - -#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -#define at91_ssc_write(a,v) __raw_writel((v),(a)) - -#endif /* _AT91_PCM_H */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c deleted file mode 100644 index a5b1a79..0000000 --- a/sound/soc/at91/at91-ssc.c +++ /dev/null @@ -1,791 +0,0 @@ -/* - * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * - * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_pmc.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x) -#else -#define DBG(x...) -#endif - -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) -#define NUM_SSC_DEVICES 1 -#else -#define NUM_SSC_DEVICES 3 -#endif - - -/* - * SSC PDC registers required by the PCM DMA engine. - */ -static struct at91_pdc_regs pdc_tx_reg = { - .xpr = ATMEL_PDC_TPR, - .xcr = ATMEL_PDC_TCR, - .xnpr = ATMEL_PDC_TNPR, - .xncr = ATMEL_PDC_TNCR, -}; - -static struct at91_pdc_regs pdc_rx_reg = { - .xpr = ATMEL_PDC_RPR, - .xcr = ATMEL_PDC_RCR, - .xnpr = ATMEL_PDC_RNPR, - .xncr = ATMEL_PDC_RNCR, -}; - -/* - * SSC & PDC status bits for transmit and receive. - */ -static struct at91_ssc_mask ssc_tx_mask = { - .ssc_enable = AT91_SSC_TXEN, - .ssc_disable = AT91_SSC_TXDIS, - .ssc_endx = AT91_SSC_ENDTX, - .ssc_endbuf = AT91_SSC_TXBUFE, - .pdc_enable = ATMEL_PDC_TXTEN, - .pdc_disable = ATMEL_PDC_TXTDIS, -}; - -static struct at91_ssc_mask ssc_rx_mask = { - .ssc_enable = AT91_SSC_RXEN, - .ssc_disable = AT91_SSC_RXDIS, - .ssc_endx = AT91_SSC_ENDRX, - .ssc_endbuf = AT91_SSC_RXBUFF, - .pdc_enable = ATMEL_PDC_RXTEN, - .pdc_disable = ATMEL_PDC_RXTDIS, -}; - - -/* - * DMA parameters. - */ -static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - {{ - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#if NUM_SSC_DEVICES == 3 - {{ - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, - {{ - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#endif -}; - -struct at91_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - -static struct at91_ssc_info { - char *name; - struct at91_ssc_periph ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* 1=SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at91_pcm_dma_params *dma_params[2]; - struct at91_ssc_state ssc_state; - -} ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = 0, - .initialized = 0, - }, -#if NUM_SSC_DEVICES == 3 - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = 0, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = 0, - .initialized = 0, - }, -#endif -}; - -static unsigned int at91_ssc_sysclk; - -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA - * interrupt handler in the PCM driver. - */ -static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id) -{ - struct at91_ssc_info *ssc_p = dev_id; - struct at91_pcm_dma_params *dma_params; - u32 ssc_sr; - int i; - - ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR) - & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - - /* - * Loop through the substreams attached to this SSC. If - * a DMA-related interrupt occurred on that substream, call - * the DMA interrupt handler function, if one has been - * registered in the dma_params structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if (dma_params != NULL && dma_params->dma_intr_handler != NULL && - (ssc_sr & - (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) - - dma_params->dma_intr_handler(ssc_sr, dma_params->substream); - } - - return IRQ_HANDLED; -} - -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at91_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - DBG("ssc_startup: SSC_SR=0x%08lx\n", - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at91_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir, dir_mask; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - if (dma_params != NULL) { - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_disable); - DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - - dma_params->ssc_base = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[dir] = NULL; - } - - dir_mask = 1 << dir; - - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc.pid, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - -/* - * Record the SSC system clock rate. - */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* - * The only clock supplied to the SSC is the AT91 master clock, - * which is only used if the SSC is generating BCLK and/or - * LRC clocks. - */ - switch (clk_id) { - case AT91_SYSCLK_MCK: - at91_ssc_sysclk = freq; - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * Record the DAI format for use in hw_params(). - */ -static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - -/* - * Record SSC clock dividers for use in hw_params(). - */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT91SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value. - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT91SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT91SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -/* - * Configure the SSC. - */ -static int at91_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at91_ssc_info *ssc_p = &ssc_info[id]; - struct at91_pcm_dma_params *dma_params; - int dir, channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - /* - * Currently, there is only one set of dma params for - * each direction. If more are added, this code will - * have to be changed to select the proper set. - */ - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc_base = ssc_p->ssc.base; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - /* - * Determine sample size in bits and the PDC increment. - */ - switch(params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format\n"); - return -EINVAL; - } - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "at91-ssc: sample size %d is too large for I2S\n", bits); - return -EINVAL; - } - - /* - * Compute SSC register settings. - */ - switch (ssc_p->daifmt - & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is generated from the - * transmit clock. - * - * For single channel data, one sample is transferred on the falling - * edge of the LRC clock. For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = channels == 1 - ? AT91_SSC_START_FALLING_RF - : AT91_SSC_START_EDGE_RF; - - rcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - - - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); - - if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - DBG("Starting pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); - - /* Reset the SSC and its PDC registers */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0); - - if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt, - 0, ssc_p->name, ssc_p)) < 0) { - printk(KERN_WARNING "at91-ssc: request_irq failure\n"); - - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - return ret; - } - - ssc_p->initialized = 1; - } - - /* set SSC clock mode register */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr); - - /* set transmit clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr); - - DBG("hw_params: SSC initialized\n"); - return 0; -} - - -static int at91_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_enable); - - DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", - at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR)); - return 0; -} - - -#ifdef CONFIG_PM -static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive. */ - ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - AT91_SSC_TXDIS | AT91_SSC_RXDIS); - - /* Save the current interrupt mask, then disable unmasked interrupts. */ - ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR); - ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR); - ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR); - ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR); - ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR); - - return 0; -} - -static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); - - return 0; -} - -#else -#define at91_ssc_suspend NULL -#define at91_ssc_resume NULL -#endif - -#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - -#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { - { .name = "at91-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[0].ssc, - }, -#if NUM_SSC_DEVICES == 3 - { .name = "at91-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[1].ssc, - }, - { .name = "at91-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[2].ssc, - }, -#endif -}; - -EXPORT_SYMBOL_GPL(at91_ssc_dai); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino(a)endrelia.com, www.endrelia.com"); -MODULE_DESCRIPTION("AT91 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h deleted file mode 100644 index 6b7bf38..0000000 --- a/sound/soc/at91/at91-ssc.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Jan 9, 2007 - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_SSC_H -#define _AT91_SSC_H - -/* SSC system clock ids */ -#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ - -/* SSC divider ids */ -#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - -extern struct snd_soc_dai at91_ssc_dai[]; - -#endif /* _AT91_SSC_H */ - -- 1.5.3.7

2 1

[alsa-devel] [PATCH] ASoC: Create a common directory for AVR32 and ARM9 atmel boards.
by Sedji Gaouaou 01 Oct '08

01 Oct '08

Create a common directory for AVR32 and ARM9 atmel boards to unify atmel audio DAI. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 2 +- sound/soc/atmel/Kconfig | 43 ++ sound/soc/atmel/Makefile | 15 + sound/soc/atmel/atmel-pcm.c | 484 +++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.h | 86 ++++ sound/soc/atmel/atmel_ssc_dai.c | 803 ++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 ++++++ sound/soc/atmel/playpaq_wm8510.c | 523 +++++++++++++++++++++++++ sound/soc/atmel/sam9g20_wm8731.c | 329 ++++++++++++++++ 10 files changed, 2406 insertions(+), 3 deletions(-) create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile create mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/atmel/atmel-pcm.h create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h create mode 100644 sound/soc/atmel/playpaq_wm8510.c create mode 100644 sound/soc/atmel/sam9g20_wm8731.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..615ebf0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349..4d475c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..170d9da --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the AT91 SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..f54a7cc --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..6d317d2 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,484 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..2ed1575 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,803 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} +#if 0 +/* + * Record the SSC system clock rate. + */ +static int atmel_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* + * The only clock supplied to the SSC is the AT91 master clock, + * which is only used if the SSC is generating BCLK and/or + * LRC clocks. + */ + switch (clk_id) { + case ATMEL_SYSCLK_MCK: + atmel_ssc_sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} +#endif +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..a828746 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..7e6560b --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,523 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..4e191df --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,329 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); -- 1.5.3.7

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[PATCH] Create a common directory for AVR32 and ARM9 atmel boards.
by Sedji Gaouaou 01 Oct '08

01 Oct '08

Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 2 +- sound/soc/atmel/Kconfig | 43 ++ sound/soc/atmel/Makefile | 15 + sound/soc/atmel/atmel-pcm.c | 484 +++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.h | 86 ++++ sound/soc/atmel/atmel_ssc_dai.c | 803 ++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 ++++++ sound/soc/atmel/playpaq_wm8510.c | 523 +++++++++++++++++++++++++ sound/soc/atmel/sam9g20_wm8731.c | 329 ++++++++++++++++ 10 files changed, 2406 insertions(+), 3 deletions(-) create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile create mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/atmel/atmel-pcm.h create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h create mode 100644 sound/soc/atmel/playpaq_wm8510.c create mode 100644 sound/soc/atmel/sam9g20_wm8731.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..615ebf0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349..4d475c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..170d9da --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the AT91 SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..f54a7cc --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..6d317d2 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,484 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..2ed1575 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,803 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} +#if 0 +/* + * Record the SSC system clock rate. + */ +static int atmel_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* + * The only clock supplied to the SSC is the AT91 master clock, + * which is only used if the SSC is generating BCLK and/or + * LRC clocks. + */ + switch (clk_id) { + case ATMEL_SYSCLK_MCK: + atmel_ssc_sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} +#endif +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..a828746 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..7e6560b --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,523 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..4e191df --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,329 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); -- 1.5.3.7 --------------030900070503000803070901--

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[PATCH] Remove at91 and at32 directories
by Sedji Gaouaou 01 Oct '08

01 Oct '08

Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/at32/Kconfig | 34 -- sound/soc/at32/Makefile | 11 - sound/soc/at32/at32-pcm.c | 492 ----------------------- sound/soc/at32/at32-pcm.h | 79 ---- sound/soc/at32/at32-ssc.c | 849 --------------------------------------- sound/soc/at32/at32-ssc.h | 59 --- sound/soc/at32/playpaq_wm8510.c | 523 ------------------------ sound/soc/at91/Kconfig | 10 - sound/soc/at91/Makefile | 6 - sound/soc/at91/at91-pcm.c | 434 -------------------- sound/soc/at91/at91-pcm.h | 72 ---- sound/soc/at91/at91-ssc.c | 791 ------------------------------------ sound/soc/at91/at91-ssc.h | 27 -- 13 files changed, 0 insertions(+), 3387 deletions(-) delete mode 100644 sound/soc/at32/Kconfig delete mode 100644 sound/soc/at32/Makefile delete mode 100644 sound/soc/at32/at32-pcm.c delete mode 100644 sound/soc/at32/at32-pcm.h delete mode 100644 sound/soc/at32/at32-ssc.c delete mode 100644 sound/soc/at32/at32-ssc.h delete mode 100644 sound/soc/at32/playpaq_wm8510.c delete mode 100644 sound/soc/at91/Kconfig delete mode 100644 sound/soc/at91/Makefile delete mode 100644 sound/soc/at91/at91-pcm.c delete mode 100644 sound/soc/at91/at91-pcm.h delete mode 100644 sound/soc/at91/at91-ssc.c delete mode 100644 sound/soc/at91/at91-ssc.h diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig deleted file mode 100644 index b0765e8..0000000 --- a/sound/soc/at32/Kconfig +++ /dev/null @@ -1,34 +0,0 @@ -config SND_AT32_SOC - tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" - depends on AVR32 && SND_SOC - help - Say Y or M if you want to add support for codecs attached to - the AT32 SSC interface. You will also need to - to select the audio interfaces to support below. - - -config SND_AT32_SOC_SSC - tristate - - - -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_AT32_SOC && BOARD_PLAYPAQ - select SND_AT32_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - - - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile deleted file mode 100644 index c03e55e..0000000 --- a/sound/soc/at32/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT32 Platform Support -snd-soc-at32-objs := at32-pcm.o -snd-soc-at32-ssc-objs := at32-ssc.o - -obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o -obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o - -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c deleted file mode 100644 index c83584f..0000000 --- a/sound/soc/at32/at32-pcm.c +++ /dev/null @@ -1,492 +0,0 @@ -/* sound/soc/at32/at32-pcm.c - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-pcm.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "at32-pcm.h" - - - -/*--------------------------------------------------------------------------*\ - * Hardware definition -\*--------------------------------------------------------------------------*/ -/* TODO: These values were taken from the AT91 platform driver, check - * them against real values for AT32 - */ -static const struct snd_pcm_hardware at32_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - - .formats = SNDRV_PCM_FMTBIT_S16, - .period_bytes_min = 32, - .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - - - -/*--------------------------------------------------------------------------*\ - * Data types -\*--------------------------------------------------------------------------*/ -struct at32_runtime_data { - struct at32_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of DMA buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - - dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ - - /* Save PDC registers (for power management) */ - u32 pdc_xpr_save; - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - - - -/*--------------------------------------------------------------------------*\ - * Helper functions -\*--------------------------------------------------------------------------*/ -static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *dmabuf = &substream->dma_buffer; - size_t size = at32_pcm_hardware.buffer_bytes_max; - - dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; - dmabuf->dev.dev = pcm->card->dev; - dmabuf->private_data = NULL; - dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, - &dmabuf->addr, GFP_KERNEL); - pr_debug("at32_pcm: preallocate_dma_buffer: " - "area=%p, addr=%p, size=%ld\n", - (void *)dmabuf->area, (void *)dmabuf->addr, size); - - if (!dmabuf->area) - return -ENOMEM; - - dmabuf->bytes = size; - return 0; -} - - - -/*--------------------------------------------------------------------------*\ - * ISR -\*--------------------------------------------------------------------------*/ -static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - static int count; - - count++; - if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "underrun" : "overrun", params->name, ssc_sr, count); - - /* re-start the PDC */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - } - - - if (ssc_sr & params->mask->ssc_endx) { - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - - snd_pcm_period_elapsed(substream); -} - - - -/*--------------------------------------------------------------------------*\ - * PCM operations -\*--------------------------------------------------------------------------*/ -static int at32_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params - */ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at32_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - pr_debug("hw_params: DMA for %s initialized " - "(dma_bytes=%ld, period_size=%ld)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - - return 0; -} - - - -static int at32_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - - - -static int at32_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - ssc_writex(params->ssc->regs, SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - return 0; -} - - -static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - int ret = 0; - - pr_debug("at32_pcm_trigger: buffer_size = %ld, " - "dma_area = %p, dma_bytes = %ld\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - pr_debug("trigger: period_ptr=%lx, xpr=%x, " - "xcr=%d, xnpr=%x, xncr=%d\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); - - ssc_writex(params->ssc->regs, SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_enable); - - pr_debug("sr=%x, imr=%x\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); - break; /* SNDRV_PCM_TRIGGER_START */ - - - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - break; - - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - - - -static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - - return x; -} - - - -static int at32_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - -out: - return ret; -} - - - -static int at32_pcm_close(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - - -static int at32_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - - - -static struct snd_pcm_ops at32_pcm_ops = { - .open = at32_pcm_open, - .close = at32_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at32_pcm_hw_params, - .hw_free = at32_pcm_hw_free, - .prepare = at32_pcm_prepare, - .trigger = at32_pcm_trigger, - .pointer = at32_pcm_pointer, - .mmap = at32_pcm_mmap, -}; - - - -/*--------------------------------------------------------------------------*\ - * ASoC platform driver -\*--------------------------------------------------------------------------*/ -static u64 at32_pcm_dmamask = 0xffffffff; - -static int at32_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at32_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - - -out: - return ret; -} - - - -static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream == NULL) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - - - -#ifdef CONFIG_PM -static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Disable the PDC and save the PDC registers */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); - - return 0; -} - - - -static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else /* CONFIG_PM */ -# define at32_pcm_suspend NULL -# define at32_pcm_resume NULL -#endif /* CONFIG_PM */ - - - -struct snd_soc_platform at32_soc_platform = { - .name = "at32-audio", - .pcm_ops = &at32_pcm_ops, - .pcm_new = at32_pcm_new, - .pcm_free = at32_pcm_free_dma_buffers, - .suspend = at32_pcm_suspend, - .resume = at32_pcm_resume, -}; -EXPORT_SYMBOL_GPL(at32_soc_platform); - - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("Atmel AT32 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h deleted file mode 100644 index 2a52430..0000000 --- a/sound/soc/at32/at32-pcm.h +++ /dev/null @@ -1,79 +0,0 @@ -/* sound/soc/at32/at32-pcm.h - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_PCM_H -#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ - -#include <linux/atmel-ssc.h> - - -/* - * Registers and status bits that are required by the PCM driver - * TODO: Is ptcr really used? - */ -struct at32_pdc_regs { - u32 xpr; /* PDC RX/TX pointer */ - u32 xcr; /* PDC RX/TX counter */ - u32 xnpr; /* PDC next RX/TX pointer */ - u32 xncr; /* PDC next RX/TX counter */ - u32 ptcr; /* PDC transfer control */ -}; - - - -/* - * SSC mask info - */ -struct at32_ssc_mask { - u32 ssc_enable; /* SSC RX/TX enable */ - u32 ssc_disable; /* SSC RX/TX disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ - u32 pdc_enable; /* PDC RX/TX enable */ - u32 pdc_disable; /* PDC RX/TX disable */ -}; - - - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at32_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - struct ssc_device *ssc; /* SSC device for stream */ - struct at32_pdc_regs *pdc; /* PDC register info */ - struct at32_ssc_mask *mask; /* SSC mask info */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler) (u32, struct snd_pcm_substream *); -}; - - - -/* - * The AT32 ASoC platform driver - */ -extern struct snd_soc_platform at32_soc_platform; - - - -/* - * SSC register access (since ssc_writel() / ssc_readl() require literal name) - */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) - -#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c deleted file mode 100644 index 4ef6492..0000000 --- a/sound/soc/at32/at32-ssc.c +++ /dev/null @@ -1,849 +0,0 @@ -/* sound/soc/at32/at32-ssc.c - * ASoC platform driver for AT32 using SSC as DAI - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-ssc.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -/* #define DEBUG */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "at32-pcm.h" -#include "at32-ssc.h" - - - -/*-------------------------------------------------------------------------*\ - * Constants -\*-------------------------------------------------------------------------*/ -#define NUM_SSC_DEVICES 3 - -/* - * SSC direction masks - */ -#define SSC_DIR_MASK_UNUSED 0 -#define SSC_DIR_MASK_PLAYBACK 1 -#define SSC_DIR_MASK_CAPTURE 2 - -/* - * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These - * are expected to be used with SSC_BF - */ -/* START bit field values */ -#define SSC_START_CONTINUOUS 0 -#define SSC_START_TX_RX 1 -#define SSC_START_LOW_RF 2 -#define SSC_START_HIGH_RF 3 -#define SSC_START_FALLING_RF 4 -#define SSC_START_RISING_RF 5 -#define SSC_START_LEVEL_RF 6 -#define SSC_START_EDGE_RF 7 -#define SSS_START_COMPARE_0 8 - -/* CKI bit field values */ -#define SSC_CKI_FALLING 0 -#define SSC_CKI_RISING 1 - -/* CKO bit field values */ -#define SSC_CKO_NONE 0 -#define SSC_CKO_CONTINUOUS 1 -#define SSC_CKO_TRANSFER 2 - -/* CKS bit field values */ -#define SSC_CKS_DIV 0 -#define SSC_CKS_CLOCK 1 -#define SSC_CKS_PIN 2 - -/* FSEDGE bit field values */ -#define SSC_FSEDGE_POSITIVE 0 -#define SSC_FSEDGE_NEGATIVE 1 - -/* FSOS bit field values */ -#define SSC_FSOS_NONE 0 -#define SSC_FSOS_NEGATIVE 1 -#define SSC_FSOS_POSITIVE 2 -#define SSC_FSOS_LOW 3 -#define SSC_FSOS_HIGH 4 -#define SSC_FSOS_TOGGLE 5 - -#define START_DELAY 1 - - - -/*-------------------------------------------------------------------------*\ - * Module data -\*-------------------------------------------------------------------------*/ -/* - * SSC PDC registered required by the PCM DMA engine - */ -static struct at32_pdc_regs pdc_tx_reg = { - .xpr = SSC_PDC_TPR, - .xcr = SSC_PDC_TCR, - .xnpr = SSC_PDC_TNPR, - .xncr = SSC_PDC_TNCR, -}; - - - -static struct at32_pdc_regs pdc_rx_reg = { - .xpr = SSC_PDC_RPR, - .xcr = SSC_PDC_RCR, - .xnpr = SSC_PDC_RNPR, - .xncr = SSC_PDC_RNCR, -}; - - - -/* - * SSC and PDC status bits for transmit and receive - */ -static struct at32_ssc_mask ssc_tx_mask = { - .ssc_enable = SSC_BIT(CR_TXEN), - .ssc_disable = SSC_BIT(CR_TXDIS), - .ssc_endx = SSC_BIT(SR_ENDTX), - .ssc_endbuf = SSC_BIT(SR_TXBUFE), - .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), -}; - - - -static struct at32_ssc_mask ssc_rx_mask = { - .ssc_enable = SSC_BIT(CR_RXEN), - .ssc_disable = SSC_BIT(CR_RXDIS), - .ssc_endx = SSC_BIT(SR_ENDRX), - .ssc_endbuf = SSC_BIT(SR_RXBUFF), - .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), -}; - - - -/* - * DMA parameters for each SSC - */ -static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - { - { - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, -}; - - - -static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, -}; - - - - -/*-------------------------------------------------------------------------*\ - * ISR -\*-------------------------------------------------------------------------*/ -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt - * handler in the PCM driver. - */ -static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) -{ - struct at32_ssc_info *ssc_p = dev_id; - struct at32_pcm_dma_params *dma_params; - u32 ssc_sr; - u32 ssc_substream_mask; - int i; - - ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & - ssc_readl(ssc_p->ssc->regs, IMR)); - - /* - * Loop through substreams attached to this SSC. If a DMA-related - * interrupt occured on that substream, call the DMA interrupt - * handler function, if one has been registered in the dma_param - * structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if ((dma_params != NULL) && - (dma_params->dma_intr_handler != NULL)) { - ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); - if (ssc_sr & ssc_substream_mask) { - dma_params->dma_intr_handler(ssc_sr, - dma_params-> - substream); - } - } - } - - - return IRQ_HANDLED; -} - -/*-------------------------------------------------------------------------*\ - * DAI functions -\*-------------------------------------------------------------------------*/ -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at32_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - - - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at32_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - int dir_mask; - - dma_params = ssc_p->dma_params[substream->stream]; - - if (dma_params != NULL) { - ssc_writel(dma_params->ssc->regs, CR, - dma_params->mask->ssc_disable); - pr_debug("%s disabled SSC_SR=0x%08x\n", - (substream->stream ? "receiver" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - - dma_params->ssc = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[substream->stream] = NULL; - } - - - dir_mask = 1 << substream->stream; - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock */ - pr_debug("at32-ssc: Stopping user %d clock\n", - ssc_p->ssc->user); - clk_disable(ssc_p->ssc->clk); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc->irq, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - /* clear the SSC dividers */ - ssc_p->cmr_div = 0; - ssc_p->tcmr_period = 0; - ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - - - -/* - * Set the SSC system clock rate - */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* TODO: What the heck do I do here? */ - return 0; -} - - - -/* - * Record DAI format for use by hw_params() - */ -static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - - - -/* - * Record SSC clock dividers for use in hw_params() - */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT32_SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT32_SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT32_SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - - - -/* - * Configure the SSC - */ -static int at32_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at32_ssc_info *ssc_p = &ssc_info[id]; - struct at32_pcm_dma_params *dma_params; - int channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - - /* - * Currently, there is only one set of dma_params for each direction. - * If more are added, this code will have to be changed to select - * the proper set - */ - dma_params = &ssc_dma_params[id][substream->stream]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[substream->stream] = dma_params; - - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the PCM driver's hw_params() - * function. It should not be used for other purposes as it - * is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - - /* - * Determine sample size in bits and the PDC increment - */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - - case SNDRV_PCM_FORMAT_S16: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - - case SNDRV_PCM_FORMAT_S24: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - - case SNDRV_PCM_FORMAT_S32: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - - default: - pr_warning("at32-ssc: Unsupported PCM format %d", - params_format(params)); - return -EINVAL; - } - pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", - bits, dma_params->pdc_xfer_size, channels); - - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) - if (bits > 16) { - pr_warning("at32-ssc: " - "sample size %d is too large for I2S\n", - bits); - return -EINVAL; - } - - - /* - * Compute the SSC register settings - */ - switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRS clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, SSC_START_FALLING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(RFMR_FSLEN, bits - 1) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, SSC_START_FALLING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(TFMR_FSLEN, bits - 1) | - SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | - SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clock. - * - * The SSC transmit clock is obtained from the BCLK signal - * on the TK line, and the SSC receive clock is generated from - * the transmit clock. - * - * For single channel data, one sample is transferred on the - * falling edge of the LRC clock. For two channel data, one - * sample is transferred on both edges of the LRC clock. - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); - start_event = ((channels == 1) ? - SSC_START_FALLING_RF : SSC_START_EDGE_RF); - - rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, start_event) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, start_event) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_NONE) | - SSC_BF(TCMR_CKS, SSC_CKS_PIN)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, 1) | - SSC_BF(RCMR_START, SSC_START_RISING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, 1) | - SSC_BF(TCMR_START, SSC_START_RISING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_RISING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(TFMR_DATNB, channels - 1) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - pr_warning("at32-ssc: unsupported DAI format 0x%x\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", - rcmr, rfmr, tcmr, tfmr); - - - if (!ssc_p->initialized) { - /* enable peripheral clock */ - pr_debug("at32-ssc: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); - - ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, - ssc_p->name, ssc_p); - if (ret < 0) { - pr_warning("at32-ssc: request irq failed (%d)\n", ret); - pr_debug("at32-ssc: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - return ret; - } - - ssc_p->initialized = 1; - } - - /* Set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); - - /* set transmit clock mode and format */ - ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); - ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); - - pr_debug("at32-ssc: SSC initialized\n"); - return 0; -} - - - -static int at32_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - - dma_params = ssc_p->dma_params[substream->stream]; - - ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); - - return 0; -} - - - -#ifdef CONFIG_PM -static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive */ - ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); - - /* Save the current interrupt mask, then disable unmasked interrupts */ - ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); - ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); - ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); - ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); - ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); - ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); - - return 0; -} - - - -static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - u32 cr; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* restore SSC register settings */ - ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); - ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); - ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); - - /* re-enable interrupts */ - ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - - /* Re-enable recieve and transmit as appropriate */ - cr = 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; - ssc_writel(ssc_p->ssc->regs, CR, cr); - - return 0; -} -#else /* CONFIG_PM */ -# define at32_ssc_suspend NULL -# define at32_ssc_resume NULL -#endif /* CONFIG_PM */ - - -#define AT32_SSC_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - - -#define AT32_SSC_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) - - -struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { - { - .name = "at32-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[0], - }, - { - .name = "at32-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[1], - }, - { - .name = "at32-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[2], - }, -}; -EXPORT_SYMBOL_GPL(at32_ssc_dai); - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h deleted file mode 100644 index 3c052db..0000000 --- a/sound/soc/at32/at32-ssc.h +++ /dev/null @@ -1,59 +0,0 @@ -/* sound/soc/at32/at32-ssc.h - * ASoC SSC interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_SSC_H -#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ - -#include <linux/types.h> -#include <linux/atmel-ssc.h> - -#include "at32-pcm.h" - - - -struct at32_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - - - -struct at32_ssc_info { - char *name; - struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* true if SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at32_pcm_dma_params *dma_params[2]; - struct at32_ssc_state ssc_state; -}; - - -/* SSC divider ids */ -#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - - -extern struct snd_soc_dai at32_ssc_dai[]; - - - -#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c deleted file mode 100644 index 7e6560b..0000000 --- a/sound/soc/at32/playpaq_wm8510.c +++ /dev/null @@ -1,523 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/version.h> -#include <linux/kernel.h> -#include <linux/errno.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/at32ap700x.h> -#include <mach/portmux.h> - -#include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_3; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const char *intercon[][3] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_codec *codec) -{ - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } - - - /* always connected pins */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync(codec); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai = &at32_ssc_dai[0], - .codec_dai = &wm8510_dai, - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_machine snd_soc_machine_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - - - -static struct wm8510_setup_data playpaq_wm8510_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - - - -static struct snd_soc_device playpaq_wm8510_snd_devdata = { - .machine = &snd_soc_machine_playpaq, - .platform = &at32_soc_platform, - .codec_dev = &soc_codec_dev_wm8510, - .codec_data = &playpaq_wm8510_setup, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc = NULL; - - - /* - * Request SSC device - */ - ssc = ssc_request(0); - if (IS_ERR(ssc)) { - ret = PTR_ERR(ssc); - ssc = NULL; - goto err_ssc; - } - ssc_p->ssc = ssc; - - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - goto err_pll0; - } - if (clk_set_parent(_gclk0, _pll0)) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); - playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } -err_gclk0: - if (ssc != NULL) { - ssc_free(ssc); - ssc = NULL; - } -err_ssc: - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc; - - if (ssc_p != NULL) { - ssc = ssc_p->ssc; - if (ssc != NULL) - ssc_free(ssc); - ssc_p->ssc = NULL; - } - - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig deleted file mode 100644 index 85a8832..0000000 --- a/sound/soc/at91/Kconfig +++ /dev/null @@ -1,10 +0,0 @@ -config SND_AT91_SOC - tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 - help - Say Y or M if you want to add support for codecs attached to - the AT91 SSC interface. You will also need - to select the audio interfaces to support below. - -config SND_AT91_SOC_SSC - tristate diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile deleted file mode 100644 index b817f11..0000000 --- a/sound/soc/at91/Makefile +++ /dev/null @@ -1,6 +0,0 @@ -# AT91 Platform Support -snd-soc-at91-objs := at91-pcm.o -snd-soc-at91-ssc-objs := at91-ssc.o - -obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c deleted file mode 100644 index 7ab48bd..0000000 --- a/sound/soc/at91/at91-pcm.c +++ /dev/null @@ -1,434 +0,0 @@ -/* - * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "at91-pcm: " x) -#else -#define DBG(x...) -#endif - -static const struct snd_pcm_hardware at91_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - -struct at91_runtime_data { - struct at91_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - u32 pdc_xpr_save; /* PDC register save */ - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - -static void at91_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - static int count = 0; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - - printk(KERN_WARNING - "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, - prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - -static int at91_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at91_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - return 0; -} - -static int at91_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int at91_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - at91_ssc_write(params->ssc_base + AT91_SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; -} - -static int at91_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", - (unsigned long) prtd->period_ptr, - at91_ssc_read(params->ssc_base + params->pdc->xpr), - at91_ssc_read(params->ssc_base + params->pdc->xcr), - at91_ssc_read(params->ssc_base + params->pdc->xnpr), - at91_ssc_read(params->ssc_base + params->pdc->xncr)); - - at91_ssc_write(params->ssc_base + AT91_SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, - params->mask->pdc_enable); - - DBG("sr=%lx imr=%lx\n", - at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t at91_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - return x; -} - -static int at91_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int at91_pcm_close(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int at91_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -struct snd_pcm_ops at91_pcm_ops = { - .open = at91_pcm_open, - .close = at91_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at91_pcm_hw_params, - .hw_free = at91_pcm_hw_free, - .prepare = at91_pcm_prepare, - .trigger = at91_pcm_trigger, - .pointer = at91_pcm_pointer, - .mmap = at91_pcm_mmap, -}; - -static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = at91_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static u64 at91_pcm_dmamask = 0xffffffff; - -static int at91_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at91_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -#ifdef CONFIG_PM -static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr); - prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr); - prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr); - prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr); - - return 0; -} - -static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save); - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define at91_pcm_suspend NULL -#define at91_pcm_resume NULL -#endif - -struct snd_soc_platform at91_soc_platform = { - .name = "at91-audio", - .pcm_ops = &at91_pcm_ops, - .pcm_new = at91_pcm_new, - .pcm_free = at91_pcm_free_dma_buffers, - .suspend = at91_pcm_suspend, - .resume = at91_pcm_resume, -}; - -EXPORT_SYMBOL_GPL(at91_soc_platform); - -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("Atmel AT91 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h deleted file mode 100644 index e5aada2..0000000 --- a/sound/soc/at91/at91-pcm.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.h by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_PCM_H -#define _AT91_PCM_H - -#include <mach/hardware.h> - -struct at91_ssc_periph { - void __iomem *base; - u32 pid; -}; - -/* - * Registers and status bits that are required by the PCM driver. - */ -struct at91_pdc_regs { - unsigned int xpr; /* PDC recv/trans pointer */ - unsigned int xcr; /* PDC recv/trans counter */ - unsigned int xnpr; /* PDC next recv/trans pointer */ - unsigned int xncr; /* PDC next recv/trans counter */ - unsigned int ptcr; /* PDC transfer control */ -}; - -struct at91_ssc_mask { - u32 ssc_enable; /* SSC recv/trans enable */ - u32 ssc_disable; /* SSC recv/trans disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ - u32 pdc_enable; /* PDC recv/trans enable */ - u32 pdc_disable; /* PDC recv/trans disable */ -}; - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at91_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - void __iomem *ssc_base; /* SSC base address */ - struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */ - struct at91_ssc_mask *mask;/* SSC & PDC status bits */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler)(u32, struct snd_pcm_substream *); -}; - -extern struct snd_soc_platform at91_soc_platform; - -#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -#define at91_ssc_write(a,v) __raw_writel((v),(a)) - -#endif /* _AT91_PCM_H */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c deleted file mode 100644 index a5b1a79..0000000 --- a/sound/soc/at91/at91-ssc.c +++ /dev/null @@ -1,791 +0,0 @@ -/* - * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * - * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_pmc.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x) -#else -#define DBG(x...) -#endif - -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) -#define NUM_SSC_DEVICES 1 -#else -#define NUM_SSC_DEVICES 3 -#endif - - -/* - * SSC PDC registers required by the PCM DMA engine. - */ -static struct at91_pdc_regs pdc_tx_reg = { - .xpr = ATMEL_PDC_TPR, - .xcr = ATMEL_PDC_TCR, - .xnpr = ATMEL_PDC_TNPR, - .xncr = ATMEL_PDC_TNCR, -}; - -static struct at91_pdc_regs pdc_rx_reg = { - .xpr = ATMEL_PDC_RPR, - .xcr = ATMEL_PDC_RCR, - .xnpr = ATMEL_PDC_RNPR, - .xncr = ATMEL_PDC_RNCR, -}; - -/* - * SSC & PDC status bits for transmit and receive. - */ -static struct at91_ssc_mask ssc_tx_mask = { - .ssc_enable = AT91_SSC_TXEN, - .ssc_disable = AT91_SSC_TXDIS, - .ssc_endx = AT91_SSC_ENDTX, - .ssc_endbuf = AT91_SSC_TXBUFE, - .pdc_enable = ATMEL_PDC_TXTEN, - .pdc_disable = ATMEL_PDC_TXTDIS, -}; - -static struct at91_ssc_mask ssc_rx_mask = { - .ssc_enable = AT91_SSC_RXEN, - .ssc_disable = AT91_SSC_RXDIS, - .ssc_endx = AT91_SSC_ENDRX, - .ssc_endbuf = AT91_SSC_RXBUFF, - .pdc_enable = ATMEL_PDC_RXTEN, - .pdc_disable = ATMEL_PDC_RXTDIS, -}; - - -/* - * DMA parameters. - */ -static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - {{ - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#if NUM_SSC_DEVICES == 3 - {{ - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, - {{ - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#endif -}; - -struct at91_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - -static struct at91_ssc_info { - char *name; - struct at91_ssc_periph ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* 1=SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at91_pcm_dma_params *dma_params[2]; - struct at91_ssc_state ssc_state; - -} ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = 0, - .initialized = 0, - }, -#if NUM_SSC_DEVICES == 3 - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = 0, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = 0, - .initialized = 0, - }, -#endif -}; - -static unsigned int at91_ssc_sysclk; - -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA - * interrupt handler in the PCM driver. - */ -static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id) -{ - struct at91_ssc_info *ssc_p = dev_id; - struct at91_pcm_dma_params *dma_params; - u32 ssc_sr; - int i; - - ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR) - & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - - /* - * Loop through the substreams attached to this SSC. If - * a DMA-related interrupt occurred on that substream, call - * the DMA interrupt handler function, if one has been - * registered in the dma_params structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if (dma_params != NULL && dma_params->dma_intr_handler != NULL && - (ssc_sr & - (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) - - dma_params->dma_intr_handler(ssc_sr, dma_params->substream); - } - - return IRQ_HANDLED; -} - -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at91_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - DBG("ssc_startup: SSC_SR=0x%08lx\n", - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at91_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir, dir_mask; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - if (dma_params != NULL) { - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_disable); - DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - - dma_params->ssc_base = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[dir] = NULL; - } - - dir_mask = 1 << dir; - - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc.pid, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - -/* - * Record the SSC system clock rate. - */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* - * The only clock supplied to the SSC is the AT91 master clock, - * which is only used if the SSC is generating BCLK and/or - * LRC clocks. - */ - switch (clk_id) { - case AT91_SYSCLK_MCK: - at91_ssc_sysclk = freq; - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * Record the DAI format for use in hw_params(). - */ -static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - -/* - * Record SSC clock dividers for use in hw_params(). - */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT91SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value. - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT91SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT91SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -/* - * Configure the SSC. - */ -static int at91_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at91_ssc_info *ssc_p = &ssc_info[id]; - struct at91_pcm_dma_params *dma_params; - int dir, channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - /* - * Currently, there is only one set of dma params for - * each direction. If more are added, this code will - * have to be changed to select the proper set. - */ - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc_base = ssc_p->ssc.base; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - /* - * Determine sample size in bits and the PDC increment. - */ - switch(params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format\n"); - return -EINVAL; - } - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "at91-ssc: sample size %d is too large for I2S\n", bits); - return -EINVAL; - } - - /* - * Compute SSC register settings. - */ - switch (ssc_p->daifmt - & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is generated from the - * transmit clock. - * - * For single channel data, one sample is transferred on the falling - * edge of the LRC clock. For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = channels == 1 - ? AT91_SSC_START_FALLING_RF - : AT91_SSC_START_EDGE_RF; - - rcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - - - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); - - if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - DBG("Starting pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); - - /* Reset the SSC and its PDC registers */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0); - - if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt, - 0, ssc_p->name, ssc_p)) < 0) { - printk(KERN_WARNING "at91-ssc: request_irq failure\n"); - - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - return ret; - } - - ssc_p->initialized = 1; - } - - /* set SSC clock mode register */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr); - - /* set transmit clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr); - - DBG("hw_params: SSC initialized\n"); - return 0; -} - - -static int at91_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_enable); - - DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", - at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR)); - return 0; -} - - -#ifdef CONFIG_PM -static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive. */ - ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - AT91_SSC_TXDIS | AT91_SSC_RXDIS); - - /* Save the current interrupt mask, then disable unmasked interrupts. */ - ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR); - ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR); - ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR); - ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR); - ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR); - - return 0; -} - -static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); - - return 0; -} - -#else -#define at91_ssc_suspend NULL -#define at91_ssc_resume NULL -#endif - -#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - -#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { - { .name = "at91-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[0].ssc, - }, -#if NUM_SSC_DEVICES == 3 - { .name = "at91-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[1].ssc, - }, - { .name = "at91-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[2].ssc, - }, -#endif -}; - -EXPORT_SYMBOL_GPL(at91_ssc_dai); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino(a)endrelia.com, www.endrelia.com"); -MODULE_DESCRIPTION("AT91 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h deleted file mode 100644 index 6b7bf38..0000000 --- a/sound/soc/at91/at91-ssc.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Jan 9, 2007 - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_SSC_H -#define _AT91_SSC_H - -/* SSC system clock ids */ -#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ - -/* SSC divider ids */ -#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - -extern struct snd_soc_dai at91_ssc_dai[]; - -#endif /* _AT91_SSC_H */ - -- 1.5.3.7 --------------060703030706060207010806--

1 0

[alsa-devel] [PATCH] ASoC: Remove references to Endrelia ETI-B1 board
by Frank Mandarino 01 Oct '08

01 Oct '08

See attached. -- Frank Mandarino fmandarino(a)endrelia.com Endrelia Technologies Inc. Toronto, Ontario, Canada The ASoC machine drivers for this board were only provided as examples for the new AT91 ASoC platform driver. Since the ETI-B1 board is proprietary and there are other AT91 ASoC machine drivers available, it makes sense to remove these drivers. Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> --- sound/soc/at91/Kconfig | 27 --- sound/soc/at91/Makefile | 7 - sound/soc/at91/eti_b1_bluecore.c | 284 ------------------------------- sound/soc/at91/eti_b1_wm8731.c | 350 -------------------------------------- 4 files changed, 0 insertions(+), 668 deletions(-) diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 5bdd98f..85a8832 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -8,30 +8,3 @@ config SND_AT91_SOC config SND_AT91_SOC_SSC tristate - -config SND_AT91_SOC_ETI_B1_WM8731 - tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on WM8731-based - Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. - -config SND_AT91_SOC_ETI_SLAVE - bool "Run codec in slave Mode on Endrelia boards" - depends on SND_AT91_SOC_ETI_B1_WM8731 - default n - help - Say Y if you want to run with the AT91 SSC generating the BCLK - and LRC signals on Endrelia boards. - -config SND_AT91_SOC_ETI_B1_BLUECORE - tristate "SoC Audio support for BlueCore PCM on Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - help - Say Y if you want to add support for SoC audio on the CSR BlueCore - Bluetooth PCM interface on Endrelia Technologies Inc ETI-B1 or - ETI-C1 boards. - diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile index 0003740..b817f11 100644 --- a/sound/soc/at91/Makefile +++ b/sound/soc/at91/Makefile @@ -4,10 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o - -# AT91 Machine Support -snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o -snd-soc-eti-b1-bluecore-objs := eti_b1_bluecore.o - -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_BLUECORE) += snd-soc-eti-b1-bluecore.o diff --git a/sound/soc/at91/eti_b1_bluecore.c b/sound/soc/at91/eti_b1_bluecore.c deleted file mode 100644 index 00e1dfc..0000000 --- a/sound/soc/at91/eti_b1_bluecore.c +++ /dev/null @@ -1,284 +0,0 @@ -/* - * eti_b1_bluecore -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board - * - * ASoC Codec and Machine driver for CSR BlueCore PCM interface. - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 16, 2007 - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/version.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/driver.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/arch/hardware.h> -#include <asm/arch/at91_pio.h> -#include <asm/arch/gpio.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_bluecore: " x) -#else -#define DBG(x...) -#endif - - -/* - * CSR BlueCore PCM interface codec driver. - */ -#define BLUECORE_VERSION "0.1" - -#define BLUECORE_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) - -static struct snd_soc_codec_dai bluecore_dai = { - .name = "BlueCore", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = BLUECORE_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = BLUECORE_FORMATS,}, -}; - -static int bluecore_soc_probe(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec; - int ret = 0; - - printk(KERN_INFO "bluecore: BlueCore PCM SoC Audio %s\n", BLUECORE_VERSION); - - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) - return -ENOMEM; - codec = socdev->codec; - mutex_init(&codec->mutex); - - codec->name = "BlueCore"; - codec->owner = THIS_MODULE; - codec->dai = &bluecore_dai; - codec->num_dai = 1; - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if(ret < 0) - goto err; - - ret = snd_soc_register_card(socdev); - if (ret < 0) - goto bus_err; - return 0; - -bus_err: - snd_soc_free_pcms(socdev); - -err: - kfree(socdev->codec); - socdev->codec = NULL; - return ret; -} - -static int bluecore_soc_remove(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; - - if(codec == NULL) - return 0; - - snd_soc_free_pcms(socdev); - kfree(socdev->codec); - - return 0; -} - -static struct snd_soc_codec_device soc_codec_dev_bluecore = { - .probe = bluecore_soc_probe, - .remove = bluecore_soc_remove, -}; - - -/* - * Endrelia ETI_B1 BlueCore Machine driver. - */ -#define AT91_PIO_TF2 (1 << (AT91_PIN_PB12 - PIN_BASE) % 32) -#define AT91_PIO_TK2 (1 << (AT91_PIN_PB13 - PIN_BASE) % 32) -#define AT91_PIO_TD2 (1 << (AT91_PIN_PB14 - PIN_BASE) % 32) -#define AT91_PIO_RD2 (1 << (AT91_PIN_PB15 - PIN_BASE) % 32) -#define AT91_PIO_RK2 (1 << (AT91_PIN_PB16 - PIN_BASE) % 32) -#define AT91_PIO_RF2 (1 << (AT91_PIN_PB17 - PIN_BASE) % 32) - - -/* - * CSR BlueCore PCM interface machine driver operations. - * - * Only hw_params() is required, and only the CPU_DAI needs to be - * initiailzed, as there is no real CODEC DAI. The CODEC DAI - * is controlled by BlueZ BCCMD and SCO operations. - */ -static int eti_b1_bcore_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - int cmr_div, period; - - /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The BlueCore PCM interface only operates at 8kHz, and - * the PCM_CLK signal must be less than 2048kHz. - */ - cmr_div = 75; /* PCM_CLK = 60MHz/(2*75) = 400kHz */ - period = 24; /* PCM_SYNC = PCM_CLK/(2*(24+1)) = 8000Hz */ - - /* set the MCK divider for PCM_CLK */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - /* set the BCLK divider for DACLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, period); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops eti_b1_bcore_ops = { - .hw_params = eti_b1_bcore_hw_params, -}; - -static struct snd_soc_dai_link eti_b1_bcore_dai = { - .name = "BlueCore PCM", - .stream_name = "BlueCore", - .cpu_dai = &at91_ssc_dai[2], - .codec_dai = &bluecore_dai, - .ops = &eti_b1_bcore_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1_bcore = { - .name = "ETI_B1_BLUECORE", - .dai_link = &eti_b1_bcore_dai, - .num_links = 1, -}; - -static struct snd_soc_device eti_b1_bcore_snd_devdata = { - .machine = &snd_soc_machine_eti_b1_bcore, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_bluecore, -}; - -static struct platform_device *eti_b1_bcore_snd_device; - -static int __init eti_b1_bcore_init(void) -{ - int ret; - u32 ssc_pio_lines; - struct at91_ssc_periph *ssc = eti_b1_bcore_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC2, SZ_16K, "soc-audio-2")) { - DBG("SSC2 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC2, SZ_16K); - if (!ssc->base) { - DBG("SSC2 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC2; - - eti_b1_bcore_snd_device = platform_device_alloc("soc-audio", 2); - if (!eti_b1_bcore_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_bcore_snd_device, &eti_b1_bcore_snd_devdata); - eti_b1_bcore_snd_devdata.dev = &eti_b1_bcore_snd_device->dev; - - ret = platform_device_add(eti_b1_bcore_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_bcore_snd_device); - goto fail_io_unmap; - } - - ssc_pio_lines = AT91_PIO_TF2 | AT91_PIO_TK2 | AT91_PIO_TD2 - | AT91_PIO_RD2 /* | AT91_PIO_RK1 | AT91_PIO_RF1 */; - - /* Reset all PIO registers and assign lines to peripheral A */ - at91_sys_write(AT91_PIOB + PIO_PDR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_ODR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_IDR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_ASR, ssc_pio_lines); - at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines); - - printk(KERN_INFO "eti_b1_bcore: BlueCore PCM in Slave Mode\n"); - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC2, SZ_16K); - return ret; -} - -static void __exit eti_b1_bcore_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_bcore_dai.cpu_dai->private_data; - - platform_device_unregister(eti_b1_bcore_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC2, SZ_16K); -} - -module_init(eti_b1_bcore_init); -module_exit(eti_b1_bcore_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-BlueCore"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c deleted file mode 100644 index cdc388f..0000000 --- a/sound/soc/at91/eti_b1_wm8731.c +++ /dev/null @@ -1,350 +0,0 @@ -/* - * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 29, 2006 - * - * Based on corgi.c by: - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - * Richard Purdie <richard(a)openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/version.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/hardware.h> -#include <mach/gpio.h> - -#include "../codecs/wm8731.h" -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) -#else -#define DBG(x...) -#endif - -static struct clk *pck1_clk; -static struct clk *pllb_clk; - - -static int eti_b1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* cpu clock is the AT91 master clock sent to the SSC */ - ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, - 60000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Start PCK1 clock. */ - clk_enable(pck1_clk); - DBG("pck1 started\n"); - - return 0; -} - -static void eti_b1_shutdown(struct snd_pcm_substream *substream) -{ - /* Stop PCK1 clock. */ - clk_disable(pck1_clk); - DBG("pck1 stopped\n"); -} - -static int eti_b1_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - unsigned int rate; - int cmr_div, period; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ - break; - case 32000: - cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ - period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ - break; - case 48000: - cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ - period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_RCMR_PERIOD, period); - } - if (ret < 0) - return ret; - -#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - /* - * Codec in Master Mode. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - -#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - - return 0; -} - -static struct snd_soc_ops eti_b1_ops = { - .startup = eti_b1_startup, - .hw_params = eti_b1_hw_params, - .shutdown = eti_b1_shutdown, -}; - - -static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route intercon[] = { - - /* speaker connected to LHPOUT */ - {"Ext Spk", NULL, "LHPOUT"}, - - /* mic is connected to Mic Jack, with WM8731 Mic Bias */ - {"MICIN", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Int Mic"}, -}; - -/* - * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. - */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -{ - DBG("eti_b1_wm8731_init() called\n"); - - /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, - ARRAY_SIZE(eti_b1_dapm_widgets)); - - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); - - /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - - /* always connected */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link eti_b1_dai = { - .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai = &at91_ssc_dai[1], - .codec_dai = &wm8731_dai, - .init = eti_b1_wm8731_init, - .ops = &eti_b1_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1 = { - .name = "ETI_B1_WM8731", - .dai_link = &eti_b1_dai, - .num_links = 1, -}; - -static struct wm8731_setup_data eti_b1_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - -static struct snd_soc_device eti_b1_snd_devdata = { - .machine = &snd_soc_machine_eti_b1, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &eti_b1_wm8731_setup, -}; - -static struct platform_device *eti_b1_snd_device; - -static int __init eti_b1_init(void) -{ - int ret; - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio-1")) { - DBG("SSC1 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K); - if (!ssc->base) { - DBG("SSC1 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC1; - - eti_b1_snd_device = platform_device_alloc("soc-audio", 1); - if (!eti_b1_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); - eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; - - ret = platform_device_add(eti_b1_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_snd_device); - goto fail_io_unmap; - } - - at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ - at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ - at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ - at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ -/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ - at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ - - /* - * Set PCK1 parent to PLLB and its rate to 12 Mhz. - */ - pllb_clk = clk_get(NULL, "pllb"); - pck1_clk = clk_get(NULL, "pck1"); - - clk_set_parent(pck1_clk, pllb_clk); - clk_set_rate(pck1_clk, 12000000); - - DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); - - /* assign the GPIO pin to PCK1 */ - at91_set_B_periph(AT91_PIN_PA24, 0); - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); -#else - printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); -#endif - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); - return ret; -} - -static void __exit eti_b1_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - clk_put(pck1_clk); - clk_put(pllb_clk); - - platform_device_unregister(eti_b1_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); -} - -module_init(eti_b1_init); -module_exit(eti_b1_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -MODULE_LICENSE("GPL"); -- 1.4.2

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[alsa-devel] soc-core: Question about cpu_dai and codec_dai conversions
by Troy Kisky 30 Sep '08

30 Sep '08

Currently, we have the line runtime->hw.formats = codec_dai->playback.formats & cpu_dai->playback.formats; in soc-core.c This seems to force the two to support a common subset. For example some codec support only big-endian or little-endian My codec expects the most significant bit to be shifted 1st. I would call that big-endian. But the codec marks itself as only doing little endian, because the cpu_dai converts from little-endian to big-endian. Do all codecs expect the msb 1st? If so, then endianness would seem irrelevant to codecs and both should be set. The same goes with number of channels. The davinci dma engine can convert a playback stream from mono to stereo, or a capture stream from stereo to mono. The davinci McBsp can also be more efficient if allowed to swap left and right channels. Is there a method to let the codec know to swap them back? Thanks Troy

2 1

[alsa-devel] [PATCH] ASoC: Fix cs4270 error path
by Jean Delvare 30 Sep '08

30 Sep '08

The error path in cs4270_probe/cs4270_remove is pretty broken: * If cs4270_probe fails, codec is leaked. * If snd_soc_register_card fails, cs4270_i2c_driver stays registered. * If I2C support is enabled but no I2C device is found, i2c_del_driver is never called (neither in cs4270_probe nor in cs4270_remove.) Fix the first 2 problems by implementing a clean error path in cs4270_probe and jumping to its labels as needed. Fix the 3rd problem by removing the condition to call i2c_del_driver in cs4270_remove. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- Timur, can you please review and test this patch? I do not have the hardware so I couldn't test it myself. sound/soc/codecs/cs4270.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/codecs/cs4270.c 2008-08-31 13:59:17.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/cs4270.c 2008-08-31 14:23:34.000000000 +0200 @@ -727,7 +727,7 @@ static int cs4270_probe(struct platform_ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "cs4270: failed to create PCMs\n"); - return ret; + goto error_free_codec; } #ifdef USE_I2C @@ -736,8 +736,7 @@ static int cs4270_probe(struct platform_ ret = i2c_add_driver(&cs4270_i2c_driver); if (ret) { printk(KERN_ERR "cs4270: failed to attach driver"); - snd_soc_free_pcms(socdev); - return ret; + goto error_free_pcms; } /* Did we find a CS4270 on the I2C bus? */ @@ -759,10 +758,23 @@ static int cs4270_probe(struct platform_ ret = snd_soc_register_card(socdev); if (ret < 0) { printk(KERN_ERR "cs4270: failed to register card\n"); - snd_soc_free_pcms(socdev); - return ret; + goto error_del_driver; } + return 0; + +error_del_driver: +#ifdef USE_I2C + i2c_del_driver(&cs4270_i2c_driver); + +error_free_pcms: +#endif + snd_soc_free_pcms(socdev); + +error_free_codec: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; } @@ -773,8 +785,7 @@ static int cs4270_remove(struct platform snd_soc_free_pcms(socdev); #ifdef USE_I2C - if (socdev->codec->control_data) - i2c_del_driver(&cs4270_i2c_driver); + i2c_del_driver(&cs4270_i2c_driver); #endif kfree(socdev->codec); -- Jean Delvare

3 13

[alsa-devel] Salsa-lib : doesn't handle multiple device
by Matthieu CASTET 30 Sep '08

30 Sep '08

Hi, it seems there is a bug in card parsing (snd_card_get_index). If the device name is "hw:0,1", then in _snd_dev_get_device, snd_card_get_index will be called with "0,1". But snd_card_get_index expect to be called with only the card string number or name : it first check that the string is a number [1], and then try for each card to compare with the card name [2] I don't know what suppose to do snd_card_get_index, but I thing a easy way to fix it is to pass to snd_card_get_index only the string before the first comma. Matthieu [1] (isdigit(*string) && *(string + 1) == 0) || (isdigit(*string) && isdigit(*(string + 1)) && *(string + 2) == 0 [2] if (!strcmp((const char *)info.id, string))

2 3

[alsa-devel] [PATCH 3/5] [RFC] ALSA ASOC Adds DSP DAI format support for platform driver
by Arun KS 30 Sep '08

30 Sep '08

Enables DSP DAI format for mcbsp in omap platform driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3a4cc4b..1c024aa 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -292,6 +292,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; + default: /* Unsupported data format */ return -EINVAL; -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 5

[alsa-devel] [PATCH 4/5 v1] ASoC adds destination and source port for dma
by Arun KS 30 Sep '08

30 Sep '08

Adds destination and source port for dma in platform driver as required by OMAP1 Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-pcm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfea..e9084fd 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!cpu_is_omap1510()) { + if (!err & !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame -- 1.5.3.4

3 2

[alsa-devel] [PATCH 1/5 v1] ASoC Add TLV320AIC23 codec driver
by Arun KS 30 Sep '08

30 Sep '08

ASoC codec driver for TLV320AIC23 device Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 631 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 125 ++++++++ 4 files changed, 763 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0507fcf..bdead2d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" depends on SND_SOC && SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0731844..90f0a58 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..b602c87 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,631 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "tlv320aic23" +#define AIC23_VERSION "0.1" + +/* codec private data */ +struct tlv320aic23_priv { + unsigned int sysclk; + int master; +}; + +struct tlv320aic23_samplerate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, (u8) data, + (u8) (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *tlv320aic23_rec_src[] = { "Line", "Mic" }; +static const char *tlv320aic23_sidetone_src[] = + { "-6db", "-9db", "-12db", "-18db", "0db" }; + +static const struct soc_enum tlv320aic23_enum[] = { + SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 2, 2, + tlv320aic23_rec_src), + SOC_ENUM_SINGLE(TLV320AIC23_ANALOG_AUDIO_CONTROL, 6, 5, + tlv320aic23_sidetone_src), +}; + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[0]); + +static const struct snd_kcontrol_new tlv320aic23_sidetone_src_controls = +SOC_DAPM_ENUM("Route", tlv320aic23_enum[1]); + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R("PCM Playback Volume", TLV320AIC23_LEFT_CHANNEL_VOLUME, + TLV320AIC23_RIGHT_CHANNEL_VOLUME, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", TLV320AIC23_DIGITAL_AUDIO_CONTROL, 3, + 0x01, 1), + SOC_DOUBLE_R("Line Input Mute", TLV320AIC23_LEFT_LINE_VOLUME, + TLV320AIC23_RIGHT_LINE_VOLUME, 7, 0x01, 0), + SOC_DOUBLE_R("Line Input Volume", TLV320AIC23_LEFT_LINE_VOLUME, + TLV320AIC23_RIGHT_LINE_VOLUME, 0, 0x1f, 0), + SOC_SINGLE("Mic Switch", TLV320AIC23_ANALOG_AUDIO_CONTROL, 1, 0x01, 0), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANALOG_AUDIO_CONTROL, 0, + 0x01, 0), + SOC_SINGLE("Sidetone Switch", TLV320AIC23_ANALOG_AUDIO_CONTROL, 5, 1, + 0), + SOC_ENUM("Sidetone Gain", tlv320aic23_enum[1]), + SOC_SINGLE("Line Bypass", TLV320AIC23_ANALOG_AUDIO_CONTROL, 3, 1, 0), + +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", TLV320AIC23_POWER_DOWN_CONTROL, 0, 1, 1), + SOC_DAPM_SINGLE("Mic Switch", TLV320AIC23_POWER_DOWN_CONTROL, 1, 1, 1), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_POWER_DOWN_CONTROL, 3, + 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_POWER_DOWN_CONTROL, 2, + 1), + SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0, + &tlv320aic23_pga_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + {"LHPOUT", NULL, "DAC"}, + {"RHPOUT", NULL, "DAC"}, + + {"LOUT", NULL, "DAC"}, + {"ROUT", NULL, "DAC"}, + + {"Capture Source", "Line", "LLINEIN"}, + {"Capture Source", "Line", "RLINEIN"}, + + {"Capture Source", "Mic", "MICIN"}, + + {"PGA Mixer", "Line Switch", "Capture Source"}, + {"PGA Mixer", "Mic Switch", "Capture Source"}, + + {"ADC", NULL, "PGA Mixer"}, +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_samplerate_reg_info rate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_AUDIO_FORMAT) + & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((rate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(rate_reg_info))) { + count++; + } + + data = (rate_reg_info[count].divider << CLKIN_SHIFT) | + (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SAMPLE_RATE_CONTROL, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_FORMAT, iface_reg); + + return 0; +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_AUDIO_CONTROL) & + ~DACM_MUTE; + + if (mute) + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_CONTROL, + reg | DACM_MUTE); + + else + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_CONTROL, + reg); + + return 0; +} + +static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320aic23_priv *tlv320aic23 = codec->private_data; + + tlv320aic23->sysclk = freq; + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320aic23_priv *tlv320aic23 = codec->private_data; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_AUDIO_FORMAT) & + (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + tlv320aic23->master = 1; + iface_reg |= MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + tlv320aic23->master = 0; + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_FORMAT, iface_reg); + + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .hw_params = tlv320aic23_hw_params, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_sysclk = tlv320aic23_set_dai_sysclk, + .set_fmt = tlv320aic23_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_POWER_DOWN_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_POWER_DOWN_CONTROL, + reg | DEVICE_POWER_OFF | OUT_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 reg; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(tlv320aic23_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_POWER_DOWN_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_POWER_DOWN_CONTROL, + reg & (~DEVICE_POWER_OFF) & (~OUT_OFF)); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET_CONTROL, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + /* Power on the device */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_POWER_DOWN_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_POWER_DOWN_CONTROL, + reg & (~DEVICE_POWER_OFF) & (~OUT_OFF)); + + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_AUDIO_CONTROL, DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LEFT_LINE_VOLUME); + tlv320aic23_write(codec, TLV320AIC23_LEFT_LINE_VOLUME, + (reg & (~LIM_MUTED)) | (LRS_ENABLED)); + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RIGHT_LINE_VOLUME); + tlv320aic23_write(codec, TLV320AIC23_RIGHT_LINE_VOLUME, + (reg & (~LIM_MUTED)) | LRS_ENABLED); + reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANALOG_AUDIO_CONTROL); + tlv320aic23_write(codec, TLV320AIC23_ANALOG_AUDIO_CONTROL, + (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LEFT_CHANNEL_VOLUME, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + tlv320aic23_write(codec, TLV320AIC23_RIGHT_CHANNEL_VOLUME, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + + /* Activate interface */ + reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGITAL_INTERFACE_ACT); + tlv320aic23_write(codec, TLV320AIC23_DIGITAL_INTERFACE_ACT, + reg | ACT_ON); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320aic23_priv *tlv320aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + tlv320aic23 = kzalloc(sizeof(struct tlv320aic23_priv), GFP_KERNEL); + if (tlv320aic23 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = tlv320aic23; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..7ffeaf6 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,125 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LEFT_LINE_VOLUME 0x00 +#define TLV320AIC23_RIGHT_LINE_VOLUME 0x01 +#define TLV320AIC23_LEFT_CHANNEL_VOLUME 0x02 +#define TLV320AIC23_RIGHT_CHANNEL_VOLUME 0x03 +#define TLV320AIC23_ANALOG_AUDIO_CONTROL 0x04 +#define TLV320AIC23_DIGITAL_AUDIO_CONTROL 0x05 +#define TLV320AIC23_POWER_DOWN_CONTROL 0x06 +#define TLV320AIC23_DIGITAL_AUDIO_FORMAT 0x07 +#define TLV320AIC23_SAMPLE_RATE_CONTROL 0x08 +#define TLV320AIC23_DIGITAL_INTERFACE_ACT 0x09 +#define TLV320AIC23_RESET_CONTROL 0x0F + +/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define LZC_ON 0x0080 +#define LHV_DEFAULT 0x0079 +#define LHV_MAX 0x007f +#define LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define STA_REG(x) ((x)<<6) +#define STE_ENABLED 0x0020 +#define DAC_SELECTED 0x0010 +#define BYPASS_ON 0x0008 +#define INSEL_MIC 0x0004 +#define MICM_MUTED 0x0002 +#define MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define DACM_MUTE 0x0008 +#define DEEMP_32K 0x0002 +#define DEEMP_44K 0x0004 +#define DEEMP_48K 0x0006 +#define ADCHP_ON 0x0001 + +/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define MS_MASTER 0x0040 +#define LRSWAP_ON 0x0020 +#define LRP_ON 0x0010 +#define IWL_16 0x0000 +#define IWL_20 0x0004 +#define IWL_24 0x0008 +#define IWL_32 0x000C +#define FOR_I2S 0x0002 +#define FOR_DSP 0x0003 +#define FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define CLKOUT_HALF 0x0080 +#define CLKIN_HALF 0x0040 +#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */ +#define USB_CLK_ON 0x0001 +#define SR_MASK 0xf +#define CLKOUT_SHIFT 7 +#define CLKIN_SHIFT 6 +#define SR_SHIFT 2 +#define BOSR_SHIFT 1 + +/* Digital interface register */ +#define ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define DEFAULT_OUTPUT_VOLUME 0x70 +#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */ + +#define OUTPUT_VOLUME_MIN LHV_MIN +#define OUTPUT_VOLUME_MAX LHV_MAX +#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN) +#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX + +#define INPUT_VOLUME_MIN LIV_MIN +#define INPUT_VOLUME_MAX LIV_MAX +#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN) +#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX + +#define SIDETONE_MASK 0x1c0 +#define SIDETONE_0 0x100 +#define SIDETONE_6 0x000 +#define SIDETONE_9 0x040 +#define SIDETONE_12 0x080 +#define SIDETONE_18 0x0c0 + +#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \ + INSEL_MIC | MICB_20DB) + + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ + -- 1.5.3.4

2 3

[alsa-devel] [PATCH 2/5 v1] Adds ASOC Support for 5912osk
by Arun KS 30 Sep '08

30 Sep '08

Adding ASOC machine driver for 5912osk Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/5912osk.c | 232 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + 3 files changed, 242 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/5912osk.c diff --git a/sound/soc/omap/5912osk.c b/sound/soc/omap/5912osk.c new file mode 100644 index 0000000..30932a1 --- /dev/null +++ b/sound/soc/omap/5912osk.c @@ -0,0 +1,232 @@ +/* + * 5912osk.c -- SoC audio for 5912 OSK + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks(a)mistralsolutions.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add 5912osk specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up 5912osk specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "5912OSK", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC 5912 OSK"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e7..fc946bc 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_5912OSK + tristate "SoC Audio support for omap 5912osk" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on 5912osk. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58..d4f62fc 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-5912osk-objs := 5912osk.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_5912OSK) += snd-soc-5912osk.o -- 1.5.3.4

2 1

[alsa-devel] [PATCH 5/5 v1] ASoC Adds DSP DAI format support for platform driver
by Arun KS 30 Sep '08

30 Sep '08

Enables DSP DAI format for mcbsp in omap platform driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e1..fb920e1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -245,6 +245,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; -- 1.5.3.4

1 0

[alsa-devel] [PATCH 3/5 v1] Adds I2C board info for tlvaic23
by Arun KS 30 Sep '08

30 Sep '08

Adding I2C board info required for tlvaic23 i2c chip driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- arch/arm/mach-omap1/board-osk.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 3e766e4..e506aef 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -258,8 +258,10 @@ static struct i2c_board_info __initdata osk_i2c_board_info[] = { .platform_data = &tps_board, }, + { + I2C_BOARD_INFO("tlv320aic23", 0x1B), + }, /* TODO when driver support is ready: - * - aic23 audio chip at 0x1a * - on Mistral, 24c04 eeprom at 0x50 * - optionally on Mistral, ov9640 camera sensor at 0x30 */ -- 1.5.3.4

1 0

[alsa-devel] [PATCH 0/5 v1] ASoC Support for TLVaic23b audio codec
by Arun KS 30 Sep '08

30 Sep '08

Hi all, I have incorporated the comments from Mark Brown, Jarkko Nikula and Felipe Balbi. The following series of patches are against the tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git The new series of patches have the following changes from previous patches: 1) The codec is named tlv320aic23 uniformly throughout the code. 2) Removed set_bias_level function from the codec driver and is switching on/off the codec at resume/suspend function. 3) Added mic sidetone and line bypass. 4) Renamed the machine driver file as 5912osk.c instead of osk.c Thanks Arun

1 0

[alsa-devel] [PATCH] ASoC: Fix cs4270 error path
by Jean Delvare 30 Sep '08

30 Sep '08

The error path in cs4270_probe/cs4270_remove is pretty broken: * If cs4270_probe fails, codec is leaked. * If snd_soc_register_card fails, cs4270_i2c_driver stays registered. * If I2C support is enabled but no I2C device is found, i2c_del_driver is never called (neither in cs4270_probe nor in cs4270_remove. Fix all 3 problems by implementing a clean error path in cs4270_probe and jumping to its labels as needed. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Acked-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) --- linux-2.6.27-rc8.orig/sound/soc/codecs/cs4270.c 2008-09-30 11:11:22.000000000 +0200 +++ linux-2.6.27-rc8/sound/soc/codecs/cs4270.c 2008-09-30 11:33:03.000000000 +0200 @@ -681,7 +681,7 @@ static int cs4270_probe(struct platform_ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "cs4270: failed to create PCMs\n"); - return ret; + goto error_free_codec; } #ifdef USE_I2C @@ -690,8 +690,7 @@ static int cs4270_probe(struct platform_ ret = i2c_add_driver(&cs4270_i2c_driver); if (ret) { printk(KERN_ERR "cs4270: failed to attach driver"); - snd_soc_free_pcms(socdev); - return ret; + goto error_free_pcms; } /* Did we find a CS4270 on the I2C bus? */ @@ -713,10 +712,23 @@ static int cs4270_probe(struct platform_ ret = snd_soc_register_card(socdev); if (ret < 0) { printk(KERN_ERR "cs4270: failed to register card\n"); - snd_soc_free_pcms(socdev); - return ret; + goto error_del_driver; } + return 0; + +error_del_driver: +#ifdef USE_I2C + i2c_del_driver(&cs4270_i2c_driver); + +error_free_pcms: +#endif + snd_soc_free_pcms(socdev); + +error_free_codec: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; } @@ -727,8 +739,7 @@ static int cs4270_remove(struct platform snd_soc_free_pcms(socdev); #ifdef USE_I2C - if (socdev->codec->control_data) - i2c_del_driver(&cs4270_i2c_driver); + i2c_del_driver(&cs4270_i2c_driver); #endif kfree(socdev->codec); -- Jean Delvare

1 0

[alsa-devel] Support for ALC1200
by Norbert Piotrowski 30 Sep '08

30 Sep '08

I have a question about ALC1200 which is used by ASUS in P5Q motherboard. Unfortunately I have that board and problems with spdif output ( simply no light, it's not a problem with mute/unmute IEC958 ). ALC1200 is recognized as ALC883, below more details. Realtek don't publish specification for that chip because it was made for Asus request. From what i gathered, 1200 is 890 in disguise, but I'm not sure about that. I would like to know if there is any chance to add support in alsa driver for that kind of codec in the near future ? I will be very grateful for any help or suggestion. Best regards, Norbert norbert@nelson / $ uname -a Linux nelson 2.6.24-16-generic #1 SMP Thu Apr 10 13:23:42 UTC 2008 i686 GNU/Linux norbert@nelson / $ alsactl -v alsactl version 1.0.18rc3 norbert@nelson / $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 norbert@nelson / $ aplay -L default:CARD=Intel HDA Intel, ALC883 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=Intel,DEV=0 HDA Intel HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) norbert@nelson / $ cat /proc/asound/card0/codec#0 Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x104382fe Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x13 0x13] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Trigger Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Trigger Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0836: IN OUT Detect Trigger Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0836: IN OUT Detect Trigger Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19850: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19d60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181305f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214d20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4015e601: [N/A] Speaker at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x01447130: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Yellow DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

1 0

[alsa-devel] sound tree slightly rebased
by Takashi Iwai 30 Sep '08

30 Sep '08

Hi, since I replaced Jean's patch to fix Vladimir's e-mail address, I had to rebase topic/asoc, master and for-next branches. If you are based on it, please be careful at the next pull. Sorry for inconvenience. Takashi

1 0

[alsa-devel] AIC23 codec ADC problem
by Troy Kisky 30 Sep '08

30 Sep '08

Hi All I'm using the TLV320AIC23 to record sound from a mic. It sounds fine when I play it back, but when I look at the data, the value in one channel is always exactly double the value in the other. Since the mic gives the same input to both ADC's I was expecting to have the same value in the left & right channels. Has anyone seen this before? Any guesses to what I might be doing wrong? Can someone using an AIC23 check their data for me? I'm using a davinci DM644x platform and spi interface to the codec in dsp_a mode. sample data 00ec40 fffeffff 00040002 00040002 00000000 00ec50 fffeffff fffeffff 00000000 fffafffd 00ec60 fffeffff 00020001 00040002 00080004 00ec70 000a0005 0016000b 0014000a 001a000d 00ec80 001c000e 001e000f 00200010 00260013 00ec90 001e000f 001e000f 001e000f 001c000e 00eca0 001c000e 0018000c 0014000a 0014000a 00ecb0 0014000a 000e0007 00120009 0014000a ^ ^ ^ ^ Always even After swapping the left & right channels at the codec 011af0 fffcfffe fffeffff 00060003 00060003 011b00 000a0005 000c0006 000e0007 000c0006 011b10 00080004 00080004 00060003 00060003 011b20 00080004 00060003 00060003 00040002 011b30 00020001 00020001 00000000 00020001 011b40 fffeffff 00000000 00040002 00060003 011b50 000c0006 000e0007 00100008 0014000a 011b60 00120009 000e0007 00100008 00100008 011b70 000e0007 00080004 00080004 00060003 011b80 00060003 00060003 00060003 000a0005 No change, so the ADC's both do convert to the same value. But there appears to be a 1 bit delay between the two channels. My McBsp is reading 1 32 bit element, so I don't think it can shift only half of the element. After switching to dsp_b mode (an extra bit delay before data starts) 00b130 fff8fff8 fff5fff5 fffafffa fffcfffc 00b140 fffefffe 00010001 00000000 00010001 00b150 00020002 00010001 00010001 00010001 00b160 00000000 ffffffff 00020002 00000000 00b170 00000000 00000000 fffefffe fffefffe 00b180 fffdfffd fffdfffd 00000000 ffffffff 00b190 00020002 00010001 00020002 00030003 00b1a0 00030003 00030003 00050005 00030003 00b1b0 00050005 00030003 00010001 00000000 When I play this back in dsp_b mode I hear random soft popping from the speaker, but it is loud. Playing it in dsp_a mode is quiet with no popping. So, I conclude that recording with the codec is buggy in dsp_a mode and playback is buggy in dsp_b mode. I hope someone will tell me I'm wrong. Thanks Troy

1 1

[alsa-devel] maintaining fixed latency between output and input?
by Aaron J. Grier 30 Sep '08

30 Sep '08

what's the most straight-forward way to maintain fixed latency between output and input without sitting in loop and continuously checking snd_pcm_avail_update() for the output and input descriptors? I was originally going to use callbacks, but I've recently read that is not a good idea. poll() seems to be the reccomended method. I could use separate threads for output and input, but then I have to perform synchronization between them, marshal external requests between the two threads, and make a lot more work for myself. I'm visualizing something like this: for(;;) { poll() if (output_is_ready) { /* write output buffer */ /* do output-side housekeeping */ } if (input_is_ready) { /* read input buffer */ /* do input-side housekeeping */ } /* communicate with other threads */ } combining the output and input snd_pcm_t into a single descriptor list for poll() seems straight-forward, but can I pass this superset list to snd_pcm_poll_descriptors_revents() without it getting confused? -- Aaron J. Grier | "Not your ordinary poofy goof." | agrier(a)poofygoof.com

2 2

[alsa-devel] [PATCH 0/5] [RFC] ALSA ASoC Support for TLVaic23b audio codec
by Arun KS 30 Sep '08

30 Sep '08

Hi all, The following patches adds ASOC driver for TLVaic23b audio codec. Tested playback and capture on omap 5912osk board with some hacks in the omap platform and machine driver. Thanks, Arun -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 11

Re: [alsa-devel] Secure delivery
by ink＠jurassic.park.msu.ru 29 Sep '08

29 Sep '08

Bad Gateway: The message has been attached. ++++ Attachment: No Virus found ++++ F-Secure AntiVirus - www.f-secure.com

1 0

[alsa-devel] [PATCH 0/6] ASoC Blackfin and API updates
by Mark Brown 29 Sep '08

29 Sep '08

The following changes since commit ad7f41b60efde140df0f929e8d408ebe2e59cb0e: Jean Delvare (1): ALSA: Drop Vladimir Barinov's e-mail address adding the Blackfin updates Brian posted over the weekend plus some more usage of snd_soc_dapm_nc_pin() are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Cliff Cai (4): ASoC codec: AD73311 audio codec driver ASoC Blackfin: add I2S DAI support for AD73311 ASoC Blackfin: add asoc ad73311 driver supporting in Blackfin boards ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled Mark Brown (2): ASoC: Use snd_soc_dapm_nc_pin() in Zaurus machine drivers ASoC: Use snd_soc_dapm_nc_pin() in N810 machine driver sound/soc/blackfin/Kconfig | 16 +++ sound/soc/blackfin/Makefile | 3 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 42 +++++- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 240 +++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-i2s.c | 47 +++++-- sound/soc/blackfin/bf5xx-sport.h | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad73311.c | 107 ++++++++++++++++ sound/soc/codecs/ad73311.h | 90 +++++++++++++ sound/soc/omap/n810.c | 6 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/spitz.c | 14 +- sound/soc/pxa/tosa.c | 4 +- 16 files changed, 547 insertions(+), 39 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad73311.c create mode 100644 sound/soc/codecs/ad73311.c create mode 100644 sound/soc/codecs/ad73311.h

2 8

[alsa-devel] [PATCH 4/5] [RFC] ALSA ASOC adds destination and source port for dma
by Arun KS 29 Sep '08

29 Sep '08

Adds destination and source ports for dma which is required for OMAP1 Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfea..9e670ef 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,6 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); + if (!cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any @@ -147,12 +148,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 3

[alsa-devel] [PATCH] ALSA: cs46xx: Add PCI IDs for TerraTec and Hercules cards
by Vedran Miletić 29 Sep '08

29 Sep '08

This patch adds PCI IDs for: * TerraTec DMX XFire 1024 * Hercules Gamesurround Fortissimo II * Hercules Gamesurround Fortissimo III 7.1 All those cards were supported as generic CS46xx device, so they will work as before. I'm pretty sure that first two cards work, as they have same hardware design as reference card. Not sure about Fortissimo III, but this won't break it if it worked. Tested on TerraTec DMX XFire 1024. -- Vedran Miletić

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Re: [alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

Mark Brown wrote: > On Mon, Sep 29, 2008 at 03:15:04PM +0530, Arun KS wrote: >> ASOC Audio driver for TLVaic23b audio codec > > This looks good overall, thanks! There's quite a few comments below but > they're mostly minor things at the coding style issues. > > The only issue I've got overall is that the name of the codec seems to > vary through the driver - sometimes it's referred to as the tlv320aic23, > sometimes as aic23 and sometimes as tlvaic23. It'd be good if the > naming scheme were more consistent. My inclination would be to go with > tlv320aic23 since it will reduce the risk of collisions and the general > style is to use the full part number (then again, I know the tlv320aic3x > driver didn't do that). > >> --- a/sound/soc/codecs/Kconfig >> +++ b/sound/soc/codecs/Kconfig >> @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 >> tristate >> depends on SND_SOC && TWL4030_CORE >> > > This patch is against the OMAP tree - please generate all submitted > patches against either the ASoC dev tree or the topic/asoc branch of: > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > Even if other patches in the series depend on the OMAP tree there should > be no reason not to merge an I2C codec driver like this with no external > dependencies immediately. > >> +config SND_SOC_TLV320AIC23 >> + tristate >> + depends on I2C >> + > > Please also add this to the SND_SOC_ALL_CODECS Kconfig option. > >> +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, >> + unsigned int value) >> +{ >> + >> + u8 data; >> + >> + /* TLV320AIC23 has 7 bit address and 9 bits of data >> + * so we need to switch one data bit into reg and rest >> + * of data into val >> + */ >> + >> + if ((reg < 0 || reg > 9) && (reg != 15)) { >> + printk(KERN_WARNING "Invalid register R%d\n", reg); >> + return -1; >> + } > > It'd be nice if this said which device was being written to - adding in > the driver name to the printk(), for example. It'd make it easier for > someone seeing the message in their logs to work out where it came from. > >> + data = (reg << 1) | (value >> 8 & 0x01); >> + >> + aic23_write_reg_cache(codec, reg, value); >> + >> + if (codec->hw_write(codec->control_data, (u8) data, >> + (u8) (value & 0xff)) == 0) >> + return 0; >> + >> + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); > > Again, saying where the error came from would be nice. > >> + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), > > This should be "Mic Switch" to help UIs display the control correctly. > >> +/* aic23 related */ >> +static const struct aic23_samplerate_reg_info >> + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { >> + {4000, 0x06, 1}, /* 4000 */ > > You could just use [] for the size of the array and let the compiler > figure it out - you actually use ARRAY_SIZE() rather than the #define > for the array size later anyway. > >> +static int aic23_set_bias_level(struct snd_soc_codec *codec, >> + enum snd_soc_bias_level level) >> +{ >> + u16 reg; >> + >> + switch (level) { >> + case SND_SOC_BIAS_ON: >> + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); >> + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, >> + reg & (~DEVICE_POWER_OFF)); >> + /* Activate interface */ >> + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); >> + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); >> + break; >> + case SND_SOC_BIAS_PREPARE: >> + break; >> + case SND_SOC_BIAS_STANDBY: >> + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); >> + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, >> + reg | DEVICE_POWER_OFF); >> + /* Deactivate interface */ >> + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); >> + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); > > Hrm. This probably deserves a comment about what DEVICE_POWER_OFF does > that the normal DAPM control doesn't. Given that there are no bypass > paths there should be no risk of interfering with DAPM but it raises a > bit of an eyebrow. There is bypass for Line and sidetone for Mic. I will add that also and will make DEVICE_POWER_OFF also as DAPM control. Will send the patches against ASoC dev tree. > >> +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) >> + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; >> + codec->hw_read = NULL; >> + ret = i2c_add_driver(&aic23b_i2c_driver); >> + if (ret != 0) >> + printk(KERN_ERR "can't add i2c driver"); >> +#else >> + /* Add other interfaces here */ >> +#endif > > If the device doesn't support anything except I2C then please remove all > the conditionals on I2C support from the driver (they don't add > anything). If the device does support other control methods then please > change this to be something like: > > #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) > ... > #endif > /* Add other interfaces here */ > > That way if someone adds support for the other control interfaces > they'll be encouraged to do it in a way that allows kernels to be built > supporting multiple systems simultaneously. > >> +#define AIC23_CACHEREGNUM 16 >> +/* Codec TLV320AIC23 */ >> +#define LEFT_LINE_VOLUME_ADDR 0x00 >> +#define RIGHT_LINE_VOLUME_ADDR 0x01 >> +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 >> +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 >> +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 >> +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 >> +#define POWER_DOWN_CONTROL_ADDR 0x06 >> +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 >> +#define SAMPLE_RATE_CONTROL_ADDR 0x08 >> +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 >> +#define RESET_CONTROL_ADDR 0x0F > > These (and most of the rest of the defines in the header) should all be > namespaced - there's a very good chance of them colliding with other > things and this header is going to be included in the machine drivers. > >> +/* Define to set the AIC23 as the master w.r.t McBSP */ >> +#define AIC23_MASTER > >> +#ifndef DEFAULT_BITPERSAMPLE >> +#define DEFAULT_BITPERSAMPLE 16 >> +#endif > >> +#define DEFAULT_SAMPLE_RATE 44100 >> +#define CODEC_CLOCK 12000000 >> +#define AUDIO_MCBSP OMAP_MCBSP1 > > None of these should be in this driver - they should all be part of the > machine driver. > >> +struct aic23_samplerate_reg_info { >> + u32 sample_rate; >> + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ >> + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ >> +}; > > This could just be pushed into the driver. > >

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[alsa-devel] [PATCH 2/5] [RFC] ALSA ASOC Support for 5912 osk
by Arun KS 29 Sep '08

29 Sep '08

Adding ASOC machine driver for osk Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 3 +- sound/soc/omap/osk.c | 230 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 240 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/osk.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d7b8939..f2e6d01 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -22,3 +22,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_OSK + tristate "SoC Audio support for 5912 OSK" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on 5912 OSK. + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index b96b97b..ccb3ddd 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -8,7 +8,8 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o snd-soc-overo-objs := overo.o +snd-soc-osk-objs := osk.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o - +obj-$(CONFIG_SND_OMAP_SOC_OSK) += snd-soc-osk.o diff --git a/sound/soc/omap/osk.c b/sound/soc/omap/osk.c new file mode 100644 index 0000000..f4dfe66 --- /dev/null +++ b/sound/soc/omap/osk.c @@ -0,0 +1,230 @@ +/* + * osk.c -- SoC audio for 5912 OSK + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks(a)mistralsolutions.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +static struct clk *aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_aic23_init(struct snd_soc_codec *codec) +{ + + /* Add OSK specific widgets */ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* Set up OSK specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic23_dai, + .init = osk_aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "5912OSK", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_aic23, +}; + +static struct platform_device *osk_snd_device; + +#define CODEC_CLOCK 12000000 + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + return -ECANCELED; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK, + clk_get_usecount(aic23_mclk)); + + return 0; +err1: + clk_put(aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC 5912 OSK"); +MODULE_LICENSE("GPL"); -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ICH9 Sound
by Ronald Horn 29 Sep '08

29 Sep '08

Hello, I'm using a HP Pavilion dv7-1045eg. The Problem is I have no sound, because there is no driver witch works. Here are some reports from my system: / lspci -nv/ 00:1b.0 0403: 8086:293e (rev 03) Subsystem: 103c:30f4 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at df300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel / cat /proc/asound/cards/ 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xdf300000 irq 22 /head -n 1 /proc/asound/card0/codec*/ ==> /proc/asound/card0/codec#0 <== Codec: IDT 92HD71B7X ==> /proc/asound/card0/codec#2 <== Codec: Generic 10de ID 6 I'm using openSuse 11.0 with Kernel 2.6.25.16-0.1-pae. Maybe your project could develop some drivers. Greetings Ronald

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[alsa-devel] [PATCH] ASoC: Fix another cs4270 error path
by Jean Delvare 29 Sep '08

29 Sep '08

Conversion to new-style i2c driver missed the error path of the probe function. Fix it. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) --- linux-2.6.27-rc7.orig/sound/soc/codecs/cs4270.c 2008-09-27 17:45:47.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/cs4270.c 2008-09-27 18:10:52.000000000 +0200 @@ -610,17 +610,12 @@ static int cs4270_i2c_probe(struct i2c_c return 0; error: - if (codec->control_data) { - i2c_detach_client(i2c_client); - codec->control_data = NULL; - } + codec->control_data = NULL; kfree(codec->reg_cache); codec->reg_cache = NULL; codec->reg_cache_size = 0; - kfree(i2c_client); - return ret; } -- Jean Delvare

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[alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

ASOC Audio driver for TLVaic23b audio codec Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 634 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 143 +++++++++ 4 files changed, 783 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b1a5eed..7be0b37 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 tristate depends on SND_SOC && TWL4030_CORE +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a519ced..adb5aa6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-tlv320aic23-objs := tlv320aic23.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -25,3 +26,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..eefeb2d --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,634 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/tlv320aic23.c by Vladimir Barinov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "aic23" +#define AIC23_VERSION "0.1" + +/* codec private data */ +struct aic23_priv { + unsigned int sysclk; + int master; +}; + +/* + * AIC23 register cache + */ +static const u16 aic23_reg[AIC23_CACHEREGNUM] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read aic23 register cache + */ +static inline unsigned int aic23_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write aic23 register cache + */ +static inline void aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the aic23 register space + */ +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "Invalid register R%d\n", reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, (u8) data, + (u8) (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); + + return -EIO; +} + +static const struct snd_kcontrol_new aic23_snd_controls[] = { + SOC_DOUBLE_R("PCM Playback Volume", LEFT_CHANNEL_VOLUME_ADDR, + RIGHT_CHANNEL_VOLUME_ADDR, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", DIGITAL_AUDIO_CONTROL_ADDR, 3, 0x01, + 1), + SOC_DOUBLE_R("Line Input Mute", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 7, 0x01, 0), + SOC_DOUBLE_R("Line Input Volume", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 0, 0x1f, 0), + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), + SOC_SINGLE("Mic Booster Switch", ANALOG_AUDIO_CONTROL_ADDR, 0, 0x01, 0), + +}; + +/* add non dapm controls */ +static int aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} +static const char *aic23_rec_src[] = { "Line", "Mic" }; + +static const struct soc_enum aic23_enum[] = { + SOC_ENUM_SINGLE(ANALOG_AUDIO_CONTROL_ADDR, 2, 2, aic23_rec_src), +}; + +static const struct snd_kcontrol_new aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", aic23_enum[0]); + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new aic23_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", POWER_DOWN_CONTROL_ADDR, 0, 1, 1), + SOC_DAPM_SINGLE("Mic Switch", POWER_DOWN_CONTROL_ADDR, 1, 1, 1), +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", POWER_DOWN_CONTROL_ADDR, 3, 1), + SND_SOC_DAPM_PGA("OUT", POWER_DOWN_CONTROL_ADDR, 4, 1, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", POWER_DOWN_CONTROL_ADDR, 2, 1), + SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic23_pga_mixer_controls[0], + ARRAY_SIZE(aic23_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &aic23_rec_src_mux_controls), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"OUT", NULL, "DAC"}, + + {"LHPOUT", NULL, "OUT"}, + {"RHPOUT", NULL, "OUT"}, + + {"LOUT", NULL, "OUT"}, + {"ROUT", NULL, "OUT"}, + + {"Capture Source", "Line", "LLINEIN"}, + {"Capture Source", "Line", "RLINEIN"}, + + {"Capture Source", "Mic", "MICIN"}, + + {"PGA Mixer", "Line Switch", "Capture Source"}, + {"PGA Mixer", "Mic Switch", "Capture Source"}, + + {"ADC", NULL, "PGA Mixer"}, + +}; + +/* aic23 related */ +static const struct aic23_samplerate_reg_info + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_FORMAT_ADDR) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((rate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(rate_reg_info))) { + count++; + } + + data = (rate_reg_info[count].divider << CLKIN_SHIFT) | + (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON; + + aic23_write(codec, SAMPLE_RATE_CONTROL_ADDR, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_CONTROL_ADDR) & ~DACM_MUTE; + + if (mute) + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg | DACM_MUTE); + + else + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg); + + return 0; +} + +static int aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23_priv *aic23 = codec->private_data; + + aic23->sysclk = freq; + return 0; +} + +static int aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23_priv *aic23 = codec->private_data; + u16 iface_reg; + + iface_reg = + aic23_read_reg_cache(codec, DIGITAL_AUDIO_FORMAT_ADDR) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic23->master = 1; + iface_reg |= MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic23->master = 0; + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= FOR_LJUST; + break; + default: + return -EINVAL; + + } + + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg & (~DEVICE_POWER_OFF)); + /* Activate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg | DEVICE_POWER_OFF); + /* Deactivate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); + + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->bias_level = level; + + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai aic23_dai = { + .name = "aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .hw_params = aic23_hw_params, + }, + .dai_ops = { + .digital_mute = aic23_mute, + .set_sysclk = aic23_set_dai_sysclk, + .set_fmt = aic23_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(aic23_dai); + +static int aic23_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(aic23_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "aic23"; + codec->owner = THIS_MODULE; + codec->read = aic23_read_reg_cache; + codec->write = aic23_write; + codec->set_bias_level = aic23_set_bias_level; + codec->dai = &aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(aic23_reg); + codec->reg_cache = kmemdup(aic23_reg, sizeof(aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + aic23_write(codec, RESET_CONTROL_ADDR, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to create pcms\n"); + goto pcm_err; + } + + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K); + + /* Unmute input */ + reg = aic23_read_reg_cache(codec, LEFT_LINE_VOLUME_ADDR); + aic23_write(codec, LEFT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | (LRS_ENABLED)); + reg = aic23_read_reg_cache(codec, RIGHT_LINE_VOLUME_ADDR); + aic23_write(codec, RIGHT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | LRS_ENABLED); + reg = aic23_read_reg_cache(codec, ANALOG_AUDIO_CONTROL_ADDR); + aic23_write(codec, ANALOG_AUDIO_CONTROL_ADDR, + (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON)); + + /* Default output volume */ + aic23_write(codec, LEFT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + aic23_write(codec, RIGHT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + + /* off, with power on */ + aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + aic23_add_controls(codec); + aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit aic23_i2c_remove(struct i2c_client *i2c) +{ + + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlvaic23b_id[] = { + {"tlvaic23b", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlvaic23b_id); + +static struct i2c_driver aic23b_i2c_driver = { + .driver = { + .name = "tlvaic23b", + }, + .probe = aic23_codec_probe, + .remove = __exit_p(aic23_i2c_remove), + .id_table = tlvaic23b_id, +}; + +#endif + +static int aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct aic23_priv *aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + aic23 = kzalloc(sizeof(struct aic23_priv), GFP_KERNEL); + if (aic23 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = aic23; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&aic23b_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#else + /* Add other interfaces here */ +#endif + return ret; +} + +static int aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* power down chip */ + if (codec->control_data) + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic23b_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_aic23 = { + .probe = aic23_probe, + .remove = aic23_remove, + .suspend = aic23_suspend, + .resume = aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..9c3808c --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,143 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC23_H +#define _AIC23_H + +#define AIC23_CACHEREGNUM 16 +/* Codec TLV320AIC23 */ +#define LEFT_LINE_VOLUME_ADDR 0x00 +#define RIGHT_LINE_VOLUME_ADDR 0x01 +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 +#define POWER_DOWN_CONTROL_ADDR 0x06 +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 +#define SAMPLE_RATE_CONTROL_ADDR 0x08 +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 +#define RESET_CONTROL_ADDR 0x0F + +/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define LZC_ON 0x0080 +#define LHV_DEFAULT 0x0079 +#define LHV_MAX 0x007f +#define LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define STA_REG(x) ((x)<<6) +#define STE_ENABLED 0x0020 +#define DAC_SELECTED 0x0010 +#define BYPASS_ON 0x0008 +#define INSEL_MIC 0x0004 +#define MICM_MUTED 0x0002 +#define MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define DACM_MUTE 0x0008 +#define DEEMP_32K 0x0002 +#define DEEMP_44K 0x0004 +#define DEEMP_48K 0x0006 +#define ADCHP_ON 0x0001 + +/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define MS_MASTER 0x0040 +#define LRSWAP_ON 0x0020 +#define LRP_ON 0x0010 +#define IWL_16 0x0000 +#define IWL_20 0x0004 +#define IWL_24 0x0008 +#define IWL_32 0x000C +#define FOR_I2S 0x0002 +#define FOR_DSP 0x0003 +#define FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define CLKOUT_HALF 0x0080 +#define CLKIN_HALF 0x0040 +#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */ +#define USB_CLK_ON 0x0001 +#define SR_MASK 0xf +#define CLKOUT_SHIFT 7 +#define CLKIN_SHIFT 6 +#define SR_SHIFT 2 +#define BOSR_SHIFT 1 + +/* Digital interface register */ +#define ACT_ON 0x0001 + +/* Define to set the AIC23 as the master w.r.t McBSP */ +#define AIC23_MASTER + +#define NUMBER_SAMPLE_RATES_SUPPORTED 10 + +/* + * AUDIO related MACROS + */ +#ifndef DEFAULT_BITPERSAMPLE +#define DEFAULT_BITPERSAMPLE 16 +#endif + +#define DEFAULT_SAMPLE_RATE 44100 +#define CODEC_CLOCK 12000000 +#define AUDIO_MCBSP OMAP_MCBSP1 + +#define DEFAULT_OUTPUT_VOLUME 0x70 +#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */ + +#define OUTPUT_VOLUME_MIN LHV_MIN +#define OUTPUT_VOLUME_MAX LHV_MAX +#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN) +#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX + +#define INPUT_VOLUME_MIN LIV_MIN +#define INPUT_VOLUME_MAX LIV_MAX +#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN) +#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX + +#define SIDETONE_MASK 0x1c0 +#define SIDETONE_0 0x100 +#define SIDETONE_6 0x000 +#define SIDETONE_9 0x040 +#define SIDETONE_12 0x080 +#define SIDETONE_18 0x0c0 + +#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \ + INSEL_MIC | MICB_20DB) + +struct aic23_samplerate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +extern struct snd_soc_dai aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_aic23; + +#endif /* _AIC23_H */ + -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] alsa: make the CS4270 driver a new-style I2C driver
by Timur Tabi 29 Sep '08

29 Sep '08

Update the CS4270 ALSA device driver to use the new-style I2C interface. Starting with the 2.6.27 PowerPC kernel, I2C devices that have entries in the device trees can no longer be probed by old-style I2C drivers. The device tree for Freescale MPC8610 HPCD has included an entry for the CS4270 since 2.6.25, but that entry was previously ignored by the PowerPC I2C subsystem. Since that's no longer the case, the best solution is to update the CS4270 driver to a new-style interface, rather than try to revert the behavior of new PowerPC I2C subsystem. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for 2.6.27 (ASoC V1). It's not needed for 2.6.26, because that kernel uses the older PowerPC I2C code. It's not needed for ASoC V2, because the ASoC V2 CS4270 driver has always been a new-style driver. sound/soc/codecs/cs4270.c | 71 +++++++++----------------------------------- 1 files changed, 15 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 9deb8c7..82d94f0 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -490,34 +490,7 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) #endif -static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); - -/* - * Notify the driver that a new I2C bus has been found. - * - * This function is called for each I2C bus in the system. The function - * then asks the I2C subsystem to probe that bus at the addresses on which - * our device (the CS4270) could exist. If a device is found at one of - * those addresses, then our probe function (cs4270_i2c_probe) is called. - */ -static int cs4270_i2c_attach(struct i2c_adapter *adapter) -{ - return i2c_probe(adapter, &addr_data, cs4270_i2c_probe); -} - -static int cs4270_i2c_detach(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - - i2c_detach_client(client); - codec->control_data = NULL; - - kfree(codec->reg_cache); - codec->reg_cache = NULL; - - kfree(client); - return 0; -} +static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { @@ -525,14 +498,19 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "CS4270 I2C", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_CS4270, - .attach_adapter = cs4270_i2c_attach, - .detach_client = cs4270_i2c_detach, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, }; /* @@ -561,11 +539,11 @@ static struct snd_soc_device *cs4270_socdev; * Note: snd_soc_new_pcms() must be called before this function can be called, * because of snd_ctl_add(). */ -static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) +static int cs4270_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = cs4270_socdev; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c_client = NULL; int i; int ret = 0; @@ -578,12 +556,6 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) /* Note: codec_dai->codec is NULL here */ - i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); - if (!i2c_client) { - printk(KERN_ERR "cs4270: could not allocate I2C client\n"); - return -ENOMEM; - } - codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); if (!codec->reg_cache) { printk(KERN_ERR "cs4270: could not allocate register cache\n"); @@ -591,13 +563,6 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) goto error; } - i2c_set_clientdata(i2c_client, codec); - strcpy(i2c_client->name, "CS4270"); - - i2c_client->driver = &cs4270_i2c_driver; - i2c_client->adapter = adapter; - i2c_client->addr = addr; - /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); @@ -612,18 +577,10 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) goto error; } - printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr); + printk(KERN_INFO "cs4270: found device at I2C address %X\n", + i2c_client->addr); printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); - /* Tell the I2C layer a new client has arrived */ - - ret = i2c_attach_client(i2c_client); - if (ret) { - printk(KERN_ERR "cs4270: could not attach codec, " - "I2C address %x, error code %i\n", addr, ret); - goto error; - } - codec->control_data = i2c_client; codec->read = cs4270_read_reg_cache; codec->write = cs4270_i2c_write; @@ -648,6 +605,8 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) goto error; } + i2c_set_clientdata(i2c_client, codec); + return 0; error: -- 1.5.5

2 3

[alsa-devel] [PATCH] Drop Vladimir Barinov's e-mail address
by Jean Delvare 29 Sep '08

29 Sep '08

Every e-mail I've sent to Vladimir Barinov bounced back to me, so I guess he no longer works at MontaVista. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- arch/arm/plat-omap/include/mach/mtd-xip.h | 2 +- drivers/usb/host/ehci-ixp4xx.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/davinci/davinci-pcm.h | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) --- linux-2.6.27-rc7.orig/arch/arm/plat-omap/include/mach/mtd-xip.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/arch/arm/plat-omap/include/mach/mtd-xip.h 2008-09-27 12:56:40.000000000 +0200 @@ -3,7 +3,7 @@ * * Do not include this file directly. It's included from linux/mtd/xip.h * - * Author: Vladimir Barinov <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * * (c) 2005 MontaVista Software, Inc. This file is licensed under the * terms of the GNU General Public License version 2. This program is --- linux-2.6.27-rc7.orig/drivers/usb/host/ehci-ixp4xx.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/drivers/usb/host/ehci-ixp4xx.c 2008-09-27 12:56:30.000000000 +0200 @@ -1,7 +1,7 @@ /* * IXP4XX EHCI Host Controller Driver * - * Author: Vladimir Barinov <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * * Based on "ehci-fsl.c" by Randy Vinson <rvinson(a)mvista.com> * --- linux-2.6.27-rc7.orig/sound/soc/codecs/tlv320aic3x.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/tlv320aic3x.c 2008-09-27 12:56:23.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC TLV320AIC3X codec driver * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * Based on sound/soc/codecs/wm8753.c by Liam Girdwood --- linux-2.6.27-rc7.orig/sound/soc/codecs/tlv320aic3x.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/tlv320aic3x.h 2008-09-27 12:56:19.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC TLV320AIC3X codec driver * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-evm.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-evm.c 2008-09-27 12:56:14.000000000 +0200 @@ -1,7 +1,7 @@ /* * ASoC driver for TI DAVINCI EVM platform * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-i2s.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-i2s.c 2008-09-27 12:56:11.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-i2s.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-i2s.h 2008-09-27 12:56:07.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-pcm.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-pcm.c 2008-09-27 12:56:03.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA PCM interface for the TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-pcm.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-pcm.h 2008-09-27 12:55:59.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA PCM interface for the TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify -- Jean Delvare

3 2

[alsa-devel] [PATCH] hda: appletv support
by Peter Korsgaard 29 Sep '08

29 Sep '08

The AppleTV needs the same handling as the 24" iMac. Signed-off-by: Peter Korsgaard <jacmet(a)sunsite.dk> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c2ff85d..b0b665e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6656,6 +6656,7 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MACPRO; break; case 0x106b1000: /* iMac 24 */ + case 0x106b2800: /* AppleTV */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ -- 1.5.6.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Map ALC662 model for Gigabyte 945GCM-S2L
by Herton Ronaldo Krzesinski 29 Sep '08

29 Sep '08

Use 3stack-6ch-dig ALC662 model for Gigabyte 945GCM-S2L motherboard. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff -p -up linux-2.6.26/sound/pci/hda/patch_realtek.c.orig linux-2.6.26/sound/pci/hda/patch_realtek.c --- linux-2.6.26/sound/pci/hda/patch_realtek.c.orig 2008-09-26 23:11:04.000000000 -0300 +++ linux-2.6.26/sound/pci/hda/patch_realtek.c 2008-09-26 23:22:29.000000000 -0300 @@ -15705,6 +15705,8 @@ static struct snd_pci_quirk alc662_cfg_t SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),

2 1

[alsa-devel] [PATCH] hda: SPDIF stream muting support
by Matthew Ranostay 29 Sep '08

29 Sep '08

Added support for muting S/DPIF outs using the 'Digital Playback Source' muxs on supporting codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..cb419a0 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -138,6 +138,7 @@ struct sigmatel_spec { unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; + unsigned int spdif_mute: 1; /* gpio lines */ unsigned int eapd_mask; @@ -547,10 +548,32 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *smux = &spec->private_smux; unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int err, val; + hda_nid_t nid; - return snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, + err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]); + if (err < 0) + return err; + + if (spec->spdif_mute) { + if (smux_idx == 0) + nid = spec->multiout.dig_out_nid; + else + nid = codec->slave_dig_outs[smux_idx - 1]; + if (spec->cur_smux[smux_idx] == smux->num_items - 1) + val = AMP_OUT_MUTE; + if (smux_idx == 0) + nid = spec->multiout.dig_out_nid; + else + nid = codec->slave_dig_outs[smux_idx - 1]; + /* un/mute SPDIF out */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, val); + } + return 0; } static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1228,6 +1251,15 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } if (spec->num_smuxes > 0) { + int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid); + struct hda_input_mux *smux = &spec->private_smux; + /* check for mute support on SPDIF out */ + if (wcaps & AC_WCAP_OUT_AMP) { + smux->items[smux->num_items].label = "Off"; + smux->items[smux->num_items].index = 0; + smux->num_items++; + spec->spdif_mute = 1; + } stac_smux_mixer.count = spec->num_smuxes; err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&stac_smux_mixer, codec)); @@ -4370,7 +4402,6 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); switch (spec->board_config) { case STAC_HP_M4:

2 1

[alsa-devel] [PATCH 5/5] [RFC] Adds I2C board info for tlvaic23b in osk board file
by Arun KS 29 Sep '08

29 Sep '08

Adding I2C board info required for tlvaic23b i2c chip driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- arch/arm/mach-omap1/board-osk.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 3e766e4..8ddae82 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -258,8 +258,10 @@ static struct i2c_board_info __initdata osk_i2c_board_info[] = { .platform_data = &tps_board, }, + { + I2C_BOARD_INFO("tlvaic23b", 0x1B), + }, /* TODO when driver support is ready: - * - aic23 audio chip at 0x1a * - on Mistral, 24c04 eeprom at 0x50 * - optionally on Mistral, ov9640 camera sensor at 0x30 */ -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] Compiling an Alsa plugin with Intel compiler.
by Motti Daniel 28 Sep '08

28 Sep '08

Hello people! I'm working on an Alsa plugin for the Xandros Linux distribution my plugin runs very well except it consume about 20% of resources from the target machine (eee pc at 1.6Ghz) I'm trying to compile the same code with Intel Compiler (version 10.1.017) (before that it was compiled with gcc 4.1.2) the plugin compiles well but when I try to play music it tells me that the plugin library (the .so file in /usr/lib/alsa-lib/ is not there. I tried to analyze the symbols in the executable and found that the difference between gcc and Intel compiler is a list of symbols with a suffix __FUNCTION__ or __PRETTY_FUNCTION__ is this the problem? any suggestions? can you tell me what are the minimum alsa requirements from a dynamic library so it can be loaded and run as an Alsa plugin? thanks! Motti Daniel.

2 4

[alsa-devel] GeForce 8200 HD audio seems to work!
by Vedran Miletić 28 Sep '08

28 Sep '08

I just installed Linux on ASROCK GeForce 8200 board. Being a relatively new chipset, I had a couple of issues booting older kernels, finally settling for Debian lenny since it had a CD that booted. I was worried about onboard audio, since it's one of thousands of HD audio solutions, and I expected to have problems with it. To make the long story short: 1.0.16 -> only noise, doesn't sound close to track played 1.0.17 -> sound + lots of noise, but you can recongnize the song that is playing takashi's sound-2.6 -> everything works (including front panel) + HDMI audio seeems to be supported Massive props to whoever made it work. I just have two questions: 1) On RS690 I used to have separate device for RS690/RS780 HDMI (that is HDA ATI and HDA ATI HDMI), on NVIDIA I only have HDA NVIDIA, which has subdevice for HDMI, why? 2) alsamixer says: Card: HDA NVidia | Chip: Generic 10de NVIDIA MCP78 HDMI. Shouldn't chip be "Realtek ALC662 rev1" or something like that? Also, why does it say Generic 10de? Where is that read from? Regards, -- Vedran Miletić

1 0

[alsa-devel] [PATCH 1/1] ASoC Blackfin: add I2S DAI support for AD73311
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/bf5xx-i2s.c | 47 ++++++++++++++++++++++++++++------------ 1 files changed, 33 insertions(+), 14 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 43a4092..827587f 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,6 +70,13 @@ static struct sport_param sport_params[2] = { } }; +static u16 sport_req[][7] = { + { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, + { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, + P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, +}; + static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -78,6 +85,14 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + bf5xx_i2s.tcr1 |= TFSR | TCKFE; + bf5xx_i2s.rcr1 |= RFSR | RCKFE; + bf5xx_i2s.tcr2 |= TSFSE; + bf5xx_i2s.rcr2 |= RSFSE; + break; + case SND_SOC_DAIFMT_DSP_A: + bf5xx_i2s.tcr1 |= TFSR; + bf5xx_i2s.rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; @@ -127,14 +142,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; + sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bf5xx_i2s.tcr2 |= 23; bf5xx_i2s.rcr2 |= 23; + sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: bf5xx_i2s.tcr2 |= 31; bf5xx_i2s.rcr2 |= 31; + sport_handle->wdsize = 4; break; } @@ -145,17 +163,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * need to configure both of them at the time when the first * stream is opened. * - * CPU DAI format:I2S, slave mode. + * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, RFSR | RCKFE, - RSFSE|bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, - TSFSE|bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -174,13 +192,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) static int bf5xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, - }; - pr_debug("%s enter\n", __func__); if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); @@ -198,6 +209,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("%s enter\n", __func__); + peripheral_free_list(&sport_req[sport_num][0]); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct platform_device *dev, struct snd_soc_dai *dai) @@ -263,15 +281,16 @@ struct snd_soc_dai bf5xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, + .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, -- 1.5.6

2 1

[alsa-devel] [PATCH 1/1] ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled (v2)
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> With MMAP enabled (DMA mode) on the AD1981, there is +/- 250ms of delay between writing data to alsa and audio starts coming out of the AD1981. Copy more data to local buffer before starting DMA v1-v2: - remove the changelogs in source file because we have git log Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 42 +++++++++++++++++++++++++++++----- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-sport.h | 2 + 3 files changed, 37 insertions(+), 8 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907..25e50d2 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { bf5xx_ac97_to_pcm( (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, @@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data) struct snd_pcm_substream *pcm = data; #if defined(CONFIG_SND_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + memset(runtime->dma_area, 0, runtime->buffer_size); snd_pcm_lib_free_pages(substream); return 0; } @@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) * SPORT working in TMD mode(include AC97). */ #if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bf5xx_mmap_copy(substream, runtime->period_size); + snd_pcm_period_elapsed(substream); + sport->tx_delay_pos = 0; sport_tx_start(sport); + } else sport_rx_start(sport); break; @@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) #if defined(CONFIG_SND_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + pr_debug("%s enter\n", __func__); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + + return 0; +} + #ifdef CONFIG_SND_MMAP_SUPPORT static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -272,6 +299,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, + .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e31..5e5aafb 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -129,7 +129,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) struct ac97_frame *nextwrite; sport_incfrag(sport, &nextfrag, 1); - sport_incfrag(sport, &nextfrag, 1); nextwrite = (struct ac97_frame *)(sport->tx_buf + \ nextfrag * sport->tx_fragsize); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c16345..fcadcc0 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -123,6 +123,8 @@ struct sport_device { int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; -- 1.5.6

1 0

[alsa-devel] [PATCH 1/1] ASoC Blackfin: add asoc ad73311 driver supporting in Blackfin boards (v2)
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v1-v2: - split out I2S DAI changes to a seperated patch Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/Kconfig | 16 +++ sound/soc/blackfin/Makefile | 3 +- sound/soc/blackfin/bf5xx-ad73311.c | 240 ++++++++++++++++++++++++++++++++++++ 3 files changed, 258 insertions(+), 1 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad73311.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index f98331d..dc00620 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602 help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_BF5XX_SOC_AD73311 + tristate "SoC AD73311 Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD73311 + help + Say Y if you want to add support for AD73311 codec on Blackfin. + +config SND_BFIN_AD73311_SE + int "PF pin for AD73311L Chip Select" + depends on SND_BF5XX_SOC_AD73311 + default 4 + help + Enter the GPIO used to control AD73311's SE pin. Acceptable + values are 0 to 7 + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN && SND_SOC diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 9ea8bd9..97bb37a 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o - +snd-ad73311-objs := bf5xx-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o +obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c new file mode 100644 index 0000000..622c9b9 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -0,0 +1,240 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad73311.c + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Thur Sep 25 2008 + * Description: Board driver for ad73311 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad73311.h" +#include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" +#include "bf5xx-i2s.h" + +#if CONFIG_SND_BF5XX_SPORT_NUM == 0 +#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT +#else +#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT +#endif + +#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE + +static struct snd_soc_machine bf5xx_ad73311; + +static int snd_ad73311_startup(void) +{ + pr_debug("%s enter\n", __func__); + + /* Pull up SE pin on AD73311L */ + gpio_set_value(GPIO_SE, 1); + return 0; +} + +static int snd_ad73311_configure(void) +{ + unsigned short ctrl_regs[6]; + unsigned short status = 0; + int count = 0; + + /* DMCLK = MCLK = 16.384 MHz + * SCLK = DMCLK/8 = 2.048 MHz + * Sample Rate = DMCLK/2048 = 8 KHz + */ + ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ + REGB_SCDIV(0) | REGB_DIRATE(0); + ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ + REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; + ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ + REGD_IGS(2); + ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); + ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; + ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; + + local_irq_disable(); + snd_ad73311_startup(); + udelay(1); + + bfin_write_SPORT_TCR1(TFSR); + bfin_write_SPORT_TCR2(0xF); + SSYNC(); + + /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to + * FIFO before enable SPORT to transfer the data + */ + for (count = 0; count < 6; count++) + bfin_write_SPORT_TX16(ctrl_regs[count]); + SSYNC(); + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); + SSYNC(); + + /* When TUVF is set, the data is already send out */ + while (!(status & TUVF) && count++ < 10000) { + udelay(1); + status = bfin_read_SPORT_STAT(); + SSYNC(); + } + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); + SSYNC(); + local_irq_enable(); + + if (count == 10000) { + printk(KERN_ERR "ad73311: failed to configure codec\n"); + return -1; + } + return 0; +} + +static int bf5xx_probe(struct platform_device *pdev) +{ + int err; + if (gpio_request(GPIO_SE, "AD73311_SE")) { + printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); + return -EBUSY; + } + + gpio_direction_output(GPIO_SE, 0); + + err = snd_ad73311_configure(); + if (err < 0) + return -EFAULT; + + return 0; +} + +static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + pr_debug("%s enter\n", __func__); + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + pr_debug("%s rate %d format %x\n", __func__, params_rate(params), + params_format(params)); + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + + +static struct snd_soc_ops bf5xx_ad73311_ops = { + .startup = bf5xx_ad73311_startup, + .hw_params = bf5xx_ad73311_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad73311_dai = { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai = &bf5xx_i2s_dai, + .codec_dai = &ad73311_dai, + .ops = &bf5xx_ad73311_ops, +}; + +static struct snd_soc_machine bf5xx_ad73311 = { + .name = "bf5xx_ad73311", + .probe = bf5xx_probe, + .dai_link = &bf5xx_ad73311_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad73311_snd_devdata = { + .machine = &bf5xx_ad73311, + .platform = &bf5xx_i2s_soc_platform, + .codec_dev = &soc_codec_dev_ad73311, +}; + +static struct platform_device *bf52x_ad73311_snd_device; + +static int __init bf5xx_ad73311_init(void) +{ + int ret; + + pr_debug("%s enter\n", __func__); + bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf52x_ad73311_snd_device) + return -ENOMEM; + + platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; + ret = platform_device_add(bf52x_ad73311_snd_device); + + if (ret) + platform_device_put(bf52x_ad73311_snd_device); + + return ret; +} + +static void __exit bf5xx_ad73311_exit(void) +{ + pr_debug("%s enter\n", __func__); + platform_device_unregister(bf52x_ad73311_snd_device); +} + +module_init(bf5xx_ad73311_init); +module_exit(bf5xx_ad73311_exit); + +/* Module information */ +MODULE_AUTHOR("Cliff Cai"); +MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); +MODULE_LICENSE("GPL"); + -- 1.5.6

1 0

[alsa-devel] [PATCH 1/1] ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> With MMAP enabled (DMA mode) on the AD1981, there is +/- 250ms of delay between writing data to alsa and audio starts coming out of the AD1981. Copy more data to local buffer before starting DMA Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 42 +++++++++++++++++++++++++++++----- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-sport.h | 2 + 3 files changed, 37 insertions(+), 8 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907..25e50d2 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { bf5xx_ac97_to_pcm( (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, @@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data) struct snd_pcm_substream *pcm = data; #if defined(CONFIG_SND_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + memset(runtime->dma_area, 0, runtime->buffer_size); snd_pcm_lib_free_pages(substream); return 0; } @@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) * SPORT working in TMD mode(include AC97). */ #if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bf5xx_mmap_copy(substream, runtime->period_size); + snd_pcm_period_elapsed(substream); + sport->tx_delay_pos = 0; sport_tx_start(sport); + } else sport_rx_start(sport); break; @@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) #if defined(CONFIG_SND_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + pr_debug("%s enter\n", __func__); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + + return 0; +} + #ifdef CONFIG_SND_MMAP_SUPPORT static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -272,6 +299,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, + .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e31..5e5aafb 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -129,7 +129,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) struct ac97_frame *nextwrite; sport_incfrag(sport, &nextfrag, 1); - sport_incfrag(sport, &nextfrag, 1); nextwrite = (struct ac97_frame *)(sport->tx_buf + \ nextfrag * sport->tx_fragsize); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c16345..fcadcc0 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -123,6 +123,8 @@ struct sport_device { int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; -- 1.5.6

1 1

[alsa-devel] [PATCH 1/1] ASoC Blackfin: add asoc ad73311 driver supporting in Blackfin boards
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/Kconfig | 16 +++ sound/soc/blackfin/Makefile | 3 +- sound/soc/blackfin/bf5xx-ad73311.c | 240 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-i2s.c | 52 ++++++-- 4 files changed, 296 insertions(+), 15 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad73311.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index f98331d..dc00620 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602 help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_BF5XX_SOC_AD73311 + tristate "SoC AD73311 Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD73311 + help + Say Y if you want to add support for AD73311 codec on Blackfin. + +config SND_BFIN_AD73311_SE + int "PF pin for AD73311L Chip Select" + depends on SND_BF5XX_SOC_AD73311 + default 4 + help + Enter the GPIO used to control AD73311's SE pin. Acceptable + values are 0 to 7 + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN && SND_SOC diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 9ea8bd9..97bb37a 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o - +snd-ad73311-objs := bf5xx-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o +obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c new file mode 100644 index 0000000..622c9b9 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -0,0 +1,240 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad73311.c + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Thur Sep 25 2008 + * Description: Board driver for ad73311 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad73311.h" +#include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" +#include "bf5xx-i2s.h" + +#if CONFIG_SND_BF5XX_SPORT_NUM == 0 +#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT +#else +#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT +#endif + +#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE + +static struct snd_soc_machine bf5xx_ad73311; + +static int snd_ad73311_startup(void) +{ + pr_debug("%s enter\n", __func__); + + /* Pull up SE pin on AD73311L */ + gpio_set_value(GPIO_SE, 1); + return 0; +} + +static int snd_ad73311_configure(void) +{ + unsigned short ctrl_regs[6]; + unsigned short status = 0; + int count = 0; + + /* DMCLK = MCLK = 16.384 MHz + * SCLK = DMCLK/8 = 2.048 MHz + * Sample Rate = DMCLK/2048 = 8 KHz + */ + ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ + REGB_SCDIV(0) | REGB_DIRATE(0); + ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ + REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; + ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ + REGD_IGS(2); + ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); + ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; + ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; + + local_irq_disable(); + snd_ad73311_startup(); + udelay(1); + + bfin_write_SPORT_TCR1(TFSR); + bfin_write_SPORT_TCR2(0xF); + SSYNC(); + + /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to + * FIFO before enable SPORT to transfer the data + */ + for (count = 0; count < 6; count++) + bfin_write_SPORT_TX16(ctrl_regs[count]); + SSYNC(); + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); + SSYNC(); + + /* When TUVF is set, the data is already send out */ + while (!(status & TUVF) && count++ < 10000) { + udelay(1); + status = bfin_read_SPORT_STAT(); + SSYNC(); + } + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); + SSYNC(); + local_irq_enable(); + + if (count == 10000) { + printk(KERN_ERR "ad73311: failed to configure codec\n"); + return -1; + } + return 0; +} + +static int bf5xx_probe(struct platform_device *pdev) +{ + int err; + if (gpio_request(GPIO_SE, "AD73311_SE")) { + printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); + return -EBUSY; + } + + gpio_direction_output(GPIO_SE, 0); + + err = snd_ad73311_configure(); + if (err < 0) + return -EFAULT; + + return 0; +} + +static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + pr_debug("%s enter\n", __func__); + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + pr_debug("%s rate %d format %x\n", __func__, params_rate(params), + params_format(params)); + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + + +static struct snd_soc_ops bf5xx_ad73311_ops = { + .startup = bf5xx_ad73311_startup, + .hw_params = bf5xx_ad73311_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad73311_dai = { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai = &bf5xx_i2s_dai, + .codec_dai = &ad73311_dai, + .ops = &bf5xx_ad73311_ops, +}; + +static struct snd_soc_machine bf5xx_ad73311 = { + .name = "bf5xx_ad73311", + .probe = bf5xx_probe, + .dai_link = &bf5xx_ad73311_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad73311_snd_devdata = { + .machine = &bf5xx_ad73311, + .platform = &bf5xx_i2s_soc_platform, + .codec_dev = &soc_codec_dev_ad73311, +}; + +static struct platform_device *bf52x_ad73311_snd_device; + +static int __init bf5xx_ad73311_init(void) +{ + int ret; + + pr_debug("%s enter\n", __func__); + bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf52x_ad73311_snd_device) + return -ENOMEM; + + platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; + ret = platform_device_add(bf52x_ad73311_snd_device); + + if (ret) + platform_device_put(bf52x_ad73311_snd_device); + + return ret; +} + +static void __exit bf5xx_ad73311_exit(void) +{ + pr_debug("%s enter\n", __func__); + platform_device_unregister(bf52x_ad73311_snd_device); +} + +module_init(bf5xx_ad73311_init); +module_exit(bf5xx_ad73311_exit); + +/* Module information */ +MODULE_AUTHOR("Cliff Cai"); +MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 43a4092..8ccc778 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -24,6 +24,11 @@ * along with this program; if not, see the file COPYING, or write * to the Free Software Foundation, Inc., * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + * + * + * Revision history + * 6th June 2008 Initial version + * 25th Sep 2008 add DAI support for AD73311 */ #include <linux/init.h> @@ -70,6 +75,13 @@ static struct sport_param sport_params[2] = { } }; +static u16 sport_req[][7] = { + { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, + { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, + P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, +}; + static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -78,6 +90,14 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + bf5xx_i2s.tcr1 |= TFSR | TCKFE; + bf5xx_i2s.rcr1 |= RFSR | RCKFE; + bf5xx_i2s.tcr2 |= TSFSE; + bf5xx_i2s.rcr2 |= RSFSE; + break; + case SND_SOC_DAIFMT_DSP_A: + bf5xx_i2s.tcr1 |= TFSR; + bf5xx_i2s.rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; @@ -127,14 +147,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; + sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bf5xx_i2s.tcr2 |= 23; bf5xx_i2s.rcr2 |= 23; + sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: bf5xx_i2s.tcr2 |= 31; bf5xx_i2s.rcr2 |= 31; + sport_handle->wdsize = 4; break; } @@ -145,17 +168,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * need to configure both of them at the time when the first * stream is opened. * - * CPU DAI format:I2S, slave mode. + * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, RFSR | RCKFE, - RSFSE|bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, - TSFSE|bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -174,13 +197,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) static int bf5xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, - }; - pr_debug("%s enter\n", __func__); if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); @@ -198,6 +214,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("%s enter\n", __func__); + peripheral_free_list(&sport_req[sport_num][0]); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct platform_device *dev, struct snd_soc_dai *dai) @@ -263,15 +286,16 @@ struct snd_soc_dai bf5xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, + .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, -- 1.5.6

2 2

[alsa-devel] [PATCH 1/1] ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> With MMAP enabled (DMA mode) on the AD1981, there is +/- 250ms of delay between writing data to alsa and audio starts coming out of the AD1981. Copy more data to local buffer before starting DMA Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 46 +++++++++++++++++++++++++++++----- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-sport.h | 2 + 3 files changed, 41 insertions(+), 8 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907..d61febe 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -24,6 +24,10 @@ * along with this program; if not, see the file COPYING, or write * to the Free Software Foundation, Inc., * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + * + * Revision history + * 6th June 2008 Initial version + * 25th Sep 2008 fix latency when enable mmap */ #include <linux/module.h> @@ -56,6 +60,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { bf5xx_ac97_to_pcm( (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, @@ -72,7 +77,15 @@ static void bf5xx_dma_irq(void *data) struct snd_pcm_substream *pcm = data; #if defined(CONFIG_SND_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -114,6 +127,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + memset(runtime->dma_area, 0, runtime->buffer_size); snd_pcm_lib_free_pages(substream); return 0; } @@ -127,16 +144,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) * SPORT working in TMD mode(include AC97). */ #if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,8 +176,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bf5xx_mmap_copy(substream, runtime->period_size); + snd_pcm_period_elapsed(substream); + sport->tx_delay_pos = 0; sport_tx_start(sport); + } else sport_rx_start(sport); break; @@ -198,7 +214,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) #if defined(CONFIG_SND_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,6 +253,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + pr_debug("%s enter\n", __func__); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + + return 0; +} + #ifdef CONFIG_SND_MMAP_SUPPORT static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -272,6 +303,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, + .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e31..5e5aafb 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -129,7 +129,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) struct ac97_frame *nextwrite; sport_incfrag(sport, &nextfrag, 1); - sport_incfrag(sport, &nextfrag, 1); nextwrite = (struct ac97_frame *)(sport->tx_buf + \ nextfrag * sport->tx_fragsize); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c16345..fcadcc0 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -123,6 +123,8 @@ struct sport_device { int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; -- 1.5.6

2 2

[alsa-devel] [PATCH 1/1] ASoC codec: AD73311 audio codec driver
by Bryan Wu 27 Sep '08

27 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad73311.c | 107 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad73311.h | 90 +++++++++++++++++++++++++++++++++++++ 4 files changed, 203 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ad73311.c create mode 100644 sound/soc/codecs/ad73311.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0b9869..0507fcf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -3,6 +3,7 @@ config SND_SOC_ALL_CODECS depends on I2C select SPI select SPI_MASTER + select SND_SOC_AD73311 select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 @@ -34,6 +35,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1980 tristate +config SND_SOC_AD73311 + tristate + config SND_SOC_AK4535 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f977978..0731844 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o +snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o @@ -20,6 +21,7 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c new file mode 100644 index 0000000..37af860 --- /dev/null +++ b/sound/soc/codecs/ad73311.c @@ -0,0 +1,107 @@ +/* + * ad73311.c -- ALSA Soc AD73311 codec support + * + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai(a)analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 25th Sep 2008 Initial version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad73311.h" + +struct snd_soc_dai ad73311_dai = { + .name = "AD73311", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +EXPORT_SYMBOL_GPL(ad73311_dai); + +static int ad73311_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "AD73311"; + codec->owner = THIS_MODULE; + codec->dai = &ad73311_dai; + codec->num_dai = 1; + socdev->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int ad73311_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad73311 = { + .probe = ad73311_soc_probe, + .remove = ad73311_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); + +MODULE_DESCRIPTION("ASoC ad73311 driver"); +MODULE_AUTHOR("Cliff Cai "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h new file mode 100644 index 0000000..507ce0c --- /dev/null +++ b/sound/soc/codecs/ad73311.h @@ -0,0 +1,90 @@ +/* + * File: sound/soc/codec/ad73311.h + * Based on: + * Author: Cliff Cai <cliff.cai(a)analog.com> + * + * Created: Thur Sep 25, 2008 + * Description: definitions for AD73311 registers + * + * + * Modified: + * Copyright 2006 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD73311_H__ +#define __AD73311_H__ + +#define AD_CONTROL 0x8000 +#define AD_DATA 0x0000 +#define AD_READ 0x4000 +#define AD_WRITE 0x0000 + +/* Control register A */ +#define CTRL_REG_A (0 << 8) + +#define REGA_MODE_PRO 0x00 +#define REGA_MODE_DATA 0x01 +#define REGA_MODE_MIXED 0x03 +#define REGA_DLB 0x04 +#define REGA_SLB 0x08 +#define REGA_DEVC(x) ((x & 0x7) << 4) +#define REGA_RESET 0x80 + +/* Control register B */ +#define CTRL_REG_B (1 << 8) + +#define REGB_DIRATE(x) (x & 0x3) +#define REGB_SCDIV(x) ((x & 0x3) << 2) +#define REGB_MCDIV(x) ((x & 0x7) << 4) +#define REGB_CEE (1 << 7) + +/* Control register C */ +#define CTRL_REG_C (2 << 8) + +#define REGC_PUDEV (1 << 0) +#define REGC_PUADC (1 << 3) +#define REGC_PUDAC (1 << 4) +#define REGC_PUREF (1 << 5) +#define REGC_REFUSE (1 << 6) + +/* Control register D */ +#define CTRL_REG_D (3 << 8) + +#define REGD_IGS(x) (x & 0x7) +#define REGD_RMOD (1 << 3) +#define REGD_OGS(x) ((x & 0x7) << 4) +#define REGD_MUTE (x << 7) + +/* Control register E */ +#define CTRL_REG_E (4 << 8) + +#define REGE_DA(x) (x & 0x1f) +#define REGE_IBYP (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 8) + +#define REGF_SEEN (1 << 5) +#define REGF_INV (1 << 6) +#define REGF_ALB (1 << 7) + +extern struct snd_soc_dai ad73311_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad73311; +#endif -- 1.5.6

2 1

[alsa-devel] [PATCH] ALSA: hda - Map ALC662 model for Foxconn 45CMX/45GMX/45CMX-K
by Herton Ronaldo Krzesinski 27 Sep '08

27 Sep '08

Use 3stack-6ch-dig ALC662 model for Foxconn 45CMX/45GMX/45CMX-K motherboard. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff -p -up linux-2.6.26/sound/pci/hda/patch_realtek.c.orig linux-2.6.26/sound/pci/hda/patch_realtek.c --- linux-2.6.26/sound/pci/hda/patch_realtek.c.orig 2008-09-26 23:27:24.000000000 -0300 +++ linux-2.6.26/sound/pci/hda/patch_realtek.c 2008-09-26 23:28:15.000000000 -0300 @@ -15702,6 +15702,8 @@ static struct snd_pci_quirk alc662_cfg_t SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),

1 0

[alsa-devel] ALSA:ATMEL: common arch for audio via ssc(patch 3/3: adding directories)
by Sedji Gaouaou 26 Sep '08

26 Sep '08

This last patch contains the board specific files. I have not changed playpaq_wm8510.c, and I did few changes on eti_b1_wm8731.c. Regards, Sedji diff --git a/sound/soc/atmel/eti_b1_wm8731.c b/sound/soc/atmel/eti_b1_wm8731.c new file mode 100644 index 0000000..afa1aaf --- /dev/null +++ b/sound/soc/atmel/eti_b1_wm8731.c @@ -0,0 +1,353 @@ +/* + * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. + * + * Author: Frank Mandarino <fmandarino(a)endrelia.com> + * Endrelia Technologies Inc. + * Created: Mar 29, 2006 + * + * Based on corgi.c by: + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Richard Purdie <richard(a)openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel_pcm.h" +#include "atmel_ssc_dai.h" + +#if 0 +#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) +#else +#define DBG(x...) +#endif + +static struct clk *pck1_clk; +static struct clk *pllb_clk; + + +static int eti_b1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* cpu clock is the AT91 master clock sent to the SSC */ + ret = snd_soc_dai_set_sysclk(cpu_dai, ATMEL_SYSCLK_MCK, + 60000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* codec system clock is supplied by PCK1, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Start PCK1 clock. */ + clk_enable(pck1_clk); + DBG("pck1 started\n"); + + return 0; +} + +static void eti_b1_shutdown(struct snd_pcm_substream *substream) +{ + /* Stop PCK1 clock. */ + clk_disable(pck1_clk); + DBG("pck1 stopped\n"); +} + +static int eti_b1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + +#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE + unsigned int rate; + int cmr_div, period; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ + break; + case 32000: + cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ + period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ + break; + case 48000: + cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ + period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on ETI-B1 board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + +#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ + /* + * Codec in Master Mode. + */ + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + +#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ + + return 0; +} + +static struct snd_soc_ops eti_b1_ops = { + .startup = eti_b1_startup, + .hw_params = eti_b1_hw_params, + .shutdown = eti_b1_shutdown, +}; + + +static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. + */ +static int eti_b1_wm8731_init(struct snd_soc_codec *codec) +{ + DBG("eti_b1_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, + ARRAY_SIZE(eti_b1_dapm_widgets)); + + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link eti_b1_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[1], + .codec_dai = &wm8731_dai, + .init = eti_b1_wm8731_init, + .ops = &eti_b1_ops, +}; + +static struct snd_soc_machine snd_soc_machine_eti_b1 = { + .name = "ETI_B1_WM8731", + .dai_link = &eti_b1_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data eti_b1_wm8731_setup = { + .i2c_address = 0x1a, +}; + +static struct snd_soc_device eti_b1_snd_devdata = { + .machine = &snd_soc_machine_eti_b1, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &eti_b1_wm8731_setup, +}; + +static struct platform_device *eti_b1_snd_device; + +static int __init eti_b1_init(void) +{ + int ret; + struct atmel_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + /* + * Request SSC device + */ + ssc = ssc_request(1); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + eti_b1_snd_device = platform_device_alloc("soc-audio", -1); + if (!eti_b1_snd_device) { + DBG("platform device allocation failed\n"); + ret = -ENOMEM; + goto fail_io_unmap; + } + + platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); + eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; + + ret = platform_device_add(eti_b1_snd_device); + if (ret) { + DBG("platform device add failed\n"); + platform_device_put(eti_b1_snd_device); + goto fail_io_unmap; + } + + at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ + at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ + at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ + at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ +/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ + at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ + + /* + * Set PCK1 parent to PLLB and its rate to 12 Mhz. + */ + pllb_clk = clk_get(NULL, "pllb"); + pck1_clk = clk_get(NULL, "pck1"); + + clk_set_parent(pck1_clk, pllb_clk); + clk_set_rate(pck1_clk, 12000000); + + DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); + + /* assign the GPIO pin to PCK1 */ + at91_set_B_periph(AT91_PIN_PA24, 0); + +#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE + printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); +#else + printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); +#endif + return ret; +err_ssc: + return ret; +fail_io_unmap: + iounmap(ssc->base); +fail_release_mem: + release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); + return ret; +} + +static void __exit eti_b1_exit(void) +{ + struct atmel_ssc_periph *ssc_p = eti_b1_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + clk_put(pck1_clk); + clk_put(pllb_clk); + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(eti_b1_snd_device); +} + +module_init(eti_b1_init); +module_exit(eti_b1_exit); + +/* Module information */ +MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); +MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..3f32621 --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,522 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..6b1aa3a --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,328 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL");

5 11

[alsa-devel] [PATCH] hda: HP laptop quirk
by Matthew Ranostay 26 Sep '08

26 Sep '08

Added HP laptop SND_PCI_QUIRK for STAC_HP_M4. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..7db954a 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1650,6 +1650,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, + "unknown HP", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,

2 1

[alsa-devel] [PATCH] hda: 92xx S/PDIF In support
by Matthew Ranostay 26 Sep '08

26 Sep '08

Add support for S/PDIF IN on reference boards quirks. Also disable the IEC958 capture switch on boards with GPIO0 enabled. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..128b8a1 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1245,7 +1245,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; spec->multiout.share_spdif = 1; } - if (spec->dig_in_nid) { + if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); if (err < 0) return err; @@ -4075,10 +4075,6 @@ again: memcpy(&spec->private_dimux, &stac92hd73xx_dmux, sizeof(stac92hd73xx_dmux)); - /* GPIO0 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; - spec->gpio_data = 0x01; - switch (spec->board_config) { case STAC_DELL_M6: spec->init = dell_eq_core_init; @@ -4118,6 +4114,11 @@ again: spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); } + if (spec->board_config > STAC_92HD73XX_REF) { + /* GPIO0 High = Enable EAPD */ + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; + spec->gpio_data = 0x01; + } spec->dinput_mux = &spec->private_dimux; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); @@ -4354,10 +4355,12 @@ again: spec->aloopback_mask = 0x20; spec->aloopback_shift = 0; - /* GPIO0 High = EAPD */ - spec->gpio_mask = 0x01; - spec->gpio_dir = 0x01; - spec->gpio_data = 0x01; + if (spec->board_config > STAC_92HD71BXX_REF) { + /* GPIO0 = EAPD */ + spec->gpio_mask = 0x01; + spec->gpio_dir = 0x01; + spec->gpio_data = 0x01; + } spec->powerdown_adcs = 1; spec->digbeep_nid = 0x26; @@ -4599,9 +4602,11 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; default: - /* GPIO0 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; - spec->gpio_data = 0x01; + if (spec->board_config > STAC_D965_REF) { + /* GPIO0 High = Enable EAPD */ + spec->eapd_mask = spec->gpio_mask = 0x01; + spec->gpio_dir = spec->gpio_data = 0x01; + } spec->num_dmics = 0; spec->init = stac927x_core_init; @@ -4715,6 +4720,9 @@ static int patch_stac9205(struct hda_codec *codec) */ spec->gpio_data = 0x01; break; + case STAC_9205_REF: + /* SPDIF-In enabled */ + break; default: /* GPIO0 High = EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;

2 1

[alsa-devel] [PATCH 1/1] ALSA: ASoC: Fix wrong param when call snd_soc_test_bits
by Richard Zhao 26 Sep '08

26 Sep '08

From: Richard Zhao <richard(a)richard-laptop.(none)> Fix wrong param when call snd_soc_test_bits in function dapm_mux_update_power. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

1 0

[alsa-devel] ALSA:ATMEL: common arch for audio via ssc(patch 2/3: adding directories)
by Sedji Gaouaou 26 Sep '08

26 Sep '08

This patch contains: The Kconfig and Makefile, plus the interface files(atmel_ssc_dai and atmel-pcm). Regards, Sedji diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index f743530..030d9f2 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,8 +22,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 933a66d..6a38fc7 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..6a2e79f --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,61 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the AT91 SSC interface. You will also need + to select the audio interfaces to support below. + + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_ETI_B1_WM8731 + tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" + depends on SND_ATMEL_SOC && (MACH_ETI_B1 || MACH_ETI_C1) + select SND_AT91_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. + +config SND_AT91_SOC_ETI_SLAVE + bool "Run codec in slave Mode on Endrelia boards" + depends on SND_AT91_SOC_ETI_B1_WM8731 + default n + help + Say Y if you want to run with the AT91 SSC generating the BCLK + and LRC signals on Endrelia boards. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..00c5cd8 --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,17 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..67e9bfd --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,494 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + /*buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL);*/ + printk(KERN_DEBUG "atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + printk(KERN_DEBUG "atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + printk(KERN_DEBUG "atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + printk(KERN_DEBUG "atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + printk(KERN_DEBUG "sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + /*return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot);*/ +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + printk(KERN_DEBUG "at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + /*dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr);*/ + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable) + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..9ab4653 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,804 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +static unsigned int atmel_ssc_sysclk; + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dev_dbg(rtd->socdev->dev, "ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + dir = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + 0 : 1); + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + dev_dbg(rtd->socdev->dev, "%s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + dev_dbg(rtd->socdev->dev, + "atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + +/* + * Record the SSC system clock rate. + */ +static int atmel_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* + * The only clock supplied to the SSC is the AT91 master clock, + * which is only used if the SSC is generating BCLK and/or + * LRC clocks. + */ + switch (clk_id) { + case ATMEL_SYSCLK_MCK: + atmel_ssc_sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + dir = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + 0 : 1); + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + dev_dbg(rtd->socdev->dev, "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + dev_dbg(rtd->socdev->dev, "atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + dev_dbg(rtd->socdev->dev, + "Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + dev_dbg(rtd->socdev->dev, + "atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + dir = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + 0 : 1); + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + printk(KERN_DEBUG "%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_sysclk = atmel_ssc_set_dai_sysclk, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_sysclk = atmel_ssc_set_dai_sysclk, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_sysclk = atmel_ssc_set_dai_sysclk, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..78d3987 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,122 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */

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Re: [alsa-devel] [PATCH] add the nvidia HDMI codec driverfor MCP77/79
by Takashi Iwai 26 Sep '08

26 Sep '08

At Wed, 17 Sep 2008 17:00:04 +0800, peerchen wrote: > > new patch base on git tree. > > Signed-off-by: Wei Ni <wni(a)nvidia.com> > Signed-off-by: Peer Chen <peerchen(a)gmail.com> Thanks! But, the patch looks like no "rebase". You secretly added a new function call that wasn't in your previous patch, namely... > diff -uprN -X linux-2.6-tiwai-sound/Documentation/dontdiff linux-2.6-tiwai-sound/sound/pci/hda/hda_intel.c linux-2.6-tiwai-sound-niwei/sound/pci/hda/hda_intel.c > --- linux-2.6-tiwai-sound/sound/pci/hda/hda_intel.c 2008-09-10 17:40:52.000000000 +0800 > +++ linux-2.6-tiwai-sound-niwei/sound/pci/hda/hda_intel.c 2008-09-10 17:49:31.000000000 +0800 > @@ -1220,6 +1220,9 @@ static int __devinit azx_codec_create(st > if (err < 0) > return err; > > + if (chip->driver_type == AZX_DRIVER_NVIDIA) > + chip->bus->needs_damn_long_delay = 1; > + Do we really need this inevitably? I don't think so -- otherwise I would have far more bug reports. This flag is the last resort and should be avoided as much as possible. This results in a significant slow down of suspend/resume speed, for example. If it's really needed with some devices and there is no other way to fix it, add this flag rather in the codec-initialization part. Also, if you resend a patch, please add the original patch description again. This helps my patch work a lot indeed. thanks, Takashi

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[alsa-devel] [PATCH] ALSA: usb-audio: add support for E-Mu Tracker Pre
by Eran Tromer 26 Sep '08

26 Sep '08

Add support for the E-Mu "Tracker Pre" USB sound card, following the example of the (very similar) E-Mu 0202 and E-Mu 0404 USB. As with the 0202 and 0404 USB, functionality is very limited: just a couple of sample rates, no volume/mute control, etc. Signed-off-by: Eran Tromer <eran(a)tromer.org> --- sound/usb/usbaudio.c | 3 ++- sound/usb/usbquirks.h | 7 +++++++ 2 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 410be4a..0cc6fd5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -481,7 +481,7 @@ static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, } /* - * process after E-Mu 0202/0404 high speed playback sync complete + * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete * * These devices return the number of samples per packet instead of the number * of samples per microframe. @@ -2261,6 +2261,7 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo switch (as->chip->usb_id) { case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; break; } diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 82a8d14..b90d98b 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -62,6 +62,13 @@ .idProduct = 0x3f04, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu Tracker Pre */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f0a, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface

2 1

[alsa-devel] [PATCH] alsa-plugins: Start PA event loop after we have created the context.
by Lennart Poettering 26 Sep '08

26 Sep '08

Heya! We shouldn't be passing the main loop object to the pa_context while that loop is running and not locked. Push the main loop startup after we created the main loop. --- pulse/pulse.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index 617851e..022ca55 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -188,9 +188,6 @@ snd_pulse_t *pulse_new(void) if (!p->mainloop) goto fail; - if (pa_threaded_mainloop_start(p->mainloop) < 0) - goto fail; - if (pa_get_binary_name(proc, sizeof(proc))) snprintf(buf, sizeof(buf), "ALSA plug-in [%s]", pa_path_get_filename(proc)); @@ -204,6 +201,9 @@ snd_pulse_t *pulse_new(void) pa_context_set_state_callback(p->context, context_state_cb, p); + if (pa_threaded_mainloop_start(p->mainloop) < 0) + goto fail; + return p; fail: -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] Guide to Linux Sound APIs
by Lennart Poettering 26 Sep '08

26 Sep '08

Heya! At the audio microconf at the Linux plumbers conference one thing became very clear: it is very difficult for programmers to figure out which audio API to use for which purpose and which API not to use when doing audio programming on Linux. Someone needed to sit down and write up a small guide. And that's what I just finished doing. I'd thus like to draw your attention to this new (long) blog story of mine containing this guide: http://0pointer.de/blog/projects/guide-to-sound-apis I'd be very thankful for comments, especially from you Takashi and Jaroslav since it contains quite some details about the "safe" subset of the ALSA API! I'm thankful for all kinds of input! Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

4 6

[PATCH] Missing file for NVIDIA HDMI HD-audio support.
by Pawel Krzesniak 26 Sep '08

26 Sep '08

Missing file for NVIDIA HDMI HD-audio support. Signed-off-by: Pawel Krzesniak <pawel.krzesniak(a)gmail.com> --- pci/hda/patch_nvhdmi.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) create mode 100644 pci/hda/patch_nvhdmi.c diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c new file mode 100644 index 0000000..3e700fd --- /dev/null +++ b/pci/hda/patch_nvhdmi.c @@ -0,0 +1,3 @@ +#define __NO_VERSION__ +#include "adriver.h" +#include "../../alsa-kernel/pci/hda/patch_nvhdmi.c" -- 1.5.4.3 --------------020304060503040802040300--

1 0

[alsa-devel] ALSA:ATMEL: common arch for audio via ssc(patch 1/3: removing directories)
by Sedji Gaouaou 26 Sep '08

26 Sep '08

Hi, I am trying to merge two directories: sound/soc/at32 and sound/soc/at91, to have a single atmel interface. I have written a patch but I can't test it on avr32 boards, nor eti board, so if any body can test it, it would be appreciated. I have changed the file eti_b1_wm8731.c, because the ssc was allocated in it and it should be allocated in the board file. So there are some changes to do in the eti board file otherwise the dai will not work for the eti board. I didn't change the i2c initialization since I don't know what is the status of the patch(I tried this one: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commitdiff;… but I not sure it is complete?? Regards, Sedji diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig deleted file mode 100644 index b0765e8..0000000 --- a/sound/soc/at32/Kconfig +++ /dev/null @@ -1,34 +0,0 @@ -config SND_AT32_SOC - tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" - depends on AVR32 && SND_SOC - help - Say Y or M if you want to add support for codecs attached to - the AT32 SSC interface. You will also need to - to select the audio interfaces to support below. - - -config SND_AT32_SOC_SSC - tristate - - - -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_AT32_SOC && BOARD_PLAYPAQ - select SND_AT32_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - - - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile deleted file mode 100644 index c03e55e..0000000 --- a/sound/soc/at32/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT32 Platform Support -snd-soc-at32-objs := at32-pcm.o -snd-soc-at32-ssc-objs := at32-ssc.o - -obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o -obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o - -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c deleted file mode 100644 index 435f1da..0000000 --- a/sound/soc/at32/at32-pcm.c +++ /dev/null @@ -1,491 +0,0 @@ -/* sound/soc/at32/at32-pcm.c - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-pcm.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "at32-pcm.h" - - - -/*--------------------------------------------------------------------------*\ - * Hardware definition -\*--------------------------------------------------------------------------*/ -/* TODO: These values were taken from the AT91 platform driver, check - * them against real values for AT32 - */ -static const struct snd_pcm_hardware at32_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - - .formats = SNDRV_PCM_FMTBIT_S16, - .period_bytes_min = 32, - .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - - - -/*--------------------------------------------------------------------------*\ - * Data types -\*--------------------------------------------------------------------------*/ -struct at32_runtime_data { - struct at32_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of DMA buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - - dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ - - /* Save PDC registers (for power management) */ - u32 pdc_xpr_save; - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - - - -/*--------------------------------------------------------------------------*\ - * Helper functions -\*--------------------------------------------------------------------------*/ -static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *dmabuf = &substream->dma_buffer; - size_t size = at32_pcm_hardware.buffer_bytes_max; - - dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; - dmabuf->dev.dev = pcm->card->dev; - dmabuf->private_data = NULL; - dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, - &dmabuf->addr, GFP_KERNEL); - pr_debug("at32_pcm: preallocate_dma_buffer: " - "area=%p, addr=%p, size=%ld\n", - (void *)dmabuf->area, (void *)dmabuf->addr, size); - - if (!dmabuf->area) - return -ENOMEM; - - dmabuf->bytes = size; - return 0; -} - - - -/*--------------------------------------------------------------------------*\ - * ISR -\*--------------------------------------------------------------------------*/ -static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - static int count; - - count++; - if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "underrun" : "overrun", params->name, ssc_sr, count); - - /* re-start the PDC */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - } - - - if (ssc_sr & params->mask->ssc_endx) { - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - - snd_pcm_period_elapsed(substream); -} - - - -/*--------------------------------------------------------------------------*\ - * PCM operations -\*--------------------------------------------------------------------------*/ -static int at32_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params - */ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at32_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - pr_debug("hw_params: DMA for %s initialized " - "(dma_bytes=%ld, period_size=%ld)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - - return 0; -} - - - -static int at32_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - - - -static int at32_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - ssc_writex(params->ssc->regs, SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - return 0; -} - - -static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - int ret = 0; - - pr_debug("at32_pcm_trigger: buffer_size = %ld, " - "dma_area = %p, dma_bytes = %ld\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - pr_debug("trigger: period_ptr=%lx, xpr=%x, " - "xcr=%d, xnpr=%x, xncr=%d\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); - - ssc_writex(params->ssc->regs, SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_enable); - - pr_debug("sr=%x, imr=%x\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); - break; /* SNDRV_PCM_TRIGGER_START */ - - - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - break; - - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - - - -static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - - return x; -} - - - -static int at32_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - -out: - return ret; -} - - - -static int at32_pcm_close(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - - -static int at32_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - - - -static struct snd_pcm_ops at32_pcm_ops = { - .open = at32_pcm_open, - .close = at32_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at32_pcm_hw_params, - .hw_free = at32_pcm_hw_free, - .prepare = at32_pcm_prepare, - .trigger = at32_pcm_trigger, - .pointer = at32_pcm_pointer, - .mmap = at32_pcm_mmap, -}; - - - -/*--------------------------------------------------------------------------*\ - * ASoC platform driver -\*--------------------------------------------------------------------------*/ -static u64 at32_pcm_dmamask = 0xffffffff; - -static int at32_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at32_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - - -out: - return ret; -} - - - -static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream == NULL) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - - - -#ifdef CONFIG_PM -static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Disable the PDC and save the PDC registers */ - ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); - - return 0; -} - - - -static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - - ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else /* CONFIG_PM */ -# define at32_pcm_suspend NULL -# define at32_pcm_resume NULL -#endif /* CONFIG_PM */ - - - -struct snd_soc_platform at32_soc_platform = { - .name = "at32-audio", - .pcm_ops = &at32_pcm_ops, - .pcm_new = at32_pcm_new, - .pcm_free = at32_pcm_free_dma_buffers, - .suspend = at32_pcm_suspend, - .resume = at32_pcm_resume, -}; -EXPORT_SYMBOL_GPL(at32_soc_platform); - - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("Atmel AT32 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h deleted file mode 100644 index 2a52430..0000000 --- a/sound/soc/at32/at32-pcm.h +++ /dev/null @@ -1,79 +0,0 @@ -/* sound/soc/at32/at32-pcm.h - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_PCM_H -#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ - -#include <linux/atmel-ssc.h> - - -/* - * Registers and status bits that are required by the PCM driver - * TODO: Is ptcr really used? - */ -struct at32_pdc_regs { - u32 xpr; /* PDC RX/TX pointer */ - u32 xcr; /* PDC RX/TX counter */ - u32 xnpr; /* PDC next RX/TX pointer */ - u32 xncr; /* PDC next RX/TX counter */ - u32 ptcr; /* PDC transfer control */ -}; - - - -/* - * SSC mask info - */ -struct at32_ssc_mask { - u32 ssc_enable; /* SSC RX/TX enable */ - u32 ssc_disable; /* SSC RX/TX disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ - u32 pdc_enable; /* PDC RX/TX enable */ - u32 pdc_disable; /* PDC RX/TX disable */ -}; - - - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at32_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - struct ssc_device *ssc; /* SSC device for stream */ - struct at32_pdc_regs *pdc; /* PDC register info */ - struct at32_ssc_mask *mask; /* SSC mask info */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler) (u32, struct snd_pcm_substream *); -}; - - - -/* - * The AT32 ASoC platform driver - */ -extern struct snd_soc_platform at32_soc_platform; - - - -/* - * SSC register access (since ssc_writel() / ssc_readl() require literal name) - */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) - -#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c deleted file mode 100644 index 4ef6492..0000000 --- a/sound/soc/at32/at32-ssc.c +++ /dev/null @@ -1,849 +0,0 @@ -/* sound/soc/at32/at32-ssc.c - * ASoC platform driver for AT32 using SSC as DAI - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-ssc.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -/* #define DEBUG */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "at32-pcm.h" -#include "at32-ssc.h" - - - -/*-------------------------------------------------------------------------*\ - * Constants -\*-------------------------------------------------------------------------*/ -#define NUM_SSC_DEVICES 3 - -/* - * SSC direction masks - */ -#define SSC_DIR_MASK_UNUSED 0 -#define SSC_DIR_MASK_PLAYBACK 1 -#define SSC_DIR_MASK_CAPTURE 2 - -/* - * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These - * are expected to be used with SSC_BF - */ -/* START bit field values */ -#define SSC_START_CONTINUOUS 0 -#define SSC_START_TX_RX 1 -#define SSC_START_LOW_RF 2 -#define SSC_START_HIGH_RF 3 -#define SSC_START_FALLING_RF 4 -#define SSC_START_RISING_RF 5 -#define SSC_START_LEVEL_RF 6 -#define SSC_START_EDGE_RF 7 -#define SSS_START_COMPARE_0 8 - -/* CKI bit field values */ -#define SSC_CKI_FALLING 0 -#define SSC_CKI_RISING 1 - -/* CKO bit field values */ -#define SSC_CKO_NONE 0 -#define SSC_CKO_CONTINUOUS 1 -#define SSC_CKO_TRANSFER 2 - -/* CKS bit field values */ -#define SSC_CKS_DIV 0 -#define SSC_CKS_CLOCK 1 -#define SSC_CKS_PIN 2 - -/* FSEDGE bit field values */ -#define SSC_FSEDGE_POSITIVE 0 -#define SSC_FSEDGE_NEGATIVE 1 - -/* FSOS bit field values */ -#define SSC_FSOS_NONE 0 -#define SSC_FSOS_NEGATIVE 1 -#define SSC_FSOS_POSITIVE 2 -#define SSC_FSOS_LOW 3 -#define SSC_FSOS_HIGH 4 -#define SSC_FSOS_TOGGLE 5 - -#define START_DELAY 1 - - - -/*-------------------------------------------------------------------------*\ - * Module data -\*-------------------------------------------------------------------------*/ -/* - * SSC PDC registered required by the PCM DMA engine - */ -static struct at32_pdc_regs pdc_tx_reg = { - .xpr = SSC_PDC_TPR, - .xcr = SSC_PDC_TCR, - .xnpr = SSC_PDC_TNPR, - .xncr = SSC_PDC_TNCR, -}; - - - -static struct at32_pdc_regs pdc_rx_reg = { - .xpr = SSC_PDC_RPR, - .xcr = SSC_PDC_RCR, - .xnpr = SSC_PDC_RNPR, - .xncr = SSC_PDC_RNCR, -}; - - - -/* - * SSC and PDC status bits for transmit and receive - */ -static struct at32_ssc_mask ssc_tx_mask = { - .ssc_enable = SSC_BIT(CR_TXEN), - .ssc_disable = SSC_BIT(CR_TXDIS), - .ssc_endx = SSC_BIT(SR_ENDTX), - .ssc_endbuf = SSC_BIT(SR_TXBUFE), - .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), -}; - - - -static struct at32_ssc_mask ssc_rx_mask = { - .ssc_enable = SSC_BIT(CR_RXEN), - .ssc_disable = SSC_BIT(CR_RXDIS), - .ssc_endx = SSC_BIT(SR_ENDRX), - .ssc_endbuf = SSC_BIT(SR_RXBUFF), - .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), -}; - - - -/* - * DMA parameters for each SSC - */ -static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - { - { - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, -}; - - - -static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, -}; - - - - -/*-------------------------------------------------------------------------*\ - * ISR -\*-------------------------------------------------------------------------*/ -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt - * handler in the PCM driver. - */ -static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) -{ - struct at32_ssc_info *ssc_p = dev_id; - struct at32_pcm_dma_params *dma_params; - u32 ssc_sr; - u32 ssc_substream_mask; - int i; - - ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & - ssc_readl(ssc_p->ssc->regs, IMR)); - - /* - * Loop through substreams attached to this SSC. If a DMA-related - * interrupt occured on that substream, call the DMA interrupt - * handler function, if one has been registered in the dma_param - * structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if ((dma_params != NULL) && - (dma_params->dma_intr_handler != NULL)) { - ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); - if (ssc_sr & ssc_substream_mask) { - dma_params->dma_intr_handler(ssc_sr, - dma_params-> - substream); - } - } - } - - - return IRQ_HANDLED; -} - -/*-------------------------------------------------------------------------*\ - * DAI functions -\*-------------------------------------------------------------------------*/ -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at32_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - - - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at32_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - int dir_mask; - - dma_params = ssc_p->dma_params[substream->stream]; - - if (dma_params != NULL) { - ssc_writel(dma_params->ssc->regs, CR, - dma_params->mask->ssc_disable); - pr_debug("%s disabled SSC_SR=0x%08x\n", - (substream->stream ? "receiver" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - - dma_params->ssc = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[substream->stream] = NULL; - } - - - dir_mask = 1 << substream->stream; - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock */ - pr_debug("at32-ssc: Stopping user %d clock\n", - ssc_p->ssc->user); - clk_disable(ssc_p->ssc->clk); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc->irq, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - /* clear the SSC dividers */ - ssc_p->cmr_div = 0; - ssc_p->tcmr_period = 0; - ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - - - -/* - * Set the SSC system clock rate - */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* TODO: What the heck do I do here? */ - return 0; -} - - - -/* - * Record DAI format for use by hw_params() - */ -static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - - - -/* - * Record SSC clock dividers for use in hw_params() - */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT32_SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT32_SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT32_SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - - - -/* - * Configure the SSC - */ -static int at32_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at32_ssc_info *ssc_p = &ssc_info[id]; - struct at32_pcm_dma_params *dma_params; - int channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - - /* - * Currently, there is only one set of dma_params for each direction. - * If more are added, this code will have to be changed to select - * the proper set - */ - dma_params = &ssc_dma_params[id][substream->stream]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[substream->stream] = dma_params; - - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the PCM driver's hw_params() - * function. It should not be used for other purposes as it - * is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - - /* - * Determine sample size in bits and the PDC increment - */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - - case SNDRV_PCM_FORMAT_S16: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - - case SNDRV_PCM_FORMAT_S24: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - - case SNDRV_PCM_FORMAT_S32: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - - default: - pr_warning("at32-ssc: Unsupported PCM format %d", - params_format(params)); - return -EINVAL; - } - pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", - bits, dma_params->pdc_xfer_size, channels); - - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) - if (bits > 16) { - pr_warning("at32-ssc: " - "sample size %d is too large for I2S\n", - bits); - return -EINVAL; - } - - - /* - * Compute the SSC register settings - */ - switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRS clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, SSC_START_FALLING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(RFMR_FSLEN, bits - 1) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, SSC_START_FALLING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(TFMR_FSLEN, bits - 1) | - SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | - SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clock. - * - * The SSC transmit clock is obtained from the BCLK signal - * on the TK line, and the SSC receive clock is generated from - * the transmit clock. - * - * For single channel data, one sample is transferred on the - * falling edge of the LRC clock. For two channel data, one - * sample is transferred on both edges of the LRC clock. - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); - start_event = ((channels == 1) ? - SSC_START_FALLING_RF : SSC_START_EDGE_RF); - - rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, start_event) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, start_event) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_NONE) | - SSC_BF(TCMR_CKS, SSC_CKS_PIN)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, 1) | - SSC_BF(RCMR_START, SSC_START_RISING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, 1) | - SSC_BF(TCMR_START, SSC_START_RISING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_RISING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(TFMR_DATNB, channels - 1) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - pr_warning("at32-ssc: unsupported DAI format 0x%x\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", - rcmr, rfmr, tcmr, tfmr); - - - if (!ssc_p->initialized) { - /* enable peripheral clock */ - pr_debug("at32-ssc: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); - - ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, - ssc_p->name, ssc_p); - if (ret < 0) { - pr_warning("at32-ssc: request irq failed (%d)\n", ret); - pr_debug("at32-ssc: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - return ret; - } - - ssc_p->initialized = 1; - } - - /* Set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); - - /* set transmit clock mode and format */ - ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); - ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); - - pr_debug("at32-ssc: SSC initialized\n"); - return 0; -} - - - -static int at32_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - - dma_params = ssc_p->dma_params[substream->stream]; - - ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); - - return 0; -} - - - -#ifdef CONFIG_PM -static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive */ - ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); - - /* Save the current interrupt mask, then disable unmasked interrupts */ - ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); - ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); - ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); - ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); - ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); - ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); - - return 0; -} - - - -static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - u32 cr; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* restore SSC register settings */ - ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); - ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); - ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); - - /* re-enable interrupts */ - ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - - /* Re-enable recieve and transmit as appropriate */ - cr = 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; - ssc_writel(ssc_p->ssc->regs, CR, cr); - - return 0; -} -#else /* CONFIG_PM */ -# define at32_ssc_suspend NULL -# define at32_ssc_resume NULL -#endif /* CONFIG_PM */ - - -#define AT32_SSC_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - - -#define AT32_SSC_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) - - -struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { - { - .name = "at32-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[0], - }, - { - .name = "at32-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[1], - }, - { - .name = "at32-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[2], - }, -}; -EXPORT_SYMBOL_GPL(at32_ssc_dai); - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h deleted file mode 100644 index 3c052db..0000000 --- a/sound/soc/at32/at32-ssc.h +++ /dev/null @@ -1,59 +0,0 @@ -/* sound/soc/at32/at32-ssc.h - * ASoC SSC interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_SSC_H -#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ - -#include <linux/types.h> -#include <linux/atmel-ssc.h> - -#include "at32-pcm.h" - - - -struct at32_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - - - -struct at32_ssc_info { - char *name; - struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* true if SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at32_pcm_dma_params *dma_params[2]; - struct at32_ssc_state ssc_state; -}; - - -/* SSC divider ids */ -#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - - -extern struct snd_soc_dai at32_ssc_dai[]; - - - -#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c deleted file mode 100644 index 3f32621..0000000 --- a/sound/soc/at32/playpaq_wm8510.c +++ /dev/null @@ -1,522 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum(a)acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/version.h> -#include <linux/kernel.h> -#include <linux/errno.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/at32ap700x.h> -#include <mach/portmux.h> - -#include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct at32_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_1; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_3; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const char *intercon[][3] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_codec *codec) -{ - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } - - - /* always connected pins */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync(codec); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai = &at32_ssc_dai[0], - .codec_dai = &wm8510_dai, - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_machine snd_soc_machine_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - - - -static struct wm8510_setup_data playpaq_wm8510_setup = { - .i2c_address = 0x1a, -}; - - - -static struct snd_soc_device playpaq_wm8510_snd_devdata = { - .machine = &snd_soc_machine_playpaq, - .platform = &at32_soc_platform, - .codec_dev = &soc_codec_dev_wm8510, - .codec_data = &playpaq_wm8510_setup, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc = NULL; - - - /* - * Request SSC device - */ - ssc = ssc_request(0); - if (IS_ERR(ssc)) { - ret = PTR_ERR(ssc); - ssc = NULL; - goto err_ssc; - } - ssc_p->ssc = ssc; - - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - goto err_pll0; - } - if (clk_set_parent(_gclk0, _pll0)) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); - playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } -err_gclk0: - if (ssc != NULL) { - ssc_free(ssc); - ssc = NULL; - } -err_ssc: - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; - struct ssc_device *ssc; - - if (ssc_p != NULL) { - ssc = ssc_p->ssc; - if (ssc != NULL) - ssc_free(ssc); - ssc_p->ssc = NULL; - } - - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig deleted file mode 100644 index 9051865..0000000 --- a/sound/soc/at91/Kconfig +++ /dev/null @@ -1,27 +0,0 @@ -config SND_AT91_SOC - tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 - help - Say Y or M if you want to add support for codecs attached to - the AT91 SSC interface. You will also need - to select the audio interfaces to support below. - -config SND_AT91_SOC_SSC - tristate - -config SND_AT91_SOC_ETI_B1_WM8731 - tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on WM8731-based - Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. - -config SND_AT91_SOC_ETI_SLAVE - bool "Run codec in slave Mode on Endrelia boards" - depends on SND_AT91_SOC_ETI_B1_WM8731 - default n - help - Say Y if you want to run with the AT91 SSC generating the BCLK - and LRC signals on Endrelia boards. diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile deleted file mode 100644 index f23da17..0000000 --- a/sound/soc/at91/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT91 Platform Support -snd-soc-at91-objs := at91-pcm.o -snd-soc-at91-ssc-objs := at91-ssc.o - -obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o - -# AT91 Machine Support -snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o - -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c deleted file mode 100644 index 7ab48bd..0000000 --- a/sound/soc/at91/at91-pcm.c +++ /dev/null @@ -1,434 +0,0 @@ -/* - * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "at91-pcm: " x) -#else -#define DBG(x...) -#endif - -static const struct snd_pcm_hardware at91_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - -struct at91_runtime_data { - struct at91_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - u32 pdc_xpr_save; /* PDC register save */ - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - -static void at91_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - static int count = 0; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - - printk(KERN_WARNING - "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, - prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - -static int at91_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at91_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - return 0; -} - -static int at91_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int at91_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - at91_ssc_write(params->ssc_base + AT91_SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; -} - -static int at91_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", - (unsigned long) prtd->period_ptr, - at91_ssc_read(params->ssc_base + params->pdc->xpr), - at91_ssc_read(params->ssc_base + params->pdc->xcr), - at91_ssc_read(params->ssc_base + params->pdc->xnpr), - at91_ssc_read(params->ssc_base + params->pdc->xncr)); - - at91_ssc_write(params->ssc_base + AT91_SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, - params->mask->pdc_enable); - - DBG("sr=%lx imr=%lx\n", - at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t at91_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - return x; -} - -static int at91_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int at91_pcm_close(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int at91_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -struct snd_pcm_ops at91_pcm_ops = { - .open = at91_pcm_open, - .close = at91_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at91_pcm_hw_params, - .hw_free = at91_pcm_hw_free, - .prepare = at91_pcm_prepare, - .trigger = at91_pcm_trigger, - .pointer = at91_pcm_pointer, - .mmap = at91_pcm_mmap, -}; - -static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = at91_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static u64 at91_pcm_dmamask = 0xffffffff; - -static int at91_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at91_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -#ifdef CONFIG_PM -static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr); - prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr); - prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr); - prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr); - - return 0; -} - -static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save); - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define at91_pcm_suspend NULL -#define at91_pcm_resume NULL -#endif - -struct snd_soc_platform at91_soc_platform = { - .name = "at91-audio", - .pcm_ops = &at91_pcm_ops, - .pcm_new = at91_pcm_new, - .pcm_free = at91_pcm_free_dma_buffers, - .suspend = at91_pcm_suspend, - .resume = at91_pcm_resume, -}; - -EXPORT_SYMBOL_GPL(at91_soc_platform); - -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("Atmel AT91 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h deleted file mode 100644 index e5aada2..0000000 --- a/sound/soc/at91/at91-pcm.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.h by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_PCM_H -#define _AT91_PCM_H - -#include <mach/hardware.h> - -struct at91_ssc_periph { - void __iomem *base; - u32 pid; -}; - -/* - * Registers and status bits that are required by the PCM driver. - */ -struct at91_pdc_regs { - unsigned int xpr; /* PDC recv/trans pointer */ - unsigned int xcr; /* PDC recv/trans counter */ - unsigned int xnpr; /* PDC next recv/trans pointer */ - unsigned int xncr; /* PDC next recv/trans counter */ - unsigned int ptcr; /* PDC transfer control */ -}; - -struct at91_ssc_mask { - u32 ssc_enable; /* SSC recv/trans enable */ - u32 ssc_disable; /* SSC recv/trans disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ - u32 pdc_enable; /* PDC recv/trans enable */ - u32 pdc_disable; /* PDC recv/trans disable */ -}; - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at91_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - void __iomem *ssc_base; /* SSC base address */ - struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */ - struct at91_ssc_mask *mask;/* SSC & PDC status bits */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler)(u32, struct snd_pcm_substream *); -}; - -extern struct snd_soc_platform at91_soc_platform; - -#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -#define at91_ssc_write(a,v) __raw_writel((v),(a)) - -#endif /* _AT91_PCM_H */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c deleted file mode 100644 index 5d44515..0000000 --- a/sound/soc/at91/at91-ssc.c +++ /dev/null @@ -1,791 +0,0 @@ -/* - * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * - * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_pmc.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x) -#else -#define DBG(x...) -#endif - -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) -#define NUM_SSC_DEVICES 1 -#else -#define NUM_SSC_DEVICES 3 -#endif - - -/* - * SSC PDC registers required by the PCM DMA engine. - */ -static struct at91_pdc_regs pdc_tx_reg = { - .xpr = ATMEL_PDC_TPR, - .xcr = ATMEL_PDC_TCR, - .xnpr = ATMEL_PDC_TNPR, - .xncr = ATMEL_PDC_TNCR, -}; - -static struct at91_pdc_regs pdc_rx_reg = { - .xpr = ATMEL_PDC_RPR, - .xcr = ATMEL_PDC_RCR, - .xnpr = ATMEL_PDC_RNPR, - .xncr = ATMEL_PDC_RNCR, -}; - -/* - * SSC & PDC status bits for transmit and receive. - */ -static struct at91_ssc_mask ssc_tx_mask = { - .ssc_enable = AT91_SSC_TXEN, - .ssc_disable = AT91_SSC_TXDIS, - .ssc_endx = AT91_SSC_ENDTX, - .ssc_endbuf = AT91_SSC_TXBUFE, - .pdc_enable = ATMEL_PDC_TXTEN, - .pdc_disable = ATMEL_PDC_TXTDIS, -}; - -static struct at91_ssc_mask ssc_rx_mask = { - .ssc_enable = AT91_SSC_RXEN, - .ssc_disable = AT91_SSC_RXDIS, - .ssc_endx = AT91_SSC_ENDRX, - .ssc_endbuf = AT91_SSC_RXBUFF, - .pdc_enable = ATMEL_PDC_RXTEN, - .pdc_disable = ATMEL_PDC_RXTDIS, -}; - - -/* - * DMA parameters. - */ -static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - {{ - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#if NUM_SSC_DEVICES == 3 - {{ - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, - {{ - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#endif -}; - -struct at91_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - -static struct at91_ssc_info { - char *name; - struct at91_ssc_periph ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* 1=SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at91_pcm_dma_params *dma_params[2]; - struct at91_ssc_state ssc_state; - -} ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = 0, - .initialized = 0, - }, -#if NUM_SSC_DEVICES == 3 - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = 0, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = 0, - .initialized = 0, - }, -#endif -}; - -static unsigned int at91_ssc_sysclk; - -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA - * interrupt handler in the PCM driver. - */ -static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id) -{ - struct at91_ssc_info *ssc_p = dev_id; - struct at91_pcm_dma_params *dma_params; - u32 ssc_sr; - int i; - - ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR) - & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - - /* - * Loop through the substreams attached to this SSC. If - * a DMA-related interrupt occurred on that substream, call - * the DMA interrupt handler function, if one has been - * registered in the dma_params structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if (dma_params != NULL && dma_params->dma_intr_handler != NULL && - (ssc_sr & - (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) - - dma_params->dma_intr_handler(ssc_sr, dma_params->substream); - } - - return IRQ_HANDLED; -} - -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at91_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - DBG("ssc_startup: SSC_SR=0x%08lx\n", - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at91_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir, dir_mask; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - if (dma_params != NULL) { - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_disable); - DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - - dma_params->ssc_base = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[dir] = NULL; - } - - dir_mask = 1 << dir; - - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc.pid, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - -/* - * Record the SSC system clock rate. - */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* - * The only clock supplied to the SSC is the AT91 master clock, - * which is only used if the SSC is generating BCLK and/or - * LRC clocks. - */ - switch (clk_id) { - case AT91_SYSCLK_MCK: - at91_ssc_sysclk = freq; - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * Record the DAI format for use in hw_params(). - */ -static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - -/* - * Record SSC clock dividers for use in hw_params(). - */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT91SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value. - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT91SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT91SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -/* - * Configure the SSC. - */ -static int at91_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at91_ssc_info *ssc_p = &ssc_info[id]; - struct at91_pcm_dma_params *dma_params; - int dir, channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - /* - * Currently, there is only one set of dma params for - * each direction. If more are added, this code will - * have to be changed to select the proper set. - */ - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc_base = ssc_p->ssc.base; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - /* - * Determine sample size in bits and the PDC increment. - */ - switch(params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format"); - return -EINVAL; - } - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "at91-ssc: sample size %d is too large for I2S\n", bits); - return -EINVAL; - } - - /* - * Compute SSC register settings. - */ - switch (ssc_p->daifmt - & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is generated from the - * transmit clock. - * - * For single channel data, one sample is transferred on the falling - * edge of the LRC clock. For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = channels == 1 - ? AT91_SSC_START_FALLING_RF - : AT91_SSC_START_EDGE_RF; - - rcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - - - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); - - if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - DBG("Starting pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); - - /* Reset the SSC and its PDC registers */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0); - - if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt, - 0, ssc_p->name, ssc_p)) < 0) { - printk(KERN_WARNING "at91-ssc: request_irq failure\n"); - - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - return ret; - } - - ssc_p->initialized = 1; - } - - /* set SSC clock mode register */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr); - - /* set transmit clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr); - - DBG("hw_params: SSC initialized\n"); - return 0; -} - - -static int at91_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_enable); - - DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", - at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR)); - return 0; -} - - -#ifdef CONFIG_PM -static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive. */ - ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - AT91_SSC_TXDIS | AT91_SSC_RXDIS); - - /* Save the current interrupt mask, then disable unmasked interrupts. */ - ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR); - ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR); - ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR); - ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR); - ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR); - - return 0; -} - -static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); - - return 0; -} - -#else -#define at91_ssc_suspend NULL -#define at91_ssc_resume NULL -#endif - -#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - -#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { - { .name = "at91-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[0].ssc, - }, -#if NUM_SSC_DEVICES == 3 - { .name = "at91-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[1].ssc, - }, - { .name = "at91-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[2].ssc, - }, -#endif -}; - -EXPORT_SYMBOL_GPL(at91_ssc_dai); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino(a)endrelia.com, www.endrelia.com"); -MODULE_DESCRIPTION("AT91 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h deleted file mode 100644 index 6b7bf38..0000000 --- a/sound/soc/at91/at91-ssc.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Jan 9, 2007 - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_SSC_H -#define _AT91_SSC_H - -/* SSC system clock ids */ -#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ - -/* SSC divider ids */ -#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - -extern struct snd_soc_dai at91_ssc_dai[]; - -#endif /* _AT91_SSC_H */ - diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c deleted file mode 100644 index b81d6b2..0000000 --- a/sound/soc/at91/eti_b1_wm8731.c +++ /dev/null @@ -1,348 +0,0 @@ -/* - * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. - * - * Author: Frank Mandarino <fmandarino(a)endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 29, 2006 - * - * Based on corgi.c by: - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - * Richard Purdie <richard(a)openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/hardware.h> -#include <mach/gpio.h> - -#include "../codecs/wm8731.h" -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) -#else -#define DBG(x...) -#endif - -static struct clk *pck1_clk; -static struct clk *pllb_clk; - - -static int eti_b1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* cpu clock is the AT91 master clock sent to the SSC */ - ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, - 60000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Start PCK1 clock. */ - clk_enable(pck1_clk); - DBG("pck1 started\n"); - - return 0; -} - -static void eti_b1_shutdown(struct snd_pcm_substream *substream) -{ - /* Stop PCK1 clock. */ - clk_disable(pck1_clk); - DBG("pck1 stopped\n"); -} - -static int eti_b1_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - unsigned int rate; - int cmr_div, period; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ - break; - case 32000: - cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ - period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ - break; - case 48000: - cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ - period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_RCMR_PERIOD, period); - } - if (ret < 0) - return ret; - -#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - /* - * Codec in Master Mode. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - -#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - - return 0; -} - -static struct snd_soc_ops eti_b1_ops = { - .startup = eti_b1_startup, - .hw_params = eti_b1_hw_params, - .shutdown = eti_b1_shutdown, -}; - - -static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route intercon[] = { - - /* speaker connected to LHPOUT */ - {"Ext Spk", NULL, "LHPOUT"}, - - /* mic is connected to Mic Jack, with WM8731 Mic Bias */ - {"MICIN", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Int Mic"}, -}; - -/* - * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. - */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -{ - DBG("eti_b1_wm8731_init() called\n"); - - /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, - ARRAY_SIZE(eti_b1_dapm_widgets)); - - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); - - /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - - /* always connected */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link eti_b1_dai = { - .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai = &at91_ssc_dai[1], - .codec_dai = &wm8731_dai, - .init = eti_b1_wm8731_init, - .ops = &eti_b1_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1 = { - .name = "ETI_B1_WM8731", - .dai_link = &eti_b1_dai, - .num_links = 1, -}; - -static struct wm8731_setup_data eti_b1_wm8731_setup = { - .i2c_address = 0x1a, -}; - -static struct snd_soc_device eti_b1_snd_devdata = { - .machine = &snd_soc_machine_eti_b1, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &eti_b1_wm8731_setup, -}; - -static struct platform_device *eti_b1_snd_device; - -static int __init eti_b1_init(void) -{ - int ret; - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) { - DBG("SSC1 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K); - if (!ssc->base) { - DBG("SSC1 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC1; - - eti_b1_snd_device = platform_device_alloc("soc-audio", -1); - if (!eti_b1_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); - eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; - - ret = platform_device_add(eti_b1_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_snd_device); - goto fail_io_unmap; - } - - at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ - at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ - at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ - at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ -/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ - at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ - - /* - * Set PCK1 parent to PLLB and its rate to 12 Mhz. - */ - pllb_clk = clk_get(NULL, "pllb"); - pck1_clk = clk_get(NULL, "pck1"); - - clk_set_parent(pck1_clk, pllb_clk); - clk_set_rate(pck1_clk, 12000000); - - DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); - - /* assign the GPIO pin to PCK1 */ - at91_set_B_periph(AT91_PIN_PA24, 0); - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); -#else - printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); -#endif - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); - return ret; -} - -static void __exit eti_b1_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - clk_put(pck1_clk); - clk_put(pllb_clk); - - platform_device_unregister(eti_b1_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); -} - -module_init(eti_b1_init); -module_exit(eti_b1_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -MODULE_LICENSE("GPL");

1 0

[alsa-devel] alsa-plugins is missing from the listing on http://git.alsa-project.org/
by Lennart Poettering 25 Sep '08

25 Sep '08

Heya! When I open http://git.alsa-project.org/ in my web browser I don't see the alsa-plugins repo listed there although it actually is existant. There's something wrong with the gitweb installation... Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] alsa-plugins: Fix destruction logic on failed construction
by Lennart Poettering 25 Sep '08

25 Sep '08

Heya! We need to stop the main loop first, then free the context and finally free the mainloop. --- pulse/pulse.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index 022ca55..3940238 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -208,6 +208,12 @@ snd_pulse_t *pulse_new(void) fail: if (p->mainloop) + pa_threaded_mainloop_stop(p->mainloop); + + if (p->context) + pa_context_unref(p->context); + + if (p->mainloop) pa_threaded_mainloop_free(p->mainloop); if (p->main_fd >= 0) -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] alsa-plugins: Fix destruction order on failure
by Lennart Poettering 25 Sep '08

25 Sep '08

Heya! If we failure we should make sure the event loop thread is not running and hence not accessing our source/sink strings before we free those strings. --- pulse/ctl_pulse.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 0b10628..8a5546c 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -728,14 +728,11 @@ SND_CTL_PLUGIN_DEFINE_FUNC(pulse) return 0; error: - if (ctl->source) - free(ctl->source); - if (ctl->sink) - free(ctl->sink); - if (ctl->p) pulse_free(ctl->p); + free(ctl->source); + free(ctl->sink); free(ctl); return err; -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] alsa-plugins: fix minor memory leak
by Lennart Poettering 25 Sep '08

25 Sep '08

Heya! If we fail, we need to free the device string --- pulse/pcm_pulse.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 9a50aff..a5f794f 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -917,6 +917,7 @@ error: if (pcm->p) pulse_free(pcm->p); + free(pcm->device); free(pcm); return err; -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH] hda: slave_dig_outs code block in wrong location
by Matthew Ranostay 25 Sep '08

25 Sep '08

Removed invalid references to slave_dig_outs inside the S/PDIF IN capture switch control. Beforehand this was basically a mute switch for the S/PDIF outs as well. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index c742e10..cb9aae5 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -1652,15 +1652,9 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, mutex_lock(&codec->spdif_mutex); change = codec->spdif_in_enable != val; if (change) { - hda_nid_t *d; codec->spdif_in_enable = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); - - if (codec->slave_dig_outs) - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change;

2 1

[alsa-devel] How to find out which PCM handle ASCII identifier to pass to snd_pcm_open?
by Andre Kirchner 25 Sep '08

25 Sep '08

I have just started learning ALSA's api, and came accross some examples at http://equalarea.com/paul/alsa-audio.html, which pass argv[1] to snd_pcm_open. My question is which should be the values of argv[1] for the microphone, and left and right speakers devices? I'm not sure this is the right mailing list for this question, and I apologize if not :-) Thanks, Andre

2 1

[alsa-devel] OOPS on USB sound card disconnect while in use
by Lasse Kärkkäinen 25 Sep '08

25 Sep '08

This has been broken for a long time. Symtoms: programs using the card while it is disconnected get stuck and cannot be killed, and OOPS is printed on the log: [108034.041573] usb 9-2: USB disconnect, address 3 [108043.576040] usb 9-2: new full speed USB device using uhci_hcd and address 4 [108043.757185] usb 9-2: configuration #1 chosen from 1 choice [108057.602616] BUG: unable to handle kernel paging request at 000061d855001080 [108057.602621] IP: [<ffffffffa00e0c71>] hcd_buffer_free+0x11/0x90 [usbcore] [108057.602643] PGD 0 [108057.602647] Oops: 0000 [1] SMP [108057.602652] CPU 0 [108057.602655] Modules linked in: nls_cp437 cifs af_packet binfmt_misc rfcomm l2cap bluetooth ppdev ipv6 cpufreq_conservative cpufreq_ondemand cpufreq_powersave cpufreq_userspace cpufreq_stats freq_table container video output sbs sbshc pci_slot wmi battery iptable_filter ip_tables x_tables ext2 ac lp loop snd_seq_dummy arc4 snd_hda_intel snd_seq_oss ecb crypto_blkcipher snd_usb_audio snd_usb_lib snd_pcm_oss snd_mixer_oss snd_seq_midi snd_pcm snd_rawmidi ath9k snd_seq_midi_event snd_hwdep snd_seq mac80211 psmouse snd_timer snd_seq_device cfg80211 snd serio_raw iTCO_wdt iTCO_vendor_support soundcore nvidia(P) snd_page_alloc i2c_core parport_pc parport button shpchp pci_hotplug intel_agp evdev joydev ext3 jbd mbcache sg sd_mod crc_t10dif ata_generic ata_piix pata_jmicron usbhid hid pata_acpi ahci libata scsi_mod dock sky2 uhci_hcd ehci_hcd usbcore raid10 raid456 async_xor async_memcpy async_tx xor raid1 raid0 multipath linear md_mod thermal processor fan fbcon tileblit font bitblit softcursor uvesafb cn fuse [108057.602714] Pid: 15655, comm: rhythmbox Tainted: P 2.6.27-3-generic #1 [108057.602716] RIP: 0010:[<ffffffffa00e0c71>] [<ffffffffa00e0c71>] hcd_buffer_free+0x11/0x90 [usbcore] [108057.602727] RSP: 0018:ffff8801a7d97cc8 EFLAGS: 00010286 [108057.602729] RAX: ffff880219061680 RBX: ffff88021a162e00 RCX: 0000000037836060 [108057.602731] RDX: ffff880037836060 RSI: 0000000000000020 RDI: 000061d855001080 [108057.602733] RBP: ffff8801a7d97cc8 R08: ffff880037836060 R09: 0000000000000566 [108057.602734] R10: 0000000000000002 R11: 0000000000000000 R12: ffff88021a1630c8 [108057.602736] R13: 0000000000001fff R14: ffff88021a44e400 R15: 0000000000000000 [108057.602738] FS: 00007ffd6f42f7a0(0000) GS:ffffffff806ffa80(0000) knlGS:0000000000000000 [108057.602740] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [108057.602742] CR2: 000061d855001080 CR3: 00000001d80ff000 CR4: 00000000000006e0 [108057.602744] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [108057.602745] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [108057.602747] Process rhythmbox (pid: 15655, threadinfo ffff8801a7d96000, task ffff8801c68cd9c0) [108057.602749] Stack: ffff8801a7d97cd8 ffffffffa00d3425 ffff8801a7d97cf8 ffffffffa0b391d5 [108057.602754] ffff88021aa00c00 ffff88021aa00c00 ffff8801a7d97d18 ffffffffa0b39212 [108057.602758] ffff88021a163000 ffff88021a163000 ffff8801a7d97d28 ffffffffa0b39275 [108057.602761] Call Trace: [108057.602775] [<ffffffffa00d3425>] usb_buffer_free+0x25/0x30 [usbcore] [108057.602780] [<ffffffffa0b391d5>] snd_usbmidi_out_endpoint_delete+0x35/0x50 [snd_usb_lib] [108057.602784] [<ffffffffa0b39212>] snd_usbmidi_free+0x22/0x70 [snd_usb_lib] [108057.602787] [<ffffffffa0b39275>] snd_usbmidi_rawmidi_free+0x15/0x20 [snd_usb_lib] [108057.602792] [<ffffffffa0af9287>] snd_rawmidi_free+0xf7/0x110 [snd_rawmidi] [108057.602796] [<ffffffffa0af92b2>] snd_rawmidi_dev_free+0x12/0x20 [snd_rawmidi] [108057.602805] [<ffffffffa0a1a8b1>] snd_device_free+0x71/0xd0 [snd] [108057.602811] [<ffffffffa0a1a974>] snd_device_free_all+0x64/0x70 [snd] [108057.602818] [<ffffffffa0a14a1f>] snd_card_do_free+0x3f/0xe0 [snd] [108057.602824] [<ffffffffa0a14bff>] snd_card_file_remove+0x13f/0x170 [snd] [108057.602833] [<ffffffffa0b09f86>] snd_pcm_release+0xa6/0xc0 [snd_pcm] [108057.602839] [<ffffffffa0a15630>] ? snd_disconnect_release+0x0/0xb0 [snd] [108057.602846] [<ffffffffa0a156a6>] snd_disconnect_release+0x76/0xb0 [snd] [108057.602852] [<ffffffff802ec8a9>] __fput+0xc9/0x1b0 [108057.602854] [<ffffffff802ec9b5>] fput+0x25/0x30 [108057.602858] [<ffffffff802c67e3>] remove_vma+0x43/0x90 [108057.602860] [<ffffffff802c860a>] do_munmap+0x2ea/0x330 [108057.602862] [<ffffffff802c86a5>] sys_munmap+0x55/0x80 [108057.602866] [<ffffffff8021288a>] system_call_fastpath+0x16/0x1b [108057.602868] [108057.602869] [108057.602870] Code: 07 48 ff 84 02 00 02 00 00 4c 89 f1 49 89 4d 20 eb 9f 90 90 90 90 90 90 90 90 55 48 89 e5 0f 1f 44 00 00 48 85 d2 49 89 d0 74 3b <48> 8b 07 48 83 b8 c0 01 00 00 00 74 52 48 c7 c2 20 90 0e a0 31 [108057.602896] RIP [<ffffffffa00e0c71>] hcd_buffer_free+0x11/0x90 [usbcore] [108057.602907] RSP <ffff8801a7d97cc8> [108057.602909] CR2: 000061d855001080 [108057.602917] ---[ end trace 8c6a5ea5cd36c92b ]--- asound.conf: # Allow mixing of multiple output streams to this device pcm.output { type dmix ipc_key 1024 ipc_perm 0660 # Sound for everybody in your group! slave.pcm "hw:UA25" slave { # This stuff provides some fixes for latency issues. # buffer_size should be set for your audio chipset. period_time 0 period_size 1024 buffer_size 8192 rate 48000 } bindings { 0 0 1 1 } } pcm.!default { type plug slave.pcm "output" slave.rate 48000 } uname -a: Linux Kaidenn 2.6.27-3-generic #1 SMP Wed Sep 10 16:18:52 UTC 2008 x86_64 GNU/Linux And yes, I know that it is tainted. Live with it.

2 1

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 24 Sep '08

24 Sep '08

This series fixes a couple of bugs (neither of which affect 2.6.27) and introduces a new API call for marking disconnected pins, mostly as a placeholder for future development so machine drivers can start using it now and then benefit when a more substantial implementation follows. The following changes since commit 3090258fed179996ac4daecb482bad28d84cb054: Jean Delvare (1): ALSA: ASoC: Convert tlv320aic3x to a new-style i2c driver (v2) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Fix inverted input PGA mute bits in WM8903 ASoC: Fix build of GTA01 audio driver ASoC: Allow machine drivers to mark pins as not connected ASoC: Use snd_soc_dapm_nc_pin() in GTA01 audio driver include/sound/soc-dapm.h | 1 + sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/s3c24xx/neo1973_wm8753.c | 17 ++++++++--------- sound/soc/soc-dapm.c | 20 ++++++++++++++++++++ 4 files changed, 31 insertions(+), 11 deletions(-)

2 1

[alsa-devel] [PATCH 1/4] ASoC: Fix inverted input PGA mute bits in WM8903
by 20080924115907.GA12401＠opensource.wolfsonmicro.com 24 Sep '08

24 Sep '08

From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3f54ec..ce40d78 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0, 0, 31, 0), SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1, 6, 1, 0), SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, -- 1.5.6.5

2 4

[alsa-devel] hda_intel (sigmatel) defunct in mmotm 2008-09-13-03-09
by Jiri Slaby 24 Sep '08

24 Sep '08

Hi, I've found my sound defunct in mmotm 2008-09-13-03-09 (opposing to 2008-09-10-19-39). My debug shows: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 azx_codec_create (1232): t=0, max=4, def=3, mask=4, probe_mask=ffffffff snd_hda_codec_new A: ffff88007b6988f8 patch_stac927x A: 0 stac92xx_parse_auto_config A stac92xx_parse_auto_config B stac92xx_parse_auto_config E stac92xx_parse_auto_config F stac92xx_parse_auto_config G stac92xx_parse_auto_config H stac92xx_parse_auto_config L: 0 stac92xx_parse_auto_config M: 0 stac92xx_parse_auto_config O stac92xx_parse_auto_config P stac92xx_parse_auto_config Q stac92xx_auto_create_spdif_mux_ctls: num_cons=5 patch_stac927x B: -22 snd_hda_codec_new E: -22 azx_codec_create (1239): c=2, err=-22 azx_codec_create (1251): cod=0, acod=0 hda-intel: no codecs initialized HDA Intel 0000:00:1b.0: PCI INT A disabled The problem lays in the newly added function: stac92xx_auto_create_spdif_mux_ctls The count (=5) is out of bounds. The device is: 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02) Subsystem: Intel Corporation Optiplex 755 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at ffa70000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel 00: 86 80 3e 29 06 00 10 00 02 00 03 04 08 00 00 00 10: 04 00 a7 ff 00 00 00 00 00 00 00 00 00 00 00 00 20: 00 00 00 00 00 00 00 00 00 00 00 00 86 80 3e 29 30: 00 00 00 00 50 00 00 00 00 00 00 00 0a 01 00 00 codec output from OK kernel: Codec: SigmaTel STAC9274D Address: 2 Vendor Id: 0x83847621 Subsystem Id: 0x100 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0e, stepsize=0x05, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=3, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x49 0x49] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x49 0x49] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x49 0x46] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x49 0x49] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D3, actual=D3 Delay: 13 samples Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1b Processing caps: benign=0, ncoeff=0 Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1c Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1d Processing caps: benign=0, ncoeff=0 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02214020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=30, enabled=1 Connection: 3 0x02* 0x03 0x06 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19080: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x8, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 3 0x02* 0x03 0x06 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0181304e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19040: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x503301f0: [N/A] CD at Int N/A Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x50a001f0: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x50a001f0: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x18 [Audio Selector] wcaps 0x300103: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x15 Node 0x19 [Audio Selector] wcaps 0x300103: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x16 Node 0x1a [Audio Selector] wcaps 0x300103: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x17 Node 0x1b [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Connection: 3 0x18* 0x13 0x14 Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x19* 0x13 0x14 Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x1a* 0x13 0x14 Node 0x1e [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x1f [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Delay: 3 samples Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x22 Node 0x21 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x01442170: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Grey DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 5 0x1e* 0x1f 0x1b 0x1c 0x1d Node 0x22 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect EAPD 0x0: Pin Default 0x81c42090: [Fixed] SPDIF In at Ext Rear Conn = RCA, Color = Grey DefAssociation = 0x9, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x23 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x24 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=127 Connection: 5 0x02* 0x03 0x04 0x05 0x06

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[alsa-devel] Something wrong with snd_pcm_avail_update
by Lennart Poettering 24 Sep '08

24 Sep '08

Heya! There seems to be something wrong with snd_pcm_update_avail() for mmap in recent 2.6.26 kernels: Quoting Alexander Gavrilov: <snip> Fedora 9 with kernel 2.6.26.3-29 exhibits a bug where the driver decrements its hw.ptr under certain stress conditions, or at least it appears so from debugging. It causes alsa-lib to give absurdly huge avail numbers to pulseaudio, which crashes it. This patch adds a work-around to avoid such unfriendly behavior until the bug is fixed. As this is not a real fix, it displays a message every time it is activated. </snip> https://bugzilla.redhat.com/show_bug.cgi?id=462200 (see the patch attached to that bug report) Takashi, Jaroslav, do you know anything about this? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 2

[alsa-devel] [PATCH] hda: use last DAC defined for hp_pin
by Matthew Ranostay 24 Sep '08

24 Sep '08

Patch allows the last DAC in the dac_nids for the hp_nid if there is an available one this isn't in use by a line_out entry or if hp_nid isn't already defined. This solves the issues with the 'Headphone Playback' mixer controls on the 92hd73xxx branch and possibly others. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..01f5475 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -2817,6 +2817,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } + if ((spec->multiout.num_dacs - cfg->line_outs) > 0 && + cfg->hp_outs && !spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + if (cfg->hp_outs > 1) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_HP_SWITCH, @@ -4083,8 +4087,6 @@ again: case STAC_DELL_M6: spec->init = dell_eq_core_init; spec->num_smuxes = 0; - spec->multiout.hp_nid = - spec->multiout.dac_nids[spec->multiout.num_dacs - 1]; spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; spec->num_amps = 1;

2 1

[alsa-devel] Some idea about usb audio gadget driver.
by Bryan Wu 24 Sep '08

24 Sep '08

Hi ALSA and USB folks, I plan to develop a usb audio gadget driver on Blackfin platform (can be shared for other embedded system). - A new usb gadget driver which recognized by Linux/Windows PC as an USB audio card. - There is an ALSA sound card on the same embedded device. - The usb gadget driver receives audio data from host and send them to ALSA sound card to playback - The usb gadget driver send the audio data recorded by ALSA sound card to host as recording. - The usb gadget driver response to some audio card control command such as volume/mute control Questions, - Is there any similar things on this planet? - How about gmidi usb midi gadget driver? - How can I open/playback/record/close a ALSA sound card in kernel space not using alsalib in user space? Thanks a lot -Bryan

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[alsa-devel] [PATCH] ASoC: maintainers - update email address for Liam Girdwood
by Liam Girdwood 23 Sep '08

23 Sep '08

This patch updates the maintainers email address for Liam Girdwood and adds a URL for the ASoC website. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- MAINTAINERS | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 0a613cb..ef10604 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3833,11 +3833,12 @@ S: Maintained SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT P: Liam Girdwood -M: liam.girdwood(a)wolfsonmicro.com +M: lrg(a)slimlogic.co.uk P: Mark Brown M: broonie(a)opensource.wolfsonmicro.com T: git opensource.wolfsonmicro.com/linux-2.6-asoc L: alsa-devel(a)alsa-project.org (subscribers-only) +W: http://alsa-project.org/main/index.php/ASoC S: Supported SPI SUBSYSTEM -- 1.5.4.3

4 4

[alsa-devel] [PATCH] ASoC: Convert tlv320aic3x to a new-style i2c driver (v2)
by Jean Delvare 22 Sep '08

22 Sep '08

Convert the tlv320aic3x codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Vladimir Barinov <vbarinov(a)ru.mvista.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 106 +++++++++++++++++++-------------------- sound/soc/codecs/tlv320aic3x.h | 1 sound/soc/davinci/davinci-evm.c | 1 sound/soc/omap/n810.c | 1 4 files changed, 57 insertions(+), 52 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/codecs/tlv320aic3x.c 2008-09-01 11:06:20.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/tlv320aic3x.c 2008-09-01 11:06:23.000000000 +0200 @@ -1172,71 +1172,39 @@ static struct snd_soc_device *aic3x_socd * AIC3X 2 wire address can be up to 4 devices with device addresses * 0x18, 0x19, 0x1A, 0x1B */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver aic3x_i2c_driver; -static struct i2c_client client_template; /* * If the i2c layer weren't so broken, we could pass this kind of data * around */ -static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int aic3x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = aic3x_socdev; - struct aic3x_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - printk(KERN_ERR "aic3x: failed to attach codec at addr %x\n", - addr); - goto err; - } - ret = aic3x_init(socdev); - if (ret < 0) { + if (ret < 0) printk(KERN_ERR "aic3x: failed to initialise AIC3X\n"); - goto err; - } - return ret; - -err: - kfree(i2c); return ret; } -static int aic3x_i2c_detach(struct i2c_client *client) +static int aic3x_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int aic3x_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, aic3x_codec_probe); -} +static const struct i2c_device_id aic3x_i2c_id[] = { + { "tlv320aic3x", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); /* machine i2c codec control layer */ static struct i2c_driver aic3x_i2c_driver = { @@ -1244,13 +1212,9 @@ static struct i2c_driver aic3x_i2c_drive .name = "aic3x I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = aic3x_i2c_attach, - .detach_client = aic3x_i2c_detach, -}; - -static struct i2c_client client_template = { - .name = "AIC3X", - .driver = &aic3x_i2c_driver, + .probe = aic3x_i2c_probe, + .remove = aic3x_i2c_remove, + .id_table = aic3x_i2c_id, }; static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) @@ -1258,6 +1222,46 @@ static int aic3x_i2c_read(struct i2c_cli value[0] = i2c_smbus_read_byte_data(client, value[0]); return (len == 1); } + +static int aic3x_add_i2c_device(struct platform_device *pdev, + const struct aic3x_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&aic3x_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "tlv320aic3x", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&aic3x_i2c_driver); + return -ENODEV; +} #endif static int aic3x_probe(struct platform_device *pdev) @@ -1290,12 +1294,9 @@ static int aic3x_probe(struct platform_d aic3x_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t) i2c_master_send; codec->hw_read = (hw_read_t) aic3x_i2c_read; - ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = aic3x_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1320,6 +1321,7 @@ static int aic3x_remove(struct platform_ snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&aic3x_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc5.orig/sound/soc/codecs/tlv320aic3x.h 2008-09-01 11:06:20.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/tlv320aic3x.h 2008-09-01 11:06:23.000000000 +0200 @@ -224,6 +224,7 @@ int aic3x_get_gpio(struct snd_soc_codec int aic3x_headset_detected(struct snd_soc_codec *codec); struct aic3x_setup_data { + int i2c_bus; unsigned short i2c_address; unsigned int gpio_func[2]; }; --- linux-2.6.27-rc5.orig/sound/soc/davinci/davinci-evm.c 2008-09-01 11:06:20.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/davinci/davinci-evm.c 2008-09-01 11:06:23.000000000 +0200 @@ -136,6 +136,7 @@ static struct snd_soc_machine snd_soc_ma /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; --- linux-2.6.27-rc5.orig/sound/soc/omap/n810.c 2008-09-01 11:06:20.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/omap/n810.c 2008-09-01 14:29:09.000000000 +0200 @@ -290,6 +290,7 @@ static struct snd_soc_machine snd_soc_ma /* Audio private data */ static struct aic3x_setup_data n810_aic33_setup = { + .i2c_bus = 2, .i2c_address = 0x18, .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, -- Jean Delvare

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Re: [alsa-devel] [PATCH] ASoC: Fix at32-pcm build breakage with PM enabled
by Mark Brown 22 Sep '08

22 Sep '08

On Fri, Sep 19, 2008 at 06:50:45PM +0200, Haavard Skinnemoen wrote: > s/PDC_PTCR/ATMEL_PDC_PTCR/ > > Signed-off-by: Haavard Skinnemoen <haavard.skinnemoen(a)atmel.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Please submit all ALSA patches to alsa-devel (added to the CC). > sound/soc/at32/at32-pcm.c | 5 +++-- > 1 files changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c > index 435f1da..c83584f 100644 > --- a/sound/soc/at32/at32-pcm.c > +++ b/sound/soc/at32/at32-pcm.c > @@ -434,7 +434,8 @@ static int at32_pcm_suspend(struct platform_device *pdev, > params = prtd->params; > > /* Disable the PDC and save the PDC registers */ > - ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); > + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, > + params->mask->pdc_disable); > > prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); > prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); > @@ -464,7 +465,7 @@ static int at32_pcm_resume(struct platform_device *pdev, > ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); > ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); > > - ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); > + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); > return 0; > } > #else /* CONFIG_PM */ > -- > 1.5.6.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > -- "You grabbed my hand and we fell into it, like a daydream - or a fever."

4 5

[alsa-devel] Random controls with hda-intel
by niteblade＠gmx.net 22 Sep '08

22 Sep '08

Hi guys, This is a problem I already posted on alsa-users, but since the only reply I got was a me-too, I'm posting here. I have some problems with the snd-hda-intel driver. The mixer controls seem to be unpredictably appearing and disappearing or changing type (on/off-switch vs volume) from alsa-driver version to version. In the current daily snapshot (12th Sept 2008) I have no volume control at all. In the driver shipped with 2.6.25 kernel at least I get a pcm volume control. Several driver versions ago (dont remember the version number) I had a master volume control. Several other controls seem to have no effect on anything. When loading the snd-hda-intel driver, I get in my syslog: hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... The alsa info script has uploaded my infos here: http://www.alsa-project.org/db/?f=be535315befdb3dbb24430e90e46a7b4ed07a591 Any help is appreciated Kai

2 2

Re: [alsa-devel] DaVinci ASoC DMA stalls
by Mark Lokowich 20 Sep '08

20 Sep '08

I did have to add a getposition accessor to the DaVinci DMA module, which is used by the pcm_pointer function. Other than that, I believe the ASoC should build. My test is simply to run aplay or speaker-test, then quit and run it again. The first run is audible, the second and subsequent runs are not audible until I provide a manual DMA event trigger. Mark Caglar Akyuz wrote: > Mark Lokowich wrote: > > >> I've been using the DaVinci ASoC for a few months and have recently >> upgraded to the 2.6.26 based DaVinci git kernel. Now my audio DMA >> stalls more readily after stopping an active stream. I can manually >> trigger the event by poking the ESR to reactivate the stalled stream, >> suggesting the problem is in the ASP-to-DMA XEVT interface. This >> problem is less prevalent in the 2.6.25 based kernel. Any help? >> >> > > > I don't know if this helps but, I am using ASoC code on my DaVinci board > with latest git kernel(2.6.27) and I have no visible issues with the ASoC > code. However, there may be some code-coverage issue with my usage so if you > have any test method to verify that I really don't have your problem I'll > be happy to perform some tests. > > My real concern is that this code has never been built without any problems here. > First there was a separate asoc branch in davinci tree which failed to built. > I tried with 2.6.25 and there was again build issues. Finally I wasn't able to > build ASoC code with 2.6.27 so I tried to patch it. I think some part of it > relies on some dma code which is not present in davinci tree.[1] I don't know > if everybody else is patching kernel source to build asoc support, maybe I'm > doing something wrong. I'm not sure, just FYI. > > Regards, > Caglar > > [1] http://linux.omap.com/pipermail/davinci-linux-open-source/2008-March/005920… > > >

2 2

[alsa-devel] [PATCH] alsa: Emagic Audiowerk8 low-level ALSA driver
by matthias.nyffenegger＠bluewin.ch 19 Sep '08

19 Sep '08

From: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> Low-level ALSA driver for Emagic Audiowerk8 sound card. Project page: http://sourceforge.net/projects/aw8-alsa Built and tested with Vanilla 2.6.25.16, ALSA 1.0.14a. Signed-off-by: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> --- This is a request for submission to ALSA-tree. URL: http://downloads.sourceforge.net/aw8-alsa/aw-alsa-patch-0.2

2 1

[alsa-devel] [PATCH] ASoC: Analog Devices AD193X I2S codec family driver.
by Manuel Lauss 19 Sep '08

19 Sep '08

Driver for the Analog Devices AD1935/AD1937 I2C and AD1938/AD1939 SPI codecs. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1939.c | 897 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1939.h | 122 ++++++ 4 files changed, 1024 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ad1939.c create mode 100644 sound/soc/codecs/ad1939.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..466d7cb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AD1939 + tristate + config SND_SOC_AK4535 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..cfb5d22 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ad1939-objs := ad1939.o snd-soc-ak4535-objs := ak4535.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o @@ -12,6 +13,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AD1939) += snd-soc-ad1939.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/ad1939.c b/sound/soc/codecs/ad1939.c new file mode 100644 index 0000000..092f09c --- /dev/null +++ b/sound/soc/codecs/ad1939.c @@ -0,0 +1,897 @@ +/* + * AD1935/AD1936/AD1937/AD1938/AD1939 I2S ASoC Codec driver + * + * Copyright (c) 2007-2008 MSC Vertriebsges.m.b.H, www.exm32.com + * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the GPLv2 + * + * Code for the AD193X family of I2S codecs with I2C and SPI control + * interface. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ad1939.h" + +#define CODEC_NAME "ad1939" + +struct ad1939_private { + unsigned char tdm_mode; + unsigned char dev_addr; + unsigned char drvflags; + unsigned char mixpairs; +}; + +/* default register contents after reset */ +static const u16 ad1939_regcache[AD1939_REGCOUNT] = { + 0, 0, 0, 0, 0, 0, 255, 255, 255, 255, 255, 255, 255, 255, 0, 0, 0 +}; + +/* Does the codec have power? In case of STANDBY/OFF BIAS levels, the + * hardware access functions below assume the codec is without power + * to avoid futile bus transactions. + */ +static int codec_is_powered(struct snd_soc_codec *c) +{ + return (c->bias_level == SND_SOC_BIAS_PREPARE) || + (c->bias_level == SND_SOC_BIAS_ON); +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ad1939_i2c_write(struct snd_soc_codec *codec, unsigned int r, + unsigned int v) +{ + struct i2c_msg msg; + struct i2c_client *c; + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + c = (struct i2c_client *)codec->control_data; + data[0] = r & 0xff; + data[1] = v & 0xff; + msg.addr = c->addr; + msg.flags = 0; /* write */ + msg.buf = &data[0]; + msg.len = 2; + + /* no power? just update the cache then */ + if (codec_is_powered(codec)) + ret = i2c_transfer(c->adapter, &msg, 1); + else + ret = 1; + + if (ret == 1) + cache[r] = v; + return (ret == 1) ? 0 : -EIO; +} + +static unsigned int ad1939_i2c_read(struct snd_soc_codec *codec, + unsigned int r) +{ + struct i2c_msg msg[2]; + struct i2c_client *c; + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + /* no power? cached value is all we have */ + if (!codec_is_powered(codec)) + return cache[r]; + + c = (struct i2c_client *)codec->control_data; + data[0] = r & 0xff; + msg[0].addr = c->addr; + msg[0].flags = 0; + msg[0].buf = &data[0]; + msg[0].len = 1; + + msg[1].addr = c->addr; + msg[1].flags = I2C_M_RD; + msg[1].buf = &data[1]; + msg[1].len = 1; + + ret = i2c_transfer(c->adapter, &msg[0], 2); + if (ret == 2) + cache[r] = data[1]; + + return (ret == 2) ? data[1] : -EIO; +} +#endif + +#if defined(CONFIG_SPI) || defined(CONFIG_SPI_MODULE) +/* SPI communications for AD1938/AD1939: + * 24 bit data, LSB first; (I2C style, including R/W bit) + * <7bit global address|R/W#><8bit register address><8bit reg data> + */ +static int ad1939_spi_write(struct snd_soc_codec *codec, unsigned int r, + unsigned int v) +{ + struct spi_device *spi = codec->control_data; + struct ad1939_private *ad = codec->private_data; + u16 *cache = codec->reg_cache; + int ret; + u8 data[3]; + + data[0] = ad->dev_addr << 1; + data[1] = r; + data[2] = v; + + /* no power? just update the cache then */ + if (codec_is_powered(codec)) + ret = spi_write(spi, &data[0], 3); + else + ret = 0; + + if (ret == 0) + cache[r] = v; + + return ret; +} + +static unsigned int ad1939_spi_read(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + struct ad1939_private *ad = codec->private_data; + u16 *cache = codec->reg_cache; + u8 data_w[3], data_r[3]; + int ret; + + /* no power? cached value is all we have */ + if (!codec_is_powered(codec)) + return cache[r]; + + data_r[0] = data_w[0] = (ad->dev_addr << 1) | 1; + data_r[1] = data_w[1] = r; + data_r[2] = data_w[2] = cache[r]; + + ret = spi_write_then_read(spi, data_w, 3, data_r, 3); + + if (ret == 0) + cache[r] = data_r[2]; + return (ret) ? ret : data_r[2]; +} +#endif + +/* standard accessor functions. Reads can generally be satisfied by + * the cached value, writes don't hit the chip if the cached value + * is identical. + */ +static inline unsigned int ad1939_read(struct snd_soc_codec *codec, + unsigned int r) +{ + unsigned int v; + u16 *cache = codec->reg_cache; + + /* the PLLCTL1 has one read-only bit: PLL lock indicator. + * all other regs keep what was set. + * If powered-down the chip can't be reached so just return + * the cached value. + */ + if (unlikely(r == AD1939_PLLCTL1)) + v = codec->read(codec, r); + else + v = cache[r]; + + return v; +} + +static inline int ad1939_write(struct snd_soc_codec *codec, + unsigned int r, unsigned int v) +{ + u16 *cache = codec->reg_cache; + if (cache[r] == v) + return 0; + return codec->write(codec, r, v); +} + +/***** controls ******/ + +static const char *dac_deemph[] = {"Flat", "48kHz", "44.1kHz", "32kHz"}; +static const char *dac_outpol[] = {"Normal", "Inverted"}; + +static const struct soc_enum ad1939_enum[] = { + /*SOC_ENUM_SINGLE(register, startbit, choices, choices-texts) */ + SOC_ENUM_SINGLE(AD1939_DACCTL2, 1, 4, dac_deemph), + SOC_ENUM_SINGLE(AD1939_DACCTL2, 5, 2, dac_outpol), +}; + +/* Mixer controls. Keep the Playback Attenuation controls at the top, + * or the limiter breaks (see ad1939_add_controls()) + */ +static const struct snd_kcontrol_new ad1939_snd_ctls[] = { +SOC_DOUBLE_R("Master Playback", AD1939_VOL1L, AD1939_VOL1R, 0, 255, 1), +SOC_DOUBLE_R("Channel 2 Playback", AD1939_VOL2L, AD1939_VOL2R, 0, 255, 1), +SOC_DOUBLE_R("Channel 3 Playback", AD1939_VOL3L, AD1939_VOL3R, 0, 255, 1), +SOC_DOUBLE_R("Channel 4 Playback", AD1939_VOL4L, AD1939_VOL4R, 0, 255, 1), +SOC_ENUM("DAC Deemphasis", ad1939_enum[0]), +SOC_ENUM("DAC output polarity", ad1939_enum[1]), +}; + +/* add non dapm controls */ +static int ad1939_add_controls(struct snd_soc_codec *codec) +{ + struct ad1939_private *ad = codec->private_data; + int err, i; + + for (i = 0; i < ARRAY_SIZE(ad1939_snd_ctls); i++) { + if ((i <= 3) && (i >= ad->mixpairs)) + continue; + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ad1939_snd_ctls[i], codec, NULL)); + if (err < 0) + return err; + } + return 0; +} + +/***** chip interface config ******/ + +static int ad1939_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ad1939_private *ad = codec->private_data; + unsigned char dac0, dac1, dac2, adc0, adc1, adc2; + unsigned long rate; + unsigned int bits; + + dac0 = ad1939_read(codec, AD1939_DACCTL0); + dac1 = ad1939_read(codec, AD1939_DACCTL1); + dac2 = ad1939_read(codec, AD1939_DACCTL2); + adc0 = ad1939_read(codec, AD1939_ADCCTL0); + adc1 = ad1939_read(codec, AD1939_ADCCTL1); + adc2 = ad1939_read(codec, AD1939_ADCCTL2); + + rate = params_rate(params); + bits = params->msbits; + + /* sample rate */ + dac0 &= ~(3<<1); /* 48kHz */ + adc0 &= ~(3<<6); /* 48kHz */ + switch (rate) { + case 32000 ... 48000: + break; + case 64000 ... 96000: + dac0 |= (1<<1); + adc0 |= (1<<6); + break; + case 128000 ... 192000: + dac0 |= (2<<1); + adc0 |= (2<<6); + break; + default: + return -EINVAL; + } + + /* sample width (bits) */ + dac2 &= ~(3<<3); /* 24 bits */ + adc1 &= ~(3<<0); /* 24 bits */ + + + /* channels */ + dac0 &= ~(3<<6); /* DAC I2S stereo */ + dac1 &= ~(3<<1); /* 2 channels */ + adc1 &= ~(3<<5); /* ADC I2S stereo */ + adc2 &= ~(3<<4); /* 2 channels */ + switch (params_channels(params)) { + case 2: /* I2S stereo mode */ + if (ad->drvflags & AD1939_DRV_TDM_STEREO) { + dac0 |= (ad->tdm_mode & 3) << 6; + adc1 |= (ad->tdm_mode & 3) << 5; + } + break; + case 4: /* TDM mode */ + dac0 |= (ad->tdm_mode & 3) << 6; + dac1 |= (1<<1); + adc1 |= (ad->tdm_mode & 3) << 5; + adc2 |= (1<<4); + break; + case 8: /* TDM mode */ + dac0 |= (ad->tdm_mode & 3) << 6; + dac1 |= (2<<1); + adc1 |= (ad->tdm_mode & 3) << 5; + adc2 |= (2<<4); + break; + case 16: /* TDM mode */ + dac0 |= (ad->tdm_mode & 3) << 6; + dac1 |= (3<<1); + adc1 |= (ad->tdm_mode & 3) << 5; + adc2 |= (3<<4); + break; + default: + return -EINVAL; + } + + ad1939_write(codec, AD1939_DACCTL0, dac0); + ad1939_write(codec, AD1939_DACCTL1, dac1); + ad1939_write(codec, AD1939_DACCTL2, dac2); + ad1939_write(codec, AD1939_ADCCTL0, adc0); + ad1939_write(codec, AD1939_ADCCTL1, adc1); + ad1939_write(codec, AD1939_ADCCTL2, adc2); + + return 0; +} + +static int ad1939_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ad1939_private *ad = codec->private_data; + unsigned char dac0, dac1, adc1, adc2; + + dac0 = ad1939_read(codec, AD1939_DACCTL0); + dac1 = ad1939_read(codec, AD1939_DACCTL1); + adc1 = ad1939_read(codec, AD1939_ADCCTL1); + adc2 = ad1939_read(codec, AD1939_ADCCTL2); + + /* codec clocks master/slave setup */ + dac1 &= ~((1<<4) | (1<<5)); /* LRCK BCK slave */ + adc2 &= ~((1<<3) | (1<<6)); /* LRCK BCK slave */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* BCK/LCK master */ + dac1 |= (1<<4) | (1<<5); /* LRCK BCK master */ + adc2 |= (1<<3) | (1<<6); /* LRCK BCK master */ + if (ad->drvflags & AD1939_DRV_ADCDAC_COMMON_BCK) { + if (ad->drvflags & AD1939_DRV_ADC_BCK_MASTER) + dac1 &= ~(1<<5); /* DAC BCLK slave */ + else + adc2 &= ~(1<<6); /* ADC BCLK slave */ + } + if (ad->drvflags & AD1939_DRV_ADCDAC_COMMON_LRCK) { + if (ad->drvflags & AD1939_DRV_ADC_LRCK_MASTER) + dac1 &= ~(1<<4); /* DAC LRCK slave */ + else + adc2 &= ~(1<<3); /* ADC LRCK slave */ + } + break; + + case SND_SOC_DAIFMT_CBS_CFS: /* BCK/LCK slave */ + break; + + case SND_SOC_DAIFMT_CBM_CFS: /* BCK master, LRCK slave */ + dac1 &= ~(1<<4); /* DAC LRCK slave */ + adc2 &= ~(1<<3); /* ADC LRCK slave */ + dac1 |= (1<<5); /* DAC BCK master */ + adc2 |= (1<<6); /* ADC BCK master */ + if (ad->drvflags & AD1939_DRV_ADCDAC_COMMON_BCK) { + if (ad->drvflags & AD1939_DRV_ADC_BCK_MASTER) + dac1 &= ~(1<<5); /* DAC BCLK slave */ + else + adc2 &= ~(1<<6); /* ADC BCLK slave */ + } + break; + + case SND_SOC_DAIFMT_CBS_CFM: + dac1 &= ~(1<<5); /* DAC BCK slave */ + adc2 &= ~(1<<6); /* ADC BCK slave */ + dac1 |= (1<<4); /* DAC LRCK master */ + adc2 |= (1<<3); /* ADC LRCK master */ + if (ad->drvflags & AD1939_DRV_ADCDAC_COMMON_LRCK) { + if (ad->drvflags & AD1939_DRV_ADC_LRCK_MASTER) + dac1 &= ~(1<<4); /* DAC LRCK slave */ + else + adc2 &= ~(1<<3); /* ADC LRCK slave */ + } + break; + + default: + return -EINVAL; + } + + /* interface format */ + dac0 &= ~(7<<3); /* DAC: SDATA delay 1 (I2S) */ + adc1 &= ~(7<<2); /* ADC: SDATA delay 1 (I2S) */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_MSB: /* LEFT_J */ + dac0 |= (1<<3); /* no SDATA delay */ + adc1 |= (1<<2); /* no SDATA delay */ + break; + + default: + return -EINVAL; + } + + /* clock format */ + dac1 &= ~((1<<7) | (1<<3)); /* norm BCK LRCK */ + adc2 &= ~((1<<1) | (1<<2)); /* norm BCK LRCK */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_NB_IF: + dac1 |= (1<<3); /* inv LRCK */ + adc2 |= (1<<2); + break; + + case SND_SOC_DAIFMT_IB_NF: + dac1 |= (1<<7); /* inv BCK */ + adc2 |= (1<<1); + break; + + case SND_SOC_DAIFMT_IB_IF: + dac1 |= (1<<3) | (1<<7); /* inv LRCK BCK */ + adc2 |= (1<<1) | (1<<2); + break; + + default: + return -EINVAL; + } + + ad1939_write(codec, AD1939_DACCTL0, dac0); + ad1939_write(codec, AD1939_DACCTL1, dac1); + ad1939_write(codec, AD1939_ADCCTL1, adc1); + ad1939_write(codec, AD1939_ADCCTL2, adc2); + + return 0; +} + +static int ad1939_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned char pll0, adc0, dac0; + u16 *cache = codec->reg_cache; + int i; + + pll0 = ad1939_read(codec, AD1939_PLLCTL0) & 0xfe; + dac0 = ad1939_read(codec, AD1939_DACCTL0) & 0xfe; + adc0 = ad1939_read(codec, AD1939_ADCCTL0) & 0xfe; + + switch (level) { + case SND_SOC_BIAS_ON: /* power up */ + case SND_SOC_BIAS_PREPARE: + if (!codec_is_powered(codec)) { + /* writes need to hit the chip */ + codec->bias_level = level; + + for (i = 0; i < AD1939_REGCOUNT; i++) + codec->write(codec, i, cache[i]); + + ad1939_write(codec, AD1939_PLLCTL0, pll0); + ad1939_write(codec, AD1939_DACCTL0, dac0); + ad1939_write(codec, AD1939_ADCCTL0, adc0); + } + break; + + case SND_SOC_BIAS_STANDBY: /* power down */ + case SND_SOC_BIAS_OFF: + if (codec_is_powered(codec)) { + /* turn off internal PLL and DAC/ADCs */ + ad1939_write(codec, AD1939_PLLCTL0, pll0 | 1); + ad1939_write(codec, AD1939_DACCTL0, dac0 | 1); + ad1939_write(codec, AD1939_ADCCTL0, adc0 | 1); + } + break; + } + codec->bias_level = level; + return 0; +} + +static int ad1939_digmute(struct snd_soc_dai *dai, int mute) +{ + ad1939_write(dai->codec, AD1939_DACMUTE, mute ? 0xff : 0); + return 0; +} + +static int ad1939_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + + switch (freq) { + case 12288000: + ret = 0; /* I know this one works okay */ + default: + ret = -EINVAL; /* don't know about others */ + } + return ret; +} + +#define AD1939_RATES \ + (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) + +#define AD1939_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai ad1939_dai = { + .name = CODEC_NAME, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 4, /* 4/8 in single/dualline TDM */ + .rates = AD1939_RATES, + .formats = AD1939_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, /* 2 ADCs */ + .rates = AD1939_RATES, + .formats = AD1939_FORMATS, + }, + .ops = { + .hw_params = ad1939_hw_params, + }, + .dai_ops = { + .digital_mute = ad1939_digmute, + .set_sysclk = ad1939_set_dai_sysclk, + .set_fmt = ad1939_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(ad1939_dai); + +static int ad1939_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + struct ad1939_setup_data *setup = socdev->codec_data; + struct ad1939_private *ad = codec->private_data; + unsigned char r0, r1; + int ret; + + codec->name = CODEC_NAME; + codec->owner = THIS_MODULE; + codec->set_bias_level = ad1939_set_bias_level; + codec->dai = &ad1939_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ad1939_regcache); + codec->reg_cache = kmemdup(ad1939_regcache, + sizeof(ad1939_regcache), + GFP_KERNEL); + if (!codec->reg_cache) + return -ENOMEM; + + /* assume chip has no power. This way, all writes go straight + * to the cache. Once ASoC decides to wake us up, the bias + * handler will write the cache straight to the chip. + */ + codec->bias_level = SND_SOC_BIAS_OFF; + + /* remember TDM mode and set up internal clock routing */ + ad->tdm_mode = setup->tdm_mode; + ad->drvflags = setup->drvflags; + + /* limit output attenuation controls to requested number */ + ad->mixpairs = setup->mixpairs; + if ((ad->mixpairs < 1) || (ad->mixpairs > 4)) + ad->mixpairs = 4; + + /* use default TDM mode if noone wants one */ + if ((ad->tdm_mode > AD1939_TDM_MODE_DUALLINE) || (ad->tdm_mode < 1)) + ad->tdm_mode = AD1939_TDM_MODE_TDM; + + /* setup clocks */ + r0 = ad1939_read(codec, AD1939_PLLCTL0) & ~(0xf << 3); + r1 = ad1939_read(codec, AD1939_PLLCTL1) & 3; + + r0 |= (setup->pll_src & 3) << 5; + r0 |= (1<<7); /* enable internal master clock */ + r0 |= (setup->mclk_xo & 3) << 3; + r1 |= setup->dac_adc_clksrc & 7; /* DACclk/ADCclk/VREF */ + r1 ^= AD1939_CLKSRC_ENABLE_ONCHIP_VREF; /* VREF bit is inverted */ + + ad1939_write(codec, AD1939_PLLCTL0, r0); + ad1939_write(codec, AD1939_PLLCTL1, r1); + + /* Bitclock sources for the ADC and DAC I2S interfaces */ + r0 = ad1939_read(codec, AD1939_DACCTL1); + r1 = ad1939_read(codec, AD1939_ADCCTL2); + r0 &= ~AD1939_BCLKSRC_DAC_PLL; + r1 &= ~AD1939_BCLKSRC_ADC_PLL; + r0 |= setup->dac_adc_clksrc & AD1939_BCLKSRC_DAC_PLL; + r1 |= setup->dac_adc_clksrc & AD1939_BCLKSRC_ADC_PLL; + ad1939_write(codec, AD1939_DACCTL1, r0); + ad1939_write(codec, AD1939_ADCCTL2, r1); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, + SNDRV_DEFAULT_STR1); + if (unlikely(ret < 0)) { + printk(KERN_ERR CODEC_NAME ": cannot create pcms\n"); + goto pcm_err; + } + + ad1939_add_controls(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR CODEC_NAME ": cannot register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static void ad1939_deinit(struct snd_soc_device *socdev) +{ + if (socdev) { + ad1939_set_bias_level(socdev->codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + /* free memory allocated in ad1939_probe() */ + kfree(socdev->codec->private_data); + kfree(socdev->codec->reg_cache); + kfree(socdev->codec); + } +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ad1939_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = client->dev.platform_data; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = client; + i2c_set_clientdata(client, socdev); + + codec->read = ad1939_i2c_read; + codec->write = ad1939_i2c_write; + + ret = ad1939_init(socdev); + if (ret == 0) + return 0; + + printk(KERN_ERR CODEC_NAME ": i2c codec init failed\n"); + + /* undo allocations made by platform probe */ + kfree(codec->private_data); + kfree(codec); + return ret; +} + +static int ad1939_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_device *socdev = i2c_get_clientdata(client); + + ad1939_deinit(socdev); + + return 0; +} + +#define ad1939_i2c_suspend NULL +#define ad1939_i2c_resume NULL + +static const struct i2c_device_id ad1939_i2c_id[] = { + { "ad1939", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ad1939_i2c_id); + +static struct i2c_driver ad1939_i2c_driver = { + .driver = { + .name = "ad1939", + .owner = THIS_MODULE, + }, + .probe = ad1939_i2c_probe, + .remove = ad1939_i2c_remove, + .suspend = ad1939_i2c_suspend, + .resume = ad1939_i2c_resume, + .id_table = ad1939_i2c_id, +}; + +static int ad1939_add_i2c_device(struct platform_device *pdev, + struct ad1939_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ad1939_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->dev_address; + info.platform_data = platform_get_drvdata(pdev); + strlcpy(info.type, "ad1939", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&ad1939_i2c_driver); + return -ENODEV; +} +#endif /* I2C */ + +#if defined(CONFIG_SPI) && defined(CONFIG_SPI_MASTER) +static int ad1939_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = spi->dev.platform_data; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + codec->read = ad1939_spi_read; + codec->write = ad1939_spi_write; + + spi_set_drvdata(spi, socdev); + + ret = ad1939_init(socdev); + if (ret == 0) + return 0; + + printk(KERN_ERR CODEC_NAME ": SPI codec init failed\n"); + + /* undo allocations made by platform probe */ + kfree(codec->private_data); + kfree(codec); + return ret; +} + +static int ad1939_spi_remove(struct spi_device *spi) +{ + struct snd_soc_device *socdev = spi_get_drvdata(spi); + ad1939_deinit(socdev); + return 0; +} + +#define ad1939_spi_suspend NULL +#define ad1939_spi_resume NULL + +static struct spi_driver ad1939_spi_driver = { + .driver = { + .name = "ad1939", + .owner = THIS_MODULE, + }, + .probe = ad1939_spi_probe, + .remove = ad1939_spi_remove, + .suspend = ad1939_spi_suspend, + .resume = ad1939_spi_resume, +}; + +static int ad1939_add_spi_device(struct platform_device *pdev, + struct ad1939_setup_data *setup) +{ + struct spi_master *master; + struct spi_device *spidev; + int ret; + + ret = spi_register_driver(&ad1939_spi_driver); + if (ret) { + dev_err(&pdev->dev, "can't add spi driver\n"); + return ret; + } + + ret = -ENODEV; + + /* pass on socdev to spi probe (spi->dev.platform_data) */ + setup->spi_info.platform_data = platform_get_drvdata(pdev); + + master = spi_busnum_to_master(setup->spi_info.bus_num); + if (!master) { + dev_err(&pdev->dev, "can't get spi master\n"); + goto out; + } + + spidev = spi_new_device(master, &setup->spi_info); + spi_master_put(master); + if (!spidev) { + dev_err(&pdev->dev, "can't register spi device\n"); + goto out; + } + + return 0; + +out: + spi_unregister_driver(&ad1939_spi_driver); + return ret; +} +#endif /* SPI */ + +static int ad1939_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ad1939_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec; + struct ad1939_private *ad; + int ret; + + ret = -ENOMEM; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + goto out; + + ad = kzalloc(sizeof(struct ad1939_private), GFP_KERNEL); + if (ad == NULL) + goto out1; + + codec->private_data = ad; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ad->dev_addr = setup->dev_address; /* I2C/SPI device addr */ + + ret = -ENODEV; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->bus_mode == AD1939_BUS_MODE_I2C) + ret = ad1939_add_i2c_device(pdev, setup); +#endif +#if defined(CONFIG_SPI) && defined(CONFIG_SPI_MASTER) + if (setup->bus_mode == AD1939_BUS_MODE_SPI) + ret = ad1939_add_spi_device(pdev, setup); +#endif + + if (!ret) + return 0; + + kfree(ad); +out1: + kfree(codec); +out: + return ret; +} + +static int ad1939_remove(struct platform_device *pdev) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ad1939_i2c_driver); +#endif +#if defined(CONFIG_SPI) || defined(CONFIG_SPI_MODULE) + spi_unregister_driver(&ad1939_spi_driver); +#endif + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad1939 = { + .probe = ad1939_probe, + .remove = ad1939_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad1939); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ASoC AD1935-AD1939 I2S Codec family driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/codecs/ad1939.h b/sound/soc/codecs/ad1939.h new file mode 100644 index 0000000..72931d5 --- /dev/null +++ b/sound/soc/codecs/ad1939.h @@ -0,0 +1,122 @@ +/* + * AD1935/AD1936/AD1937/AD1938/AD1939 I2S ASoC Codec driver + * + * Copyright (c) 2007-2008 MSC Vertriebsges.m.b.H, www.exm32.com + * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the GPLv2 + * + */ + +#ifndef _AD1939_H_ +#define _AD1939_H_ + +#include <linux/spi/spi.h> + +#define AD1939_PLLCTL0 0x00 +#define AD1939_PLLCTL1 0x01 +#define AD1939_DACCTL0 0x02 +#define AD1939_DACCTL1 0x03 +#define AD1939_DACCTL2 0x04 +#define AD1939_DACMUTE 0x05 +#define AD1939_VOL1L 0x06 +#define AD1939_VOL1R 0x07 +#define AD1939_VOL2L 0x08 +#define AD1939_VOL2R 0x09 +#define AD1939_VOL3L 0x0A +#define AD1939_VOL3R 0x0B +#define AD1939_VOL4L 0x0C +#define AD1939_VOL4R 0x0D +#define AD1939_ADCCTL0 0x0E +#define AD1939_ADCCTL1 0x0F +#define AD1939_ADCCTL2 0x10 + +#define AD1939_REGCOUNT 0x11 + +/* + * AD1939 setup data + */ + +/* TDM modes. Have a look at the manual to understand what these do. */ +#define AD1939_TDM_MODE_TDM 1 +#define AD1939_TDM_MODE_AUX 2 +#define AD1939_TDM_MODE_DUALLINE 3 + +/* Master PLL clock source, select one */ +#define AD1939_PLL_SRC_MCLK 0 /* external clock */ +#define AD1939_PLL_SRC_DACLRCK 1 /* get from DAC LRCLK */ +#define AD1939_PLL_SRC_ADCLRCK 2 /* get from ADC LRCLK */ + +/* clock sources for ADC, DAC. Refer to the manual for more information + * (for 192000kHz modes, internal PLL _MUST_ be used). Select one for ADC + * and DAC. + */ +#define AD1939_CLKSRC_DAC_PLL 0 /* DAC clocked by int. PLL */ +#define AD1939_CLKSRC_DAC_MCLK (1<<0) /* DAC clocked by ext. MCK */ +#define AD1939_CLKSRC_ADC_PLL 0 /* ADC clocked by int. PLL */ +#define AD1939_CLKSRC_ADC_MCLK (1<<1) /* ADC clocked by ext. MCK */ +#define AD1939_CLKSRC_ENABLE_ONCHIP_VREF (1<<2) + +/* I2S Bitclock sources for DAC and ADC I2S interfaces. + * OR it to ad1939_setup_data.dac_adc_clksrc. Select one for ADC and DAC. + */ +#define AD1939_BCLKSRC_DAC_EXT 0 /* DAC I2SCLK from DBCLK pin */ +#define AD1939_BCLKSRC_DAC_PLL (1<<6) /* DAC I2SCLK from int. PLL */ +#define AD1939_BCLKSRC_ADC_EXT 0 /* DAC I2SCLK from DBCLK pin */ +#define AD1939_BCLKSRC_ADC_PLL (1<<7) /* DAC I2SCLK from int. PLL */ + +/* MCLK_XO pin configuration */ +#define AD1939_MCLKXO_MCLKXI 0 /* mirror MCLK_XI pin */ +#define AD1939_MCLKXO_256FS 1 +#define AD1939_MCLKXO_512FS 2 +#define AD1939_MCLKXO_OFF 3 /* disable MCLK_XO output */ + +/* driver specific flags */ +/* specify these flags if the LRCK and/or BCK pins of the ADC and DAC + * parts are wired together on the PCB; to prevent both units from driving + * the pin and resulting bad signals. You then have to specify WHICH + * unit gets to be the Master (clock generator) and which is slave. + * NOTE: this is only used if the CODEC is configured as either BCK or + * LRCK master; if the codec is BCK/LRCK slave (BCK and LRCK are driven + * by external components) these settings are ignored! + */ +#define AD1939_DRV_ADCDAC_COMMON_BCK (1<<0) +#define AD1939_DRV_ADCDAC_COMMON_LRCK (1<<1) + +/* define which unit gets to drive the BCK/LRCK pins if the codec is + * required to be either BCK or LRCK master + */ +#define AD1939_DRV_DAC_LRCK_MASTER 0 +#define AD1939_DRV_ADC_LRCK_MASTER (1<<2) +#define AD1939_DRV_DAC_BCK_MASTER 0 +#define AD1939_DRV_ADC_BCK_MASTER (1<<3) + +/* use TDM mode even for stereo (2-channel) signals */ +#define AD1939_DRV_TDM_STEREO (1<<4) + +#define AD1939_BUS_MODE_I2C 1 +#define AD1939_BUS_MODE_SPI 2 + +struct ad1939_setup_data { + /* control interface configuration */ + /* device address, WITHOUT the R/W bit! (default 0x04) */ + unsigned char dev_address; /* I2C or SPI device address */ + unsigned char bus_mode; /* which framework to register with */ + + unsigned char i2c_bus; /* I2C bus the device is on */ + + struct spi_board_info spi_info; /* SPI conn info */ + + /* configuration */ + unsigned char tdm_mode; /* one of AD1939_TDM_MODE_* */ + unsigned char pll_src; /* one of AD1939_PLL_SRC_* */ + unsigned char dac_adc_clksrc; /* AD1939_{B,}CLKSRC_* or'ed */ + unsigned char mclk_xo; /* one of AD1939_MCLKXO_* */ + unsigned char drvflags; /* driver flags */ + unsigned char mixpairs; /* mixer pairs (L-R) to advertise */ +}; + +extern struct snd_soc_codec_device soc_codec_dev_ad1939; +extern struct snd_soc_dai ad1939_dai; + +#endif -- 1.6.0.1

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[alsa-devel] [PATCH] ASoC V2: Remove check for fsl, eloplus-dma-channel from MPC8610 sound driver
by Timur Tabi 19 Sep '08

19 Sep '08

Remove the check in the Freescale MPC8610 sound driver for the "compatible" property in the DMA channels. Normally, the DMA channels in the device tree have "fsl,eloplus-dma-channel" in the compatible property. However, this allows the generic DMA driver to claim those channels, which conflicts with the sound driver. To resolve this, the channels used by the sound drivers should have "fsl,ssi-dma-channel" in the compatible property. In order for the sound driver to work with either value, we simply remove all checks for the compatible property. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2 only. sound/soc/fsl/fsl_ssi.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 867e770..871ba8e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -628,9 +628,6 @@ static struct device_node *find_dma_node(unsigned int controller, } for_each_child_of_node(np, np2) { - if (!of_device_is_compatible(np2, "fsl,eloplus-dma-channel")) - continue; - iprop = of_get_property(np2, "cell-index", NULL); if (!iprop) { pr_err("fsl-ssi: cell-index property not found\n"); @@ -640,13 +637,11 @@ static struct device_node *find_dma_node(unsigned int controller, break; } - if (!np2) { + of_node_put(np); + + if (!np2) pr_err("fsl-ssi: cannot find node for DMA channel %u\n", channel); - return NULL; - } - - of_node_put(np); return np2; } -- 1.5.5

2 1

[alsa-devel] ALSA: hda - Add PCI subsystem ID for MacBookPro4,1
by David Woodhouse 19 Sep '08

19 Sep '08

This seems to work if I tell it that it's a mbp3. Not sure if there are actually any differences. Signed-off-by: David Woodhouse <David.Woodhouse(a)intel.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6602516..e9b2cbd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6557,6 +6557,7 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ + case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; -- David Woodhouse Open Source Technology Centre David.Woodhouse(a)intel.com Intel Corporation

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[alsa-devel] [PATCH] ALSA: hda - Fix ALC662 DAC mixer mutes also for auto config model
by Herton Ronaldo Krzesinski 19 Sep '08

19 Sep '08

In previous change "[ALSA] hda-codec - Fix ALC662 DAC mixer mutes", I missed to fix the mixer mute switches also for the auto config model of ALC662. Now mute for mixer items "Front", "Surround", "Center" and "LFE" when available will work too with "auto" model. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff -p -up linux-2.6.27/sound/pci/hda/patch_realtek.c.orig linux-2.6.27/sound/pci/hda/patch_realtek.c --- linux-2.6.27/sound/pci/hda/patch_realtek.c.orig 2008-09-18 16:24:50.000000000 -0300 +++ linux-2.6.27/sound/pci/hda/patch_realtek.c 2008-09-18 16:25:35.000000000 -0300 @@ -15975,15 +15975,15 @@ static int alc662_auto_create_multi_out_ HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, + err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 2, + HDA_COMPOSE_AMP_VAL(0x0e, 1, 0, HDA_INPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, + err = add_control(spec, ALC_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 2, + HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_INPUT)); if (err < 0) return err; @@ -15995,9 +15995,9 @@ static int alc662_auto_create_multi_out_ if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 2, - HDA_INPUT)); + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i), + 3, 0, HDA_INPUT)); if (err < 0) return err; } -- []'s Herton

2 1

[alsa-devel] X-Fi support in ALSA.
by James Courtier-Dutton 18 Sep '08

18 Sep '08

Hi, I have now had a chance to read some of the specs of the X-Fi cards. aka. emu20k1. It turns out that some of the X-Fi cards have a compatibility mode. By default, they boot up in compatibility mode. The interesting point is that this "compatibility mode" is in fact the intel-HD standard so could use the intel-hda driver. For one X-Fi card with IDs: Device 1102:0009 Subsystem: 1102:0010 works a bit with the ALSA hda-intel driver. So far, the off the shelf hda-intel driver, with simple PCIs adjustment, works for the above card, but only sound capture works so far, so probably some quirk would be needed. This compatibility mode works with the Windows Vista UAA drivers. Not all X-Fi cards support with compatibility mode. For example, any card with (after a cold boot into Linux) Device 1102:0005 Subsystem: xxxx:xxxx will be X-Fi only. With regard to actual full X-Fi support, I am still in the progress of creating a GPL emu20k1.h file, that is the first step towards an snd-emu20k1 native ALSA driver. James

3 3

Re: [alsa-devel] Does LKML accept non-ASOC audio driver?
by Mark Brown 18 Sep '08

18 Sep '08

On Thu, Sep 18, 2008 at 11:01:07AM +0000, Cliff Cai wrote: > > outside of ASoC with just a platform data hook for doing the codec > > initialisation would probably be the best bet. Could you point me at > > the existing drivers? > One thing I should let you know ,the initialisation code will highly depend > on the low-level SPORT operation code in blackfin arch. That's absolutely fine - the machine driver is entirely specific to the hardware it's connecting so things like this are not a problem. It already explicitly references the codec and machine drivers anyway. > > Actually, looking at the datasheet for the AD73311 it is actually > > possible to reconfigure the codec at run time at the expense of one data > > bit. > You are right,but it's not a good way in real use,since we need to format every > PCM sample in that mode. Yeah, without trusting the application data to leave the bit clear it'd need some glue in hardware to force the MSB low when not doing control operations.

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Re: [alsa-devel] Does LKML accept non-ASOC audio driver?
by Mark Brown 18 Sep '08

18 Sep '08

On Thu, Sep 18, 2008 at 09:48:20AM +0000, Cliff Cai wrote: > Yes,the hardware doesn't have seperate codec control, > we use the same bus as for DMA transmission to configure the codec( e.g.AD73311) That shouldn't be an issue, AC97 does the same thing but... > and after configuring it,the codec will enter data mode,and is not configurable any more. ...this might be more of an issue. That said, is the SPORT on the DSP likely to be used with smarter codecs (it looks like it's already used by your existing ASoC drivers but perhaps differently?)? If it is then it's probably worth fitting it in to ASoC in order to avoid having two distinct drivers for it. It should be possible for the machine driver to do the codec setup when it probes and then just have a dumb DAI for the codec with no operations (actually, it's worth adding that dumb codec driver anyway - it's not like dumb ADCs and DACs are uncommon). If there's not much code to reuse then a simpler driver for the SPORT outside of ASoC with just a platform data hook for doing the codec initialisation would probably be the best bet. Could you point me at the existing drivers? Actually, looking at the datasheet for the AD73311 it is actually possible to reconfigure the codec at run time at the expense of one data bit.

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Re: [alsa-devel] Does LKML accept non-ASOC audio driver?
by Mark Brown 18 Sep '08

18 Sep '08

On Thu, Sep 18, 2008 at 09:16:08AM +0000, Cliff Cai wrote: > We(blackfin arch) have several audio codec drivers which are written in alsa style,actually they are hard to transfer to asoc. What are the problems with this - is it a case of the hardware not fitting the model well or are there other difficulties with the ASoC framework? > I also saw that there are several arch-based directories in sound/, e.g.sound/arm, > does this mean non-asoc driver is also accepted currently? Yes, ASoC only needs to be used in situations where it makes sense to do so. If the hardware doesn't have separate codec and DMA controllers, for example, then it wouldn't make much sense to try to shoehorn it into ASoC.

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[alsa-devel] scheduling midi events with pyalsa and making queue to loop after few bars..
by ki.ber.kom.uni.st 18 Sep '08

18 Sep '08

hm.. i would like to make simple example of scheduling few midi notes to queue and alsaseq where the length of the loop will be one bar or any number of ticks... i'm kind of stuck :( i got two midi notes triggered (event & event2) but don't know how to play the queue again or in loop while notes which are already there should be played in every repetition... i'm trying to do that placing alsaseq.SEQ_EVENT_SETPOS_TICK at 0 (or 1) but it doesn't play the notes again... my logic is to schedule the playhead of queue at the beginning after some number of ticks and i assume that already queued notes will be there.. i'm quite sure i'm wrong ;) what would be good logic to do that? do i need to leave queue alone and implement my own logic on top of it? something like adding notes all along the timeline simulating the repetition that way... i would like to know or to check what's the tick of running queue.. do i do that scheduling alsaseq.SEQ_EVENT_ECHO some place on the timeline? hm... lot's of questions... hope someone would explain to me a bit of the alsaseq inner logic... i'm sorry i couldn't find good documentation but testing the library inside ipython.... here is how my session looks like (i connect "myPort" to Virtual Raw Midi with a little help of qjackctl and monitor it with amidi -p hw:2,0 -d): from pyalsa import alsaseq seqsen = alsaseq.Sequencer() seqsen.create_simple_port("myPort", alsaseq.SEQ_PORT_TYPE_APPLICATION, alsaseq.SEQ_PORT_CAP_SUBS_READ | alsaseq.SEQ_PORT_CAP_READ) queue = seqsen.create_queue() seqsen.start_queue(queue) queue = seqsen.create_queue() seqsen.start_queue(queue) tempo, ppq = seqsen.queue_tempo(queue) event = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_PGMCHANGE) event.queue = queue event.time = 96 event.set_data({'control.value' : 127}) seqsen.output_event(event) event2 = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_PGMCHANGE) event2.queue = queue event2.time = 96*4 event2.set_data({'control.value' : 100}) seqsen.output_event(event2) event3 = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_SETPOS_TICK) event3.queue = queue event3.time = 96*5 event3.set_data({'queue.queue' : 1}) seqsen.output_event(event3) seqsen.drain_output() ## event and event2 being sent but event3 is unnoticeable to me...

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[alsa-devel] ALSA:AT91SAM9G20: Add support(DAI) for wolfson8731
by Sedji Gaouaou 17 Sep '08

17 Sep '08

Hi, So here is a patch to add support for the wolfson 8731 on the at91sam9g20 evaluation board. Any comments will be welcomed!! Regards, Sedji sound/soc/at91/Kconfig | 19 +- sound/soc/at91/Makefile | 6 + sound/soc/at91/at91-ssc.h | 1 + sound/soc/at91/atmel_ssc_dai.c | 715 ++++++++++++++++++++++++++++++++++ sound/soc/at91/sam9g20_wm8731.c | 290 ++++++++++++++ sound/soc/codecs/Kconfig | 2 +- 6 files changed, 1031 insertions(+), 2 deletions(-) diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 9051865..86621b9 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -7,7 +7,15 @@ config SND_AT91_SOC to select the audio interfaces to support below. config SND_AT91_SOC_SSC - tristate + tristate "at91 soc ssc" + +config SND_ATMEL_SOC_SSC + tristate "ATMEL SoC ssc dai" + depends on ATMEL_SSC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface(interface between at91sam9g20 and + WM8731 for instance). config SND_AT91_SOC_ETI_B1_WM8731 tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" @@ -18,6 +26,15 @@ config SND_AT91_SOC_ETI_B1_WM8731 Say Y if you want to add support for SoC audio on WM8731-based Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_AT91_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + config SND_AT91_SOC_ETI_SLAVE bool "Run codec in slave Mode on Endrelia boards" depends on SND_AT91_SOC_ETI_B1_WM8731 diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile index f23da17..f12a798 100644 --- a/sound/soc/at91/Makefile +++ b/sound/soc/at91/Makefile @@ -1,11 +1,17 @@ # AT91 Platform Support snd-soc-at91-objs := at91-pcm.o snd-soc-at91-ssc-objs := at91-ssc.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + # AT91 Machine Support snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index 6b7bf38..b967691 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -22,6 +22,7 @@ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ extern struct snd_soc_dai at91_ssc_dai[]; +extern struct snd_soc_dai atmel_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/atmel_ssc_dai.c b/sound/soc/at91/atmel_ssc_dai.c new file mode 100644 index 0000000..a3888c0 --- /dev/null +++ b/sound/soc/at91/atmel_ssc_dai.c @@ -0,0 +1,715 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> +#include <mach/at91_pmc.h> + +#include "at91-pcm.h" +#include "at91-ssc.h" + +#if 0 +#define DBG(x...) printk(KERN_DEBUG "atmel_ssc_dai:" x) +#else +#define DBG(x...) +#endif + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct at91_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct at91_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct at91_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct at91_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + +static struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* 1=SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at91_pcm_dma_params *dma_params[2]; +} ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = 0, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = 0, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = 0, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct at91_pcm_dma_params *dma_params; + u32 ssc_sr; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if (dma_params != NULL && dma_params->dma_intr_handler != NULL && + (ssc_sr & + (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) + + dma_params->dma_intr_handler(ssc_sr, dma_params->substream); + } + + return IRQ_HANDLED; +} + +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + DBG("ssc_startup: SSC_SR=0x%08lx\n", ssc_readl(ssc_p->ssc->regs, SR)); + dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at91_pcm_dma_params *dma_params; + int dir, dir_mask; + u32 pid; + + pid = ssc_p->ssc->pdev->resource[1].start; + + dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc_base = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + DBG("Stopping pid %d clock\n", pid); + at91_sys_write(AT91_PMC_PCDR, 1<<pid); + + if (ssc_p->initialized) { + free_irq(pid, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); + ssc_free(ssc_p->ssc); +} + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT91SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT91SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT91SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct at91_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + u32 pid; + int start_event; + int ret; + + ssc_p->ssc = ssc_request(0); + pid = ssc_p->ssc->pdev->resource[1].start; + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc_base = ssc_p->ssc->regs; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, 4) + | SSC_BF(RCMR_CKI, 1) + | SSC_BF(RCMR_CKO, 0) + | SSC_BF(RCMR_CKS, 0); + + rfmr = SSC_BF(RFMR_FSEDGE, 0) + | SSC_BF(RFMR_FSOS, 1) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BF(RFMR_MSBF, 1) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_CKI, 0) + | SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_CKS, 0); + + tfmr = SSC_BF(TFMR_FSEDGE, 0) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, 1) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BF(TFMR_MSBF, 1) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is generated from the + * transmit clock. + * + * For single channel data, one sample is transferred on the falling + * edge of the LRC clock. For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = channels == 1 + ? 4 + : 7; + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, 1) + | SSC_BF(RCMR_CKO, 0) + | SSC_BF(RCMR_CKS, 1); + + rfmr = SSC_BF(RFMR_FSEDGE, 0) + | SSC_BF(RFMR_FSOS, 0) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_MSBF, 1) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, 0) + | SSC_BF(TCMR_CKO, 0) + | SSC_BF(TCMR_CKS, 2); + + tfmr = SSC_BF(TFMR_FSEDGE, 0) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, 0) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_MSBF, 1) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, 5) + | SSC_BF(RCMR_CKI, 1) + | SSC_BF(RCMR_CKO, 0) + | SSC_BF(RCMR_CKS, 0); + + rfmr = SSC_BF(RFMR_FSEDGE, 0) + | SSC_BF(RFMR_FSOS, 2) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BF(RFMR_MSBF, 1) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, 5) + | SSC_BF(TCMR_CKI, 1) + | SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_CKS, 0); + + tfmr = SSC_BF(TFMR_FSEDGE, 0) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, 2) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BF(TFMR_MSBF, 1) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x.\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + DBG("Starting pid %d clock\n", pid); + at91_sys_write(AT91_PMC_PCER, 1<<pid); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(pid, atmel_ssc_interrupt, + 0, ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING "atmel_ssc_dai: request_irq failure\n"); + + DBG("Stopping pid %d clock\n", pid); + at91_sys_write(AT91_PMC_PCER, 1<<pid); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + DBG("hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at91_pcm_dma_params *dma_params; + int dir; + + dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +#define atmel_ssc_suspend NULL +#define atmel_ssc_resume NULL +#else +#define atmel_ssc_suspend NULL +#define atmel_ssc_resume NULL +#endif + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0].ssc, + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1].ssc, + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2].ssc, + }, +#endif +}; + +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/sam9g20_wm8731.c b/sound/soc/at91/sam9g20_wm8731.c new file mode 100644 index 0000000..8d77830 --- /dev/null +++ b/sound/soc/at91/sam9g20_wm8731.c @@ -0,0 +1,290 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "at91-pcm.h" +#include "at91-ssc.h" + + +#if 0 +#define DBG(x...) printk(KERN_INFO "at91sam9g20ek_wm8731: " x) +#else +#define DBG(x...) +#endif + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + DBG("pck0 stopped\n"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + unsigned int rate; + int cmr_div, period; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + AT91SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + AT91SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + + DBG("at91sam9g20ek_wm8731 : at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &at91_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + int ret; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + DBG("platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + DBG("platform device add failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + platform_device_unregister(at91sam9g20ek_snd_device); +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..1595b04 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_WM8510 tristate config SND_SOC_WM8731 - tristate + tristate "WM8731" config SND_SOC_WM8750 tristate

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[alsa-devel] snd-usb-audio: Roland Fantom-G Support
by Luís Filipe Lobo 17 Sep '08

17 Sep '08

Hi! The Roland Fantom-G is the most recent synth from Roland and is not supported in the current usb-audio version. I am contacting this list to know how can I contribute to get it working. I've added a new entry in usbquirks.h copied from Roland Edirol UA-100 and change the vendor/product id. Without further changes I got midi working perfectly. Now I am having difficulty establishing a relation between the output of 'lsusb -v ...' and the remaining parameters in my usbquirks entry (endpoint, ep_attr, ...). Any help/references would be much apreciated. I also attached the output of 'lsusb -v -d 0582:00de' hoping it will be of any use. Thanks in advance, Regards, -- Luís Filipe Lobo Bus 005 Device 005: ID 0582:00de Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x00de bcdDevice 0.01 iManufacturer 1 Roland iProduct 2 Fantom G iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 8 Mass Storage bDeviceSubClass 6 SCSI bDeviceProtocol 80 Bulk (Zip) bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0001 Self Powered

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Re: [alsa-devel] Unable to play aplay
by Nitin Mahajan 17 Sep '08

17 Sep '08

HI! ----- Original Message ---- > From: vasudha rao <vasudha.jrao(a)gmail.com> > To: aggarwal <anuj.aggarwal(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org > Sent: Tuesday, September 16, 2008 17:08:30 > Subject: Re: [alsa-devel] Unable to play aplay > > I inserted modules to my linux kernel (2.6.20)like.. > > $insmod snd.ko > $insmod snd-timer.ko > $insmod snd-page-alloc.ko > $insmod snd-pcm.ko > $insmod ac97_bus.ko > $insmod snd-ac97-codec.ko > $insmod snd-at91-ac97.ko > > After this, I am getting, > > $aplay > > No error messages.. > > $aplay laughter.wav > Playing WAVE 'laughter.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono > Bad pte = 5fc3e05f, process = aplay, vm_flags = 800fb, vaddr = 4032c000 > Bad pte = 5fc3f05f, process = aplay, vm_flags = 800fb, vaddr = 4032d000 > Unable to handle kernel paging request at virtual address ff800001 > pgd = c346c000 > [ff800001] *pgd=00000000 > Internal error: Oops: 1 [#1] > Modules linked in: snd_at91_ac97 snd_ac97_codec ac97_bus snd_pcm > snd_page_alloc snd_timer snd > CPU: 0 > PC is at snd_pcm_mmap_data_nopage+0xc4/0xe8 [snd_pcm] > LR is at 0x4032e000 > pc : [] lr : [<4032e000>] Not tainted > sp : c3bb3e6c ip : ff7ffffd fp : c3bb3e80 > r10: 4032e000 r9 : c001f380 r8 : c3bb2000 > r7 : c3449764 r6 : 00000000 r5 : c3bb3ea0 r4 : c34a8c00 > r3 : 00000093 r2 : 00000013 r1 : 00006000 r0 : c0aa3800 > Flags: nzcv IRQs off FIQs on Mode SVC_32 Segment user > Control: 5317F > Table: 2346C000 DAC: 00000015 > Process aplay (pid: 916, stack limit = 0xc3bb2250) > Stack: (0xc3bb3e6c to 0xc3bb4000) > 3e60: 00000000 c3e192ec c3bb3ecc c3bb3e84 > c0067c58 > 3e80: bf019030 c346d008 c001f3c0 5fc3f05f 00000000 00000000 c346d008 > 00000000 > 3ea0: 00000002 ffffffeb c3d2b040 c3449764 c001f380 c001f3b4 c3bb3fb0 > 4032e000 > 3ec0: c3bb3f04 c3bb3ed0 c00293a4 c0067b2c 00022b70 00000000 00000017 > ffffffff > 3ee0: c027900c 00000017 c3bb3fb0 4032e000 40388000 400cb5fc c3bb3fac > c3bb3f08 > 3f00: c00295c0 c00292dc c3bb3f2c c0377f20 ffffffe7 c0844123 00000000 > c0023ec4 > 3f20: 400cb5fc c3bb3f40 c3bb3f34 bf018af4 bf018194 c3bb3f5c c3bb3f44 > c00823d0 > 3f40: bf018ad0 c0377f20 00022b70 00000004 c3bb3f84 c3bb3f60 c00826ac > c00823a4 > 3f60: 000014c2 ffffffff c3bb3fac c3bb3f78 c01f7ee0 c00275a8 c3d2b080 > c3d2b154 > 3f80: 0082ca24 ffffffff 40388734 ffffffff 40389734 00000008 00000004 > 00000004 > 3fa0: 00000000 c3bb3fb0 c0023cc8 c0029594 00000000 4032e000 40354000 > 00000100 > 3fc0: 00006800 40389734 00000008 00000004 00000004 40388000 400cb5fc > 00006dcd > 3fe0: 00000100 bea539d0 00000100 4008ce20 20000010 ffffffff 00000000 > 00000029 > Backtrace: > [] (snd_pcm_mmap_data_nopage+0x0/0xe8 [snd_pcm]) from [] > (__handle_mm_fault+0x13c/0x760) > r5 = C3E192EC r4 = 00000000 > [] (__handle_mm_fault+0x0/0x760) from [] > (do_page_fault+0xd8/0x20c) > [] (do_page_fault+0x0/0x20c) from [] > (do_DataAbort+0x3c/0xa0) > [] (do_DataAbort+0x0/0xa0) from [] > (ret_from_exception+0x0/0x10) > r8 = 00000004 r7 = 00000004 r6 = 00000008 r5 = 40389734 > r4 = FFFFFFFF > Code: 1590c00c e10f2000 e3823080 e121f003 (e59c3004) > Segmentation fault > > > $aplay -l > card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > $arecord > ALSA lib pcm_dmix.c:812:(snd_pcm_dmix_open) The dmix plugin supports only > playback stream > arecord: main:446: audio open error: Invalid argument > > $arecord -l > As far as arecord is concerned,check if your codec chip has an ADC, what mixer controls are available for recording? Are u setting them ? > No message here.. > > $cat /proc/asound/pcm > 00-00: Atmel AC97 : Atmel AC97 : playback 1 > > Why I am not able to here or record sound..?? > Where I am doing wrong...?? > > Thank you.. > > > > > > > > > > > > > On Thu, Sep 11, 2008 at 10:46 AM, vasudha rao wrote: > > > Hi anuj , > > > > Sorry my mistake..that time I hav not loaded conf files in /usr/share/ .. I > > loaded now.. > > So now when I give command like, > > > > ------------------------------------------------- > > $aplay -l > > **** List of PLAYBACK Hardware Devices **** > > card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > --------------------------------------------------- > > > > but,aplay not proper,, > > ---------------------------- > > $aplay > > aplay: main:508: audio open error: Inappropriate ioctl for device > > ---------------------------- > > > > Where I am doing wrong..?? > > > > Thank you.. > > > > On Thu, Sep 11, 2008 at 10:07 AM, vasudha rao wrote: > > > >> > >> > >> > >> Hi Anuj, > >> > >> Nodes were not there before..I created now.. > >> > >> Now when I did $aplay -l , > >> I got, > >> > >> **** List of PLAYBACK Hardware Devices *** > >> *ALSA lib control.c:739:(snd_ctl_open_noupdate) Invalid CTL hw:0 > >> aplay: device_list:208: control open (0): No such file or directory > >> aplay: conf.c:3083: snd_config_update_free: Assertion `update->count > 0 > >> && update->finfo' failed.Aborted > >> > >> Where I am doing wrong..?? > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Thank you.. > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Sep 10, 2008 at 11:54 PM, anuj aggarwal > wrote: > >> > >>> Vasudha, > >>> > >>> Could you please check the device nodes in /dev/snd folder? The > >>> desired output is: > >>> > >>> [root@ /home]# ls -l /dev/snd/ > >>> crw-rw---- 1 root root 116, 5 Jan 1 00:00 controlC0 > >>> crw-rw---- 1 root root 116, 4 Jan 1 00:00 pcmC0D0c > >>> crw-rw---- 1 root root 116, 3 Jan 1 00:00 pcmC0D0p > >>> > >>> If the nodes are not there, you have to create them first before trying > >>> aplay. > >>> > >>> Regards, > >>> Anuj > >>> > >>> > >>> On Wed, Sep 10, 2008 at 6:16 PM, vasudha rao > >>> wrote: > >>> > > >>> > Hi , > >>> > > >>> > I want to play and record audio file in my custom atmel 9263 board > >>> running > >>> > with linux(2.6.20). > >>> > when I tried to play aplay for a wave file, I got > >>> > > >>> > $aplay /home/alarm.wav > >>> > ALSA lib pcm.c:1707:(snd_pcm_open_noupdate) Unknown PCM default > >>> > aplay: main:446: audio open error: No such file or directory > >>> > > >>> > $aplay -l > >>> > aplay: device_list:187: no soundcards found... > >>> > > >>> > $cat /proc/asound/devices > >>> > 2: : timer > >>> > 3: : sequencer > >>> > 4: [ 0- 0]: digital audio playback > >>> > 5: [ 0] : control > >>> > > >>> > Can you tel me where I am doing wrong..?? > >>> > > >>> > Thanks in advance.. > >>> > _______________________________________________ > >>> > Alsa-devel mailing list > >>> > Alsa-devel(a)alsa-project.org > >>> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >>> > >>> > >>> > >>> -- > >>> Best Regards, > >>> Anuj Aggarwal > >>> > >> > >> > > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/

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Re: [alsa-devel] [PATCH] ALSA: HDA: hda_local: Less magic numbers.
by Jonathan Woithe 17 Sep '08

17 Sep '08

Robin H. Johnson wrote: > Explain some of the magic numbers I saw while trying to fix the AD1989 SPDIF > issue. Maybe should just use the expanded form directly in the verbs? > > Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> > > diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h > index d688f50..7957fef 100644 > --- a/sound/pci/hda/hda_local.h > +++ b/sound/pci/hda/hda_local.h > @@ -368,12 +368,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, > #define AMP_OUT_UNMUTE 0xb000 > #define AMP_OUT_ZERO 0xb000 > /* pinctl values */ > -#define PIN_IN 0x20 > -#define PIN_VREF80 0x24 > -#define PIN_VREF50 0x21 > -#define PIN_OUT 0x40 > -#define PIN_HP 0xc0 > -#define PIN_HP_AMP 0x80 > +#define PIN_IN (AC_PINCTL_IN_EN) > +#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) > +#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50) > +#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) > +#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80) > +#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) > +#define PIN_OUT (AC_PINCTL_OUT_EN) > +#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) > +#define PIN_HP_AMP (AC_PINCTL_HP_EN) Shrug. In my opinion the above patch simply reworks the explanation, shifting the "magic" from the PIN_* definitions to the AC_PINCTL_* defines. One could take things one step further and replace the PIN_* defines with AC_PINCTL_* references throughout the code, but this would make the code far more verbose and harder to follow. At the end of the day I'm not fussed either way. Having said that, I don't really see the practical advantage of the change but if Takashi et al thing it's a good idea then I wouldn't oppose it. Regards jonathan

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[alsa-devel] [PATCH] alsactl crashes on 1.0.18rc3 alsa-utils
by Colin Guthrie 17 Sep '08

17 Sep '08

Hi, One of my Mandriva colleagues (Gustavo De Nardin) fixed this in the following patch: http://svn.mandriva.com/cgi-bin/viewvc.cgi/packages/cooker/alsa-utils/curre… Please apply if it's not already been fixed. Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] help, please nVidia Corporation MCP73
by lazaro souza 16 Sep '08

16 Sep '08

please help me, I have an equipment with a motherboard gft050vt-m5 with the descriptions below, the problem that springs to mind is with the sound front, because it uses a conexçao c, 97 and another hd audio but only works the front microphone and the sound does not come out, but connecting on the back the sound comes out perfect, how to resolve this? as tested in several router with kernel and alsa different, and the problem is the same, PS in ruindows works installing additional software. I look forward in a contact. lspci | grep Audio 00:09.0 Audio device: nVidia Corporation MCP73 High Definition Audio (rev a1) lspci -n | grep 00:09.0 00:09.0 0403: 10de:07fc (rev a1) cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfebf8000 irq 23 por favor me ajudem, tenho um equipamento com uma placa mae gft050vt-m5 com as descrições abaixo, o problema que me ocorre é com o som frontal, pois a mesma utiliza uma conexçao ac,97 e outra hd audio mas só funciona o microfone frontal e o som não sai, mas conectando na parte traseira o som sai perfeito, como resolver isto ? pois testei em varias distribuiçoes com kernel e alsa diferentes, e o problema é o mesmo, OBS: no ruindows funciona instalando um software adicional. fico no aguardo de um contato. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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[alsa-devel] [PATCH] hda: SPDIF mux fixes for STAC927x
by Matthew Ranostay 16 Sep '08

16 Sep '08

Corrected bounds-checking in stac92xx_auto_create_mux_input_ctls() and added a spec->spdif_labels pointer for custom SPDIF mux labels for non-standard codec connections. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f00334f..dc7393f 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -175,6 +175,7 @@ struct sigmatel_spec { unsigned int num_dmuxes; hda_nid_t *smux_nids; unsigned int num_smuxes; + const char **spdif_labels; hda_nid_t dig_in_nid; hda_nid_t mono_nid; @@ -383,6 +384,11 @@ static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 }; +static const char *stac927x_spdif_labels[5] = { + "Digital Playback", "ADAT", "Analog Mux 1", + "Analog Mux 2", "Analog Mux 3" +}; + static hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; @@ -2893,26 +2899,29 @@ static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) }; static const char *stac92xx_spdif_labels[3] = { - "Digital Playback", "Analog Mux 1", "Analog Mux 2" + "Digital Playback", "Analog Mux 1", "Analog Mux 2", }; static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *spdif_mux = &spec->private_smux; + const char **labels = spec->spdif_labels; int i, num_cons; - hda_nid_t con_lst[ARRAY_SIZE(stac92xx_spdif_labels)]; + hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; num_cons = snd_hda_get_connections(codec, spec->smux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); - if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_spdif_labels)) + if (!num_cons) return -EINVAL; + if (!labels) + labels = stac92xx_spdif_labels; + for (i = 0; i < num_cons; i++) { - spdif_mux->items[spdif_mux->num_items].label = - stac92xx_spdif_labels[i]; + spdif_mux->items[spdif_mux->num_items].label = labels[i]; spdif_mux->items[spdif_mux->num_items].index = i; spdif_mux->num_items++; } @@ -4376,6 +4385,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->smux_nids = stac927x_smux_nids; spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids); + spec->spdif_labels = stac927x_spdif_labels; spec->dac_list = stac927x_dac_nids; spec->multiout.dac_nids = spec->dac_nids;

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[alsa-devel] [PATCH 0/3] ASoC updates
by Mark Brown 16 Sep '08

16 Sep '08

The following changes since commit 78bd2cd2124d3a77f7f78af47849ea571a73f2a1: Dmitry Baryshkov (1): ALSA: Separate common pxa2xx-pcm code are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (3): ASoC: Fix dependency on AC97_BUS ASoC: DAPM support for ADC on WM9713 PCM interface ASoC: Add WM8750 SPI support sound/soc/Kconfig | 1 + sound/soc/codecs/wm8750.c | 70 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8750.h | 1 + sound/soc/codecs/wm9713.c | 12 ++++++- 4 files changed, 81 insertions(+), 3 deletions(-)

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[alsa-devel] [PATCH 0/7] HDA patch_via.c update
by Harald Welte 16 Sep '08

16 Sep '08

Hi! As part of my work for VIA I have stumbled upon their current internal HDA codec driver and it's various differences against the current mainline driver. I have spent the better part of two days separating the individual changesets out of the big diff between VIA's driver and current mainline linux-2.6 (which is identical to alsa-kernel.git). After this merge, I will work with product management and R&D inside VIA that all individual changesets (bugfixes / enhancements) will be sent ASAP for mainline inclusion to prevent any large differencees betwee the two trees. If you have any questions regarding VIA's hardware or need access to data sheets or the like, I am happy to assist. -- - Harald Welte <HaraldWelte(a)viatech.com> http://linux.via.com.tw/ ============================================================================ VIA Open Source Liaison

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[alsa-devel] Dai for at91sam9g20.
by Sedji Gaouaou 16 Sep '08

16 Sep '08

Hi, I have written a dai to use a wolfson 8731 with the sam9g20 atmel board. I would like to know what is the proper place to post my patch, is it here? Regards, Sedji

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[alsa-devel] Unable to play aplay
by vasudha rao 16 Sep '08

16 Sep '08

Hi , I want to play and record audio file in my custom atmel 9263 board running with linux(2.6.20). when I tried to play aplay for a wave file, I got $aplay /home/alarm.wav ALSA lib pcm.c:1707:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:446: audio open error: No such file or directory $aplay -l aplay: device_list:187: no soundcards found... $cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0] : control Can you tel me where I am doing wrong..?? Thanks in advance..

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[alsa-devel] [PATCH 0/3] HDA patch_via.c updates
by Harald Welte 15 Sep '08

15 Sep '08

Hi! Unfortunately there is one crash (headphone related), one non-working feature (second S/PDIF out) and one regular bugfix since my seven-patch series from last week. Please review + apply the following patches to your tree. I've tested them against sound-2.6/topic/hda and they apply fine. If you have any questions, please keep Logan Li in the Cc, he is the engineer who actually implemented those changes and knows all the details. Thanks in advance, -- - Harald Welte <HaraldWelte(a)viatech.com> http://linux.via.com.tw/ ============================================================================ VIA Open Source Liaison

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[alsa-devel] [PATCH 0/6] ASoC updates
by Mark Brown 15 Sep '08

15 Sep '08

The following changes since commit 3e3b5651631629482e0e1541f8c4298a435bb100: Mark Brown (1): ALSA: ASoC: Use codec digital mute when stopping playback are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Alan Horstmann (1): ASoC: Fix WM8731 SPI write Dmitry Baryshkov (4): ASoC: Support runtime selection of CPU in pxa2xx-i2s ALSA: Separate common pxa2xx-ac97 code ALSA: pxa2xx-ac97-lib: support building for several CPUs ALSA: Separate common pxa2xx-pcm code Kenneth Kiraly (1): ASoC: Add WM8971 CODEC driver include/sound/pxa2xx-lib.h | 45 ++ sound/arm/Kconfig | 9 + sound/arm/Makefile | 4 + sound/arm/pxa2xx-ac97-lib.c | 384 ++++++++++++++++++ sound/arm/pxa2xx-ac97.c | 247 +----------- sound/arm/pxa2xx-pcm-lib.c | 278 +++++++++++++ sound/arm/pxa2xx-pcm.c | 252 +----------- sound/arm/pxa2xx-pcm.h | 13 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8731.c | 5 +- sound/soc/codecs/wm8971.c | 942 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8971.h | 64 +++ sound/soc/pxa/Kconfig | 3 + sound/soc/pxa/pxa2xx-ac97.c | 272 +------------ sound/soc/pxa/pxa2xx-i2s.c | 20 +- sound/soc/pxa/pxa2xx-pcm.c | 261 +------------ sound/soc/pxa/pxa2xx-pcm.h | 15 - 18 files changed, 1802 insertions(+), 1018 deletions(-) create mode 100644 include/sound/pxa2xx-lib.h create mode 100644 sound/arm/pxa2xx-ac97-lib.c create mode 100644 sound/arm/pxa2xx-pcm-lib.c create mode 100644 sound/soc/codecs/wm8971.c create mode 100644 sound/soc/codecs/wm8971.h

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[alsa-devel] alsa on Apple Mac Pro Tower ( macPro3, 1, quad core, xenon)
by S. Aguinaga 15 Sep '08

15 Sep '08

I downloaded the latest development drivers, lib, and utils (1.0.18rc3) and compiled and installed it again. Next I made sure that the /etc/modprobe.d/sound had the following line: options snd-hda-intel model=imac24 and first I checked the gui tool called "PulseAudio Volume Control" from my "multimedia" section, and it crashes! Next I tried the aplay somesound.wav file from the command line, it my Mac Pro sings!!! I don't know what the PulseAudio issues are, but ... oh well!! Next I checked that Amrok works using the alsa engine, and it does!! If I plug in a headphone set, and I play music again it comes out of the MacPro speaker only, but no from the headphones. Do you have a handle this, if you guys have some links for me to read, please send them my way! Thank you. // Sal Aguinaga // Northwestern University // Evanston, IL ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, August 19, 2008 5:48:47 AM Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Mon, 18 Aug 2008 10:26:42 -0700 (PDT), S. Aguinaga wrote: > > ============================================== > Trying to figure out where to add this "model=macpro" > and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ > > but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? You can create any file, typically, /etc/modprobe.d/sound or so. Put the following line there options snd-hda-intel model=macpro Then reload the module, check the sysfs file /sys/modules/snd_hda_intel/parameters/macpro to see whether the specified option was really passed properly. Takashi > below are the only files I have. > > /etc/modprobe.d/ > > anaconda > blacklist-visor > i915modeset > blacklist > floppy-pnp > modprobe.conf.dist > > > ============================================== > Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 > > At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), > S. Aguinaga wrote: > > > > S. Aguinaga wrote: > > > > > > Hello Stan, > > > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > > 2. aplay does not produce sound, period! > > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > > > sudo cat /var/log/messages | grep ALSA > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > (snip) > > These messages are harmless. It's just information from the driver, > enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > > > I think your problem is a misidentification of your sound chip. The > > model being used for your 889A appears to be incompatible, thus all > > the errors above. It appears that the macpro ALC885 is being selected > > before it gets to the choice for the ALC889A. Perhaps a developer can > > look into this. > > Well, ALC882, ALC885 and ALC889A are almost compatible. In the > driver, they are all handled in the same routine. > > There are a few models for intel macs with ALC88x. Try to specify > either model=macpro, model=mbp3 or model=imac24 as the module option > of snd-hda-intel driver. > > > Takashi > > > > >

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[alsa-devel] Dev: Alsa sequencer problem (Why ???)
by Gilbux (free) 15 Sep '08

15 Sep '08

Hello, I am trying to write an alsa sequencer. So I read as I can some of your tutorials, and I don't really understand what I am doing wrong, please help me. I want to send scheduled event with the queuing system, but with RTC timer the queue doesn't start, and with the other timer the queue start but no events are sent. I have joined to this email a small code, with the same kind of functions I am using, and some /proc/asound/seq/* outputs. Regards, Gilbux /proc/asound/seq/* before starting the queue : Client info cur clients : 2 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 128 : "TestQSchedEv" [User] Port 0 : "OutputPort" (R-e-) Output pool : Pool size : 500 Cells in use : 16 Peak cells in use : 16 Alloc success : 16 Alloc failures : 0 queue 0: [TQSE_Queue] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 16 timer state : Stopped timer PPQ : 192 current tempo : 500000 current BPM : 120 current time : 0.000000000 s current tick : 0 Timer for queue 0 : RTC timer Period time : 0.000000000 Skew : 65536 / 65536 ----- /proc/asound/seq/* after starting the queue : Client info cur clients : 2 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 128 : "TestQSchedEv" [User] Port 0 : "OutputPort" (R-e-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 16 Alloc success : 16 Alloc failures : 0 queue 0: [TQSE_Queue] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 192 current tempo : 500000 current BPM : 120 current time : 0.000000000 s current tick : 0 Timer for queue 0 : RTC timer Period time : 0.000976562 Skew : 65536 / 65536

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Fix SPDIF output on AD1989B
by robbat2＠gentoo.org 15 Sep '08

15 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The SPDIF pins for AD1989 are not enabled by default. Set OUT bit so that they actually work. Also initialize the HDMI SPDIF at the same time. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b037324..42b1a23 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2255,8 +2255,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { - /* SPDIF out pin */ + /* SPDIF-1 out pin */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + /* SPDIF-2/HDMI out pin */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ { } }; -- 1.6.0.1

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[alsa-devel] [PATCH] hda: 92hd73xx fixes
by Matthew Ranostay 15 Sep '08

15 Sep '08

Added support for defining the number of amps and the creation of the private_dimux dynamically for the 92hd73xx codec family. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f41ba17..fff27cc 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -156,13 +156,11 @@ struct sigmatel_spec { unsigned int num_pwrs; unsigned int *pwr_mapping; hda_nid_t *pwr_nids; - hda_nid_t *amp_nids; hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; - unsigned int cur_amux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -201,6 +199,9 @@ struct sigmatel_spec { unsigned int cur_mux[3]; struct hda_input_mux *sinput_mux; unsigned int cur_smux[2]; + unsigned int cur_amux; + hda_nid_t *amp_nids; + unsigned int num_amps; unsigned int powerdown_adcs; /* i/o switches */ @@ -247,8 +248,9 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd73xx_amp_nids[4] = { - 0x0b, 0x0c, 0x0e, 0 +#define DELL_M6_AMP 2 +static hda_nid_t stac92hd73xx_amp_nids[3] = { + 0x0b, 0x0c, 0x0e }; #define STAC92HD73XX_NUM_DMICS 2 @@ -963,29 +965,33 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; +#define DELL_M6_MIXER 6 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - + /* start of config #1 */ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + + /* start of config #2 */ + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + { } /* end */ }; @@ -2956,17 +2962,19 @@ static int stac92xx_auto_create_amp_input_ctls(struct hda_codec *codec) struct hda_input_mux *amp_mux = &spec->private_amp_mux; int i, err; - for (i = 0; i < ARRAY_SIZE(stac92xx_amp_labels); i++) { + for (i = 0; i < spec->num_amps; i++) { amp_mux->items[amp_mux->num_items].label = stac92xx_amp_labels[i]; amp_mux->items[amp_mux->num_items].index = i; amp_mux->num_items++; } - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, - "Amp Selector Capture Switch", 0); - if (err < 0) - return err; + if (spec->num_amps > 1) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, + "Amp Selector Capture Switch", 0); + if (err < 0) + return err; + } return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, "Amp Capture Volume", HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); @@ -3342,12 +3350,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - if (spec->amp_nids) { + if (spec->num_amps) { err = stac92xx_auto_create_amp_input_ctls(codec); if (err < 0) return err; } - if (spec->num_dmics > 0) + if (spec->num_dmics > 0 && !spec->dinput_mux) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; @@ -3375,8 +3383,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) - spec->dinput_mux = &spec->private_dimux; + spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; spec->amp_mux = &spec->private_amp_mux; @@ -3975,9 +3982,9 @@ static struct hda_input_mux stac92hd73xx_dmux = { .num_items = 4, .items = { { "Analog Inputs", 0x0b }, - { "CD", 0x08 }, { "Digital Mic 1", 0x09 }, { "Digital Mic 2", 0x0a }, + { "CD", 0x08 }, } }; @@ -4048,12 +4055,14 @@ again: spec->dmux_nids = stac92hd73xx_dmux_nids; spec->smux_nids = stac92hd73xx_smux_nids; spec->amp_nids = stac92hd73xx_amp_nids; + spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids); spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); - spec->dinput_mux = &stac92hd73xx_dmux; + memcpy(&spec->private_dimux, &stac92hd73xx_dmux, + sizeof(stac92hd73xx_dmux)); + /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; @@ -4061,11 +4070,18 @@ again: switch (spec->board_config) { case STAC_DELL_M6: spec->init = dell_eq_core_init; + spec->num_smuxes = 0; + spec->multiout.hp_nid = + spec->multiout.dac_nids[spec->multiout.num_dacs - 1]; + spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; + spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; + spec->num_amps = 1; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; + spec->private_dimux.num_items = 1; break; case 0x10280271: /* Digital Mics */ case 0x10280272: @@ -4075,18 +4091,22 @@ again: case 0x10280255: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; case 0x10280256: /* Both */ case 0x10280057: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; } break; default: spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); } + spec->dinput_mux = &spec->private_dimux; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids;

2 3

[alsa-devel] alsa callback not async - Second post to this list
by William Estrada 14 Sep '08

14 Sep '08

Hi guys, I am cross posting this because I am not sure which list it belongs to. I have a test program I wrote to help me learn ALSA APIs, It work to a degree. It is a simple Capture/Playback interface. My problem is that when the Callback starts, it does not allow any cycles back to the main program. The main program waits for console input to tell the callback when to stop. Well it does not accept any console input when the callback is running. Any ideas?? The source is here: http://64.124.13.3/_ALSA_/Echo/ -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] [PATCH take 2] ALSA: HDA: patch_analog: Quirk for Asus P5Q Premium/Pro boards.
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> Use 6STACK_DIG for the AD2000BX variant of the AD1989B chip used by Asus on their Asus P5Q Premium and Pro boards. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 42b1a23..2b00c4a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2884,6 +2884,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} }; -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Quirk for Asus P5Q Premium/Pro boards.
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)orbis-terrarum.net> Use 6STACK_DIG for the AD2000BX variant of the AD1989B chip used by Asus on their Asus P5Q Premium and Pro boards. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 42b1a23..2b00c4a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2884,6 +2884,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} }; -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: hda_local: Less magic numbers.
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> Explain some of the magic numbers I saw while trying to fix the AD1989 SPDIF issue. Maybe should just use the expanded form directly in the verbs? Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d688f50..7957fef 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -368,12 +368,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, #define AMP_OUT_UNMUTE 0xb000 #define AMP_OUT_ZERO 0xb000 /* pinctl values */ -#define PIN_IN 0x20 -#define PIN_VREF80 0x24 -#define PIN_VREF50 0x21 -#define PIN_OUT 0x40 -#define PIN_HP 0xc0 -#define PIN_HP_AMP 0x80 +#define PIN_IN (AC_PINCTL_IN_EN) +#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) +#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50) +#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) +#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80) +#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) +#define PIN_OUT (AC_PINCTL_OUT_EN) +#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) +#define PIN_HP_AMP (AC_PINCTL_HP_EN) /* * get widget capabilities -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Implement multiple outputs for AD1988
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> Implement the slave_dig_outs for AD1989 cards. Untested due to lack of hardware. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0899b6b..b037324 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1826,9 +1826,14 @@ static hda_nid_t ad1988_capsrc_nids[3] = { 0x0c, 0x0d, 0x0e }; -#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 +static hda_nid_t ad1989b_slave_dig_outs[2] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +}; + static struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { @@ -2143,6 +2148,7 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2981,6 +2987,7 @@ static int patch_ad1988(struct hda_codec *codec) ad1989_spdif_out_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1989_spdif_init_verbs; + codec->slave_dig_outs = ad1989b_slave_dig_outs; } else { spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers; -- 1.6.0.1

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[alsa-devel] [PATCH] ALSA: HDA: hda_proc: Fix printf format specifier
by robbat2＠gentoo.org 14 Sep '08

14 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The Pincap output had a typod format specifier, leading to an extraneous "08" in the output, which is a reserved bit of the Vref field, and was really confused :-). Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 4927c4b..743d779 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -216,7 +216,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer, unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - snd_iprintf(buffer, " Pincap 0x08%x:", caps); + snd_iprintf(buffer, " Pincap 0x%08x:", caps); if (caps & AC_PINCAP_IN) snd_iprintf(buffer, " IN"); if (caps & AC_PINCAP_OUT) -- 1.6.0.1

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[alsa-devel] [ALSA hda_codec 1/1] NULL pointer deference fixes for slave_dig_outs
by robbat2＠gentoo.org 13 Sep '08

13 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The new multiple digital output code forgot to check the slave_dig_outs for NULL in several places. This manifests as an OOPS when changing volume amongst other things. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 696d77e..b22061b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1462,14 +1462,15 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); - for (d = codec->slave_dig_outs; *d; d++) { - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_2, - val >> 8); - } + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); + } } mutex_unlock(&codec->spdif_mutex); @@ -1507,10 +1508,12 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) @@ -1664,9 +1667,10 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, val); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -2617,10 +2621,11 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); /* turn on again (if needed) */ @@ -2628,10 +2633,11 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); } } -- 1.6.0.1

2 1

[alsa-devel] [PATCH] Fix oopses in snd-hda-intel from alsa v1.0.18rc3
by Herton Ronaldo Krzesinski 13 Sep '08

13 Sep '08

Fixes the following oopses and other likewise possible ones: BUG: unable to handle kernel NULL pointer dereference at 00000000 IP: [<f8cc5917>] :snd_hda_intel:snd_hda_spdif_out_switch_put+0x97/0x130 *pde = 00000000 Oops: 0000 [#1] SMP Modules linked in: snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device ipv6 snd_pcm_oss snd_mixer_oss coretemp it87 hwmon_vid binfmt_misc loop nls_utf8 ntfs dm_mirror dm_log dm_mod floppy cpufreq_ondemand cpufreq_conservative cpufreq_powersave acpi_cpufreq freq_table snd_hda_intel snd_pcm ub ppdev snd_timer evdev snd_page_alloc r8169 snd_hwdep usbhid snd mii iTCO_wdt button thermal iTCO_vendor_support i2c_i801 processor parport_pc parport serio_raw i2c_core usb_storage intel_agp sr_mod rtc_cmos ff_memless agpgart soundcore sg ata_piix ahci libata dock sd_mod scsi_mod crc_t10dif raid456 async_xor async_memcpy async_tx xor raid0 ext3 jbd uhci_hcd ohci_hcd ehci_hcd usbcore [last unloaded: scsi_wait_scan] Pid: 3730, comm: alsactl Not tainted (2.6.27-desktop-0.rc6.2mnb #1) EIP: 0060:[<f8cc5917>] EFLAGS: 00010246 CPU: 3 EIP is at snd_hda_spdif_out_switch_put+0x97/0x130 [snd_hda_intel] EAX: 00000000 EBX: 00000000 ECX: 00000000 EDX: 00000003 ESI: f7b86200 EDI: 00000001 EBP: f6d8bd94 ESP: f6d8bd68 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process alsactl (pid: 3730, ti=f6d8a000 task=f731d6a0 task.ti=f6d8a000) Stack: 0000070d 00000001 00000000 00000001 00000006 f7b86380 00065a80 00000001 f6c35a80 f69d8c00 f8bd12d0 f6d8bda4 f8bd106a f6c35a80 f69d8c00 f6d8bdb4 f8bd1318 f69d8c00 f711da00 f6d8bf4c f8bcfa59 f6d8bf04 bf893e30 f6cdc8c0 Call Trace: [<f8bd12d0>] ? slave_put+0x0/0x60 [snd] [<f8bd106a>] ? slave_put_val+0x2a/0xa0 [snd] [<f8bd1318>] ? slave_put+0x48/0x60 [snd] [<f8bcfa59>] ? snd_ctl_ioctl+0xc99/0xe70 [snd] [<c01b0317>] ? dput+0x87/0x130 [<c0207e40>] ? ccs_capable+0x30/0x1f0 [<f8bcedc0>] ? snd_ctl_ioctl+0x0/0xe70 [snd] [<c01a928e>] ? vfs_ioctl+0x5e/0xa0 [<c01a4805>] ? putname+0x25/0x40 [<c01a932e>] ? do_vfs_ioctl+0x5e/0x2e0 [<c019a617>] ? do_sys_open+0xc7/0xf0 [<c01a9613>] ? sys_ioctl+0x63/0x70 [<c0103e1b>] ? sysenter_do_call+0x12/0x2f ======================= Code: b7 45 ee 0f b7 c9 0f b6 f9 89 4d e0 31 c9 89 7c 24 04 89 45 e4 89 c2 89 f0 c7 04 24 0d 07 00 00 e8 3f f7 ff ff 8b 9e 9c 01 00 00 <0f> b7 03 66 85 c0 74 23 90 0f b7 d0 31 c9 83 c3 02 89 f0 89 7c EIP: [<f8cc5917>] snd_hda_spdif_out_switch_put+0x97/0x130 [snd_hda_intel] SS:ESP 0068:f6d8bd68 ---[ end trace 1e63c3510a780bec ]--- BUG: unable to handle kernel NULL pointer dereference at 00000000 IP: [<f8cb07d5>] :snd_hda_intel:setup_dig_out_stream+0xf5/0x140 *pde = 00000000 Oops: 0000 [#1] SMP Modules linked in: nfs lockd nfs_acl sunrpc af_packet radeon drm snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device ipv6 snd_pcm_oss snd_mixer_oss coretemp it87 hwmon_vid binfmt_misc loop nls_utf8 ntfs dm_mirror dm_log dm_mod floppy cpufreq_ondemand cpufreq_conservative cpufreq_powersave acpi_cpufreq freq_table ppdev evdev snd_hda_intel thermal snd_pcm snd_timer r8169 snd_page_alloc ub snd_hwdep intel_agp mii snd i2c_i801 button usbhid parport_pc agpgart i2c_core parport serio_raw processor soundcore iTCO_wdt rtc_cmos ff_memless iTCO_vendor_support sr_mod usb_storage sg ata_piix ahci libata dock sd_mod scsi_mod crc_t10dif raid456 async_xor async_memcpy async_tx xor raid0 ext3 jbd uhci_hcd ohci_hcd ehci_hcd usbcore [last unloaded: scsi_wait_scan] Pid: 5139, comm: artsd Not tainted (2.6.27-desktop-0.rc6.2mnb #1) EIP: 0060:[<f8cb07d5>] EFLAGS: 00210202 CPU: 2 EIP is at setup_dig_out_stream+0xf5/0x140 [snd_hda_intel] EAX: 00000000 EBX: 00000000 ECX: 00000000 EDX: 00000003 ESI: f6d54600 EDI: 00000006 EBP: f5afbcd8 ESP: f5afbcc0 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process artsd (pid: 5139, ti=f5afa000 task=f6b50000 task.ti=f5afa000) Stack: 0000070d 00000000 00000005 00000031 f7022050 f6d54600 f5afbd04 f8cb0a8d 00000031 0000e008 00000005 f6d54780 f8ceb39c 00000002 00000031 f6d55500 f5afbd60 f5afbd18 f8cba051 00000031 f7066d00 f7066d00 f5afbd70 f8cae925 Call Trace: [<f8cb0a8d>] ? snd_hda_multi_out_analog_prepare+0x1fd/0x240 [snd_hda_intel] [<f8cba051>] ? alc880_playback_pcm_prepare+0x21/0x30 [snd_hda_intel] [<f8cae925>] ? azx_pcm_prepare+0x345/0x3c0 [snd_hda_intel] [<f8c17d5c>] ? snd_pcm_do_prepare+0xc/0x20 [snd_pcm] [<f8c178ab>] ? snd_pcm_action_single+0x2b/0x60 [snd_pcm] [<f8c18a8b>] ? snd_pcm_action_nonatomic+0x6b/0x70 [snd_pcm] [<f8c1b437>] ? snd_pcm_common_ioctl1+0x607/0x1720 [snd_pcm] [<c0182f6f>] ? handle_mm_fault+0x11f/0x7e0 [<c012a0c0>] ? default_wake_function+0x0/0x10 [<c03820f8>] ? unlock_kernel+0x28/0x2f [<f8bc8688>] ? snd_open+0xd8/0x150 [snd] [<c011cef0>] ? do_page_fault+0x330/0x870 [<c017d7a2>] ? vma_prio_tree_insert+0x22/0xc0 [<c0185f89>] ? vma_link+0xa9/0xf0 [<f8c1c988>] ? snd_pcm_playback_ioctl1+0x38/0x3e0 [snd_pcm] [<c018699e>] ? mmap_region+0x2be/0x4b0 [<c0207e40>] ? ccs_capable+0x30/0x1f0 [<f8c1cd76>] ? snd_pcm_playback_ioctl+0x46/0x50 [snd_pcm] [<f8c1cd30>] ? snd_pcm_playback_ioctl+0x0/0x50 [snd_pcm] [<c01a928e>] ? vfs_ioctl+0x5e/0xa0 [<c01a932e>] ? do_vfs_ioctl+0x5e/0x2e0 [<c01a9613>] ? sys_ioctl+0x63/0x70 [<c0103e1b>] ? sysenter_do_call+0x12/0x2f [<c0380000>] ? schedule+0x760/0x940 ======================= Code: b4 26 00 00 00 00 81 e2 fe 00 00 00 0f b7 f8 89 54 24 04 31 c9 89 fa 89 f0 c7 04 24 0d 07 00 00 e8 91 eb ff ff 8b 9e 9c 01 00 00 <0f> b7 03 66 85 c0 0f 84 2e ff ff ff 8d b4 26 00 00 00 00 0f b7 EIP: [<f8cb07d5>] setup_dig_out_stream+0xf5/0x140 [snd_hda_intel] SS:ESP 0068:f5afbcc0 ---[ end trace d44a3a67f9b19615 ]--- --- Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 696d77e..531364d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1462,14 +1462,15 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); - for (d = codec->slave_dig_outs; *d; d++) { - snd_hda_codec_write_cache(codec, *d, 0, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - snd_hda_codec_write_cache(codec, *d, 0, + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); - } + } } mutex_unlock(&codec->spdif_mutex); @@ -1507,8 +1508,9 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); /* unmute amp switch (if any) */ @@ -1664,8 +1666,9 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); @@ -2617,9 +2620,10 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -2628,9 +2632,10 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); } -- []'s Herton

2 2

[alsa-devel] [PATCH] hda: fixed hp_nid DAC for DELL_M6
by Matthew Ranostay 13 Sep '08

13 Sep '08

This patch sets the HP out not used by the "Headphone to Line Out" switch to the hp_nid. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f41ba17..0218aa7 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -747,8 +747,8 @@ static struct hda_verb dell_m6_core_init[] = { { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, /* setup audio connections */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},

2 1

[alsa-devel] [PATCH] hda: Added a STAC_HP_M4 quirk
by Matthew Ranostay 13 Sep '08

13 Sep '08

Added a new quirk for STAC_HP_M4 series for an 92hd75xxx family laptop. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f00334f..3000ecf 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -80,6 +80,7 @@ enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, + STAC_HP_M4, STAC_92HD71BXX_MODELS }; @@ -1527,12 +1528,14 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, + [STAC_HP_M4] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", + [STAC_HP_M4] = "hp-m4", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -4204,10 +4207,24 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + switch (spec->board_config) { + case STAC_HP_M4: + spec->num_dmics = 0; + spec->num_smuxes = 1; + spec->num_dmuxes = 0; + + /* enable internal microphone */ + snd_hda_codec_write_cache(codec, 0x0e, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + break; + default: + spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + }; + spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids;

2 1

[alsa-devel] [PATCH] hda: Input port AMP controls
by Matthew Ranostay 13 Sep '08

13 Sep '08

Added support for controlling hardware gain amps on input ports using a volume control mixer with a mux selecting the port being controlled. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index 696d77e..e12626b 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -961,15 +961,6 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } #endif /* SND_HDA_NEEDS_RESUME */ -/* - * AMP control callbacks - */ -/* retrieve parameters from private_value */ -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) -#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) -#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) -#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) - /* volume */ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) diff --git a/pci/hda/hda_local.h b/pci/hda/hda_local.h index 5c9e578..d688f50 100644 --- a/pci/hda/hda_local.h +++ b/pci/hda/hda_local.h @@ -418,4 +418,13 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, hda_nid_t nid); #endif /* CONFIG_SND_HDA_POWER_SAVE */ +/* + * AMP control callbacks + */ +/* retrieve parameters from private_value */ +#define get_amp_nid(kc) ((kc)->private_value & 0xffff) +#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) +#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) +#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f00334f..ac56481 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -156,10 +156,13 @@ struct sigmatel_spec { unsigned int num_pwrs; unsigned int *pwr_mapping; hda_nid_t *pwr_nids; + hda_nid_t *amp_nids; hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; + struct hda_input_mux *amp_mux; + unsigned int cur_amux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -215,6 +218,7 @@ struct sigmatel_spec { struct hda_input_mux private_dimux; struct hda_input_mux private_imux; struct hda_input_mux private_smux; + struct hda_input_mux private_amp_mux; struct hda_input_mux private_mono_mux; }; @@ -243,6 +247,10 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; +static hda_nid_t stac92hd73xx_amp_nids[4] = { + 0x0b, 0x0c, 0x0e, 0 +}; + #define STAC92HD73XX_NUM_DMICS 2 static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -448,6 +456,34 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x21, 0x22, }; +#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info + +static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->amp_nids[spec->cur_amux]; + + kcontrol->private_value ^= get_amp_nid(kcontrol); + kcontrol->private_value |= nid; + + return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); +} + +static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->amp_nids[spec->cur_amux]; + + kcontrol->private_value ^= get_amp_nid(kcontrol); + kcontrol->private_value |= nid; + + return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); +} + static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -563,6 +599,41 @@ static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, spec->mono_nid, &spec->cur_mmux); } +static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->amp_mux, uinfo); +} + +static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_amux; + return 0; +} + +static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + struct snd_kcontrol *ctl = + snd_hda_find_mixer_ctl(codec, "Amp Capture Volume"); + if (!ctl) + return -EINVAL; + + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + + return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol, + 0, &spec->cur_amux); +} + #define stac92xx_aloopback_info snd_ctl_boolean_mono_info static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, @@ -837,6 +908,31 @@ static struct hda_verb stac9205_core_init[] = { .put = stac92xx_mono_mux_enum_put, \ } +#define STAC_AMP_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Amp Selector Capture Switch", \ + .count = 1, \ + .info = stac92xx_amp_mux_enum_info, \ + .get = stac92xx_amp_mux_enum_get, \ + .put = stac92xx_amp_mux_enum_put, \ + } + +#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .info = stac92xx_amp_volume_info, \ + .get = stac92xx_amp_volume_get, \ + .put = stac92xx_amp_volume_put, \ + .tlv = { .c = snd_hda_mixer_amp_tlv }, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -2418,6 +2514,8 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_AMP_MUX, + STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH @@ -2427,6 +2525,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_AMP_MUX, + STAC_AMP_VOL(NULL, 0, 0, 0, 0), STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), @@ -2844,6 +2944,35 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) "Mono Mux", spec->mono_nid); } +/* labels for amp mux outputs */ +static const char *stac92xx_amp_labels[3] = { + "Front Microphone", "Microphone", "Line In" +}; + +/* create amp out controls mux on capable codecs */ +static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *amp_mux = &spec->private_amp_mux; + int i, err; + + for (i = 0; i < ARRAY_SIZE(stac92xx_amp_labels); i++) { + amp_mux->items[amp_mux->num_items].label = + stac92xx_amp_labels[i]; + amp_mux->items[amp_mux->num_items].index = i; + amp_mux->num_items++; + } + + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, + "Amp Selector Capture Switch", 0); + if (err < 0) + return err; + return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, + "Amp Capture Volume", + HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); +} + + /* create PC beep volume controls */ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, hda_nid_t nid) @@ -3213,7 +3342,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - + if (spec->amp_nids) { + err = stac92xx_auto_create_amp_output_ctls(codec); + if (err < 0) + return err; + } if (spec->num_dmics > 0) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) @@ -3246,7 +3379,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; - + spec->amp_mux = &spec->private_amp_mux; return 1; } @@ -3914,6 +4047,7 @@ again: spec->dmic_nids = stac92hd73xx_dmic_nids; spec->dmux_nids = stac92hd73xx_dmux_nids; spec->smux_nids = stac92hd73xx_smux_nids; + spec->amp_nids = stac92hd73xx_amp_nids; spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);

2 1

[alsa-devel] bluetooth headset doesn't work well(Motorola HS820)
by Galaha Shine 13 Sep '08

13 Sep '08

Hi, I have a blue-tooth headset connected to Ubuntu 8.04 but it does't work well. I've created ~/.asoundrc as below: pcm.bluetooth { type bluetooth device 20:04:07:80:BC:89 profile voice } ctl.bluetooth { type bluetooth } now I can use command `aplay -Dbluetooth -v 2.wav` to play sound to the headset, but the sound is little bit wired due to inaccurate rate. Playing WAVE '2.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono Warning: rate is not accurate (requested = 16000Hz, got = 8000Hz) please, try the plug plugin (-Dplug:bluetooth) Bluetooth Audio Device Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 So I input command according to the aplay's recommendation `aplay -Dplug:bluetooth -v 2.wav`. The problem comes out. aplay -Dplug:bluetooth -v 2.wav Playing WAVE '2.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 8000 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Bluetooth Audio Device Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 It seems that ALSA plugs rate plugin. But I hear NOTHING from my headset and aplay seems suspended. I turn on bluez debug flag and rebuild it. I find the bluez suspending at 'bluetooth_playback_poll_descriptors'. Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 930 bluetooth_hsp_write: count=0 frames_to_read=24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 993 bluetooth_hsp_write: returning 24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 912 bluetooth_hsp_write: areas->step=16 areas->first=0 offset=0, size=1000 io->nonblock=0 io->hw_ptr=0 io->appl_ptr=0 stream.fd=8 link_mtu=48 data->count=0 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 930 bluetooth_hsp_write: count=0 frames_to_read=24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 993 bluetooth_hsp_write: returning 24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 802 bluetooth_playback_poll_descriptors: Space =2 pfd[0].fd=5 pfd[1].fd=6 Sep 7 23:19:01 alex-laptop audio[7979]: Received AT+VGS=11 Now I go deep into ALSA then I find ALSA suspending at: int snd_pcm_wait_nocheck(snd_pcm_t *pcm, int timeout) { struct pollfd *pfd; unsigned short *revents; int i, npfds, pollio, err, err_poll; npfds = snd_pcm_poll_descriptors_count(pcm); if (npfds <= 0 || npfds >= 16) { SNDERR("Invalid poll_fds %d\n", npfds); return -EIO; } pfd = alloca(sizeof(*pfd) * npfds); revents = alloca(sizeof(*revents) * npfds); syslog( LOG_INFO, "%s", __FUNCTION__ ); err = snd_pcm_poll_descriptors(pcm, pfd, npfds); if (err < 0) return err; if (err != npfds) { SNDMSG("invalid poll descriptors %d\n", err); return -EIO; } do { pollio = 0; err_poll = poll(pfd, npfds, timeout); <------------- It seems the bluez doesn't feed ALSA some signals. But why? I notice 'bluetooth_playback_start' haven't been invoked. Is it the point? Then I keep tracing ALSA. I find in snd_pcm_rate_commit_area result = snd_pcm_mmap_commit(rate->gen.slave, slave_offset, slave_size); if (result < (snd_pcm_sframes_t)slave_size) { if (result < 0) return result; result = snd_pcm_rewind(rate->gen.slave, result); <----------- if (result < 0) return result; return 0; } The snd_pcm_mmap_commit returns with 24 which indicates bluetooth SCO link mtu(48bytes). And it causes snd_pcm_rewind being called after every package sent by bluetooth SCO link. Look at this piece of code in snd_pcm_write_areas: if (state == SND_PCM_STATE_PREPARED) { snd_pcm_sframes_t hw_avail = pcm->buffer_size - avail; hw_avail += frames; /* some plugins might automatically start the stream */ state = snd_pcm_state(pcm); if (state == SND_PCM_STATE_PREPARED && hw_avail >= (snd_pcm_sframes_t) pcm->start_threshold) { err = snd_pcm_start(pcm); if (err < 0) goto _end; } } the 'avail' will be equal to pcm->buffer due to snd_pcm_rewind, therefore the condition hw_avail>=pcm->start_threshold will not be satisfied forever. That's why snd_pcm_start can't be called. I don't know the mechanism of ALSA and I just assume ALSA wanna bluez plugin sending whole period_size instead of only one 48bytes package. I revise the bluez plugin: #if 0 // this section is original bluez frame_size = areas->step / 8; if ((data->count + size * frame_size) <= data->link_mtu) frames_to_read = size; else frames_to_read = (data->link_mtu - data->count) / frame_size; DBG("count=%d frames_to_read=%lu", data->count, frames_to_read); /* Ready for more data */ buff = (uint8_t *) areas->addr + (areas->first + areas->step * offset) / 8; memcpy(data->buffer + data->count, buff, frame_size * frames_to_read); /* Remember we have some frames in the pipe now */ data->count += frames_to_read * frame_size; if (data->count != data->link_mtu) { ret = frames_to_read; goto done; } rsend = send(data->stream.fd, data->buffer, data->link_mtu, io->nonblock ? MSG_DONTWAIT : 0); if (rsend > 0) { /* Reset count pointer */ data->count = 0; ret = frames_to_read; } else if (rsend < 0) ret = (errno == EPIPE) ? -EIO : -errno; else ret = -EIO; #else frame_size = areas->step / 8; buff = (uint8_t *) areas->addr + (areas->first + areas->step * offset) / 8; tobesent = frame_size*size; while( rsend<tobesent ) { if( tobesent-rsend>data->link_mtu ) sent = send(data->stream.fd, buff +counter*data->link_mtu, data->link_mtu, io->nonblock ? MSG_DONTWAIT : 0); else sent = send(data->stream.fd, buff +counter*data->link_mtu, tobesent-rsend, io->nonblock ? MSG_DONTWAIT : 0); if( sent > 0 ) { if( sent != data->link_mtu && tobesent-rsend > data->link_mtu ) DBG("got problem!"); counter++; rsend+=sent; } else if (rsend < 0) { ret = (errno == EPIPE) ? -EIO : -errno; break; } else { ret = -EIO; break; } } ret = size; #endif After this revision, the headset works but the performance isn't as good as playing MONO 8000Hz wav directly to headset with 'aplay -Dbluetooth 1.wav'. Any hints would be appreciated. Regards, Galaha

1 0

Re: [alsa-devel] [PATCH] ASoC: OMAP: Fix Gumstix Overo machine driver by using snd_soc_dai struct (rev 2)
by Steve Sakoman 13 Sep '08

13 Sep '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> Structs snd_soc_codec_dai and snd_soc_cpu_dai are obsolete and are merged to snd_soc_dai by the commit 8687eb8bded8c3c5842a85bd0c30e43fc5a3e0e0. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Acked-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/overo.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index 542ff6b..1a086ce 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -39,8 +39,8 @@ static int overo_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ -- 1.5.6.5

2 1

Re: [alsa-devel] [PATCH] ASoC: OMAP: Fix Gumstix Overo machine driver by using snd_soc_dai struct
by Steve Sakoman 12 Sep '08

12 Sep '08

On Fri, Sep 12, 2008 at 1:46 PM, Felipe Balbi <me(a)felipebalbi.com> wrote: > should this be From: Jarkko ??? Indeed. Will resubmit. Steve

1 0

[alsa-devel] [PATCH] ASoC: OMAP: Fix Gumstix Overo machine driver by using snd_soc_dai struct
by Steve Sakoman 12 Sep '08

12 Sep '08

Structs snd_soc_codec_dai and snd_soc_cpu_dai are obsolete and are merged to snd_soc_dai by the commit 8687eb8bded8c3c5842a85bd0c30e43fc5a3e0e0. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Acked-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/overo.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index 542ff6b..1a086ce 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -39,8 +39,8 @@ static int overo_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ -- 1.5.6.5

1 0

[alsa-devel] alsa callback not async
by William Estrada 12 Sep '08

12 Sep '08

Hi guys, I am cross posting this because I am not sure which list it belongs to. I have a test program I wrote to help me learn ALSA APIs, It work to a degree. It is a simple Capture/Playback interface. My problem is that when the Callback starts, it does not allow any cycles back to the main program. The main program waits for console input to tell the callback when to stop. Well it does not accept any console input when the callback is running. Any ideas?? The source is here: http://64.124.13.3/_ALSA_/Echo/ -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] [PATCH 1/4] Separate common pxa2xx-ac97 code
by Dmitry Baryshkov 12 Sep '08

12 Sep '08

ASoC and non-ASoC drivers for ACLINK on PXA share lot's of common code. Move all common code into separate module snd-pxa2xx-lib. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- include/sound/pxa2xx-lib.h | 20 +++ sound/arm/Kconfig | 6 +- sound/arm/Makefile | 3 + sound/arm/pxa2xx-ac97-lib.c | 323 +++++++++++++++++++++++++++++++++++++++++++ sound/arm/pxa2xx-ac97.c | 247 +++------------------------------ sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/pxa2xx-ac97.c | 272 ++----------------------------------- 7 files changed, 380 insertions(+), 493 deletions(-) create mode 100644 include/sound/pxa2xx-lib.h create mode 100644 sound/arm/pxa2xx-ac97-lib.c diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h new file mode 100644 index 0000000..d18dd2d --- /dev/null +++ b/include/sound/pxa2xx-lib.h @@ -0,0 +1,20 @@ +#ifndef PXA2XX_LIB_H +#define PXA2XX_LIB_H + +#include <linux/platform_device.h> +#include <sound/ac97_codec.h> + +extern unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg); +extern void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val); + +extern bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97); +extern bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97); +extern void pxa2xx_ac97_finish_reset(struct snd_ac97 *ac97); + +extern int pxa2xx_ac97_hw_suspend(void); +extern int pxa2xx_ac97_hw_resume(void); + +extern int pxa2xx_ac97_hw_probe(struct platform_device *dev); +extern void pxa2xx_ac97_hw_remove(struct platform_device *dev); + +#endif diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 351e19e..3fd2642 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -32,11 +32,15 @@ config SND_PXA2XX_PCM tristate select SND_PCM +config SND_PXA2XX_LIB + tristate + select SND_AC97_CODEC + config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA select SND_PXA2XX_PCM - select SND_AC97_CODEC + select SND_PXA2XX_LIB help Say Y or M if you want to support any AC97 codec attached to the PXA2xx AC97 interface. diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 4ef6dd0..bb2ed88 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -11,5 +11,8 @@ snd-aaci-objs := aaci.o devdma.o obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o snd-pxa2xx-pcm-objs := pxa2xx-pcm.o +obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o +snd-pxa2xx-lib-objs := pxa2xx-ac97-lib.o + obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o snd-pxa2xx-ac97-objs := pxa2xx-ac97.o diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c new file mode 100644 index 0000000..61cf41f --- /dev/null +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -0,0 +1,323 @@ +/* + * Based on sound/arm/pxa2xx-ac97.c and sound/soc/pxa/pxa2xx-ac97.c which contain: + * + * Author: Nicolas Pitre + * Created: Dec 02, 2004 + * Copyright: MontaVista Software Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/clk.h> +#include <linux/delay.h> + +#include <sound/ac97_codec.h> +#include <sound/pxa2xx-lib.h> + +#include <asm/irq.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/pxa2xx-gpio.h> +#include <mach/audio.h> + +static DEFINE_MUTEX(car_mutex); +static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); +static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif + +/* + * Beware PXA27x bugs: + * + * o Slot 12 read from modem space will hang controller. + * o CDONE, SDONE interrupt fails after any slot 12 IO. + * + * We therefore have an hybrid approach for waiting on SDONE (interrupt or + * 1 jiffy timeout if interrupt never comes). + */ + +unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + unsigned short val = -1; + volatile u32 *reg_addr; + + mutex_lock(&car_mutex); + + /* set up primary or secondary codec space */ +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) + reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; +#else + if (reg == AC97_GPIO_STATUS) + reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; + else + reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; +#endif + reg_addr += (reg >> 1); + + /* start read access across the ac97 link */ + GSR = GSR_CDONE | GSR_SDONE; + gsr_bits = 0; + val = *reg_addr; + if (reg == AC97_GPIO_STATUS) + goto out; + if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1) <= 0 && + !((GSR | gsr_bits) & GSR_SDONE)) { + printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n", + __func__, reg, GSR | gsr_bits); + val = -1; + goto out; + } + + /* valid data now */ + GSR = GSR_CDONE | GSR_SDONE; + gsr_bits = 0; + val = *reg_addr; + /* but we've just started another cycle... */ + wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); + +out: mutex_unlock(&car_mutex); + return val; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_read); + +void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + volatile u32 *reg_addr; + + mutex_lock(&car_mutex); + + /* set up primary or secondary codec space */ +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) + reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; +#else + if (reg == AC97_GPIO_STATUS) + reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; + else + reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; +#endif + reg_addr += (reg >> 1); + + GSR = GSR_CDONE | GSR_SDONE; + gsr_bits = 0; + *reg_addr = val; + if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1) <= 0 && + !((GSR | gsr_bits) & GSR_CDONE)) + printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", + __func__, reg, GSR | gsr_bits); + + mutex_unlock(&car_mutex); +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_write); + +bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) +{ +#ifdef CONFIG_PXA3xx + int timeout = 100; +#endif + gsr_bits = 0; + +#ifdef CONFIG_PXA27x + /* warm reset broken on Bulverde, + so manually keep AC97 reset high */ + pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + udelay(10); + GCR |= GCR_WARM_RST; + pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + udelay(500); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts */ + GCR |= GCR_WARM_RST; + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); +#else + GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; + wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); +#endif + + if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", + __func__, gsr_bits); + + return false; + } + + return true; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); + +bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) +{ +#ifdef CONFIG_PXA3xx + int timeout = 1000; + + /* Hold CLKBPB for 100us */ + GCR = 0; + GCR = GCR_CLKBPB; + udelay(100); + GCR = 0; +#endif + + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ + GCR &= ~GCR_COLD_RST; /* then assert nCRST */ + + gsr_bits = 0; +#ifdef CONFIG_PXA27x + /* PXA27x Developers Manual section 13.5.2.2.1 */ + clk_enable(ac97conf_clk); + udelay(5); + clk_disable(ac97conf_clk); + GCR = GCR_COLD_RST; + udelay(50); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts on PXA3xx */ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + + GCR = GCR_WARM_RST | GCR_COLD_RST; + while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(10); +#else + GCR = GCR_COLD_RST; + GCR |= GCR_CDONE_IE|GCR_SDONE_IE; + wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); +#endif + + if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", + __func__, gsr_bits); + + return false; + } + + return true; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_cold_reset); + + +void pxa2xx_ac97_finish_reset(struct snd_ac97 *ac97) +{ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + GCR |= GCR_SDONE_IE|GCR_CDONE_IE; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_finish_reset); + +static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) +{ + long status; + + status = GSR; + if (status) { + GSR = status; + gsr_bits |= status; + wake_up(&gsr_wq); + +#ifdef CONFIG_PXA27x + /* Although we don't use those we still need to clear them + since they tend to spuriously trigger when MMC is used + (hardware bug? go figure)... */ + MISR = MISR_EOC; + PISR = PISR_EOC; + MCSR = MCSR_EOC; +#endif + + return IRQ_HANDLED; + } + + return IRQ_NONE; +} + +#ifdef CONFIG_PM +int pxa2xx_ac97_hw_suspend(void) +{ + GCR |= GCR_ACLINK_OFF; + clk_disable(ac97_clk); + return 0; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); + +int pxa2xx_ac97_hw_resume(void) +{ + pxa_gpio_mode(GPIO31_SYNC_AC97_MD); + pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); + pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); + pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); +#ifdef CONFIG_PXA27x + /* Use GPIO 113 as AC97 Reset on Bulverde */ + pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); +#endif + clk_enable(ac97_clk); + return 0; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); +#endif + +int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) +{ + int ret; + + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); + if (ret < 0) + goto err; + + pxa_gpio_mode(GPIO31_SYNC_AC97_MD); + pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); + pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); + pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); +#ifdef CONFIG_PXA27x + /* Use GPIO 113 as AC97 Reset on Bulverde */ + pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err_irq; + } +#endif + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err_irq; + } + + return clk_enable(ac97_clk); + +err_irq: + GCR |= GCR_ACLINK_OFF; +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; + } +#endif + free_irq(IRQ_AC97, NULL); +err: + return ret; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe); + +void pxa2xx_ac97_hw_remove(struct platform_device *dev) +{ + GCR |= GCR_ACLINK_OFF; + free_irq(IRQ_AC97, NULL); +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +} +EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_remove); + +MODULE_AUTHOR("Nicolas Pitre"); +MODULE_DESCRIPTION("Intel/Marvell PXA sound library"); +MODULE_LICENSE("GPL"); + diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 199cca3..cba71d8 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -12,198 +12,27 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/kernel.h> #include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/wait.h> -#include <linux/clk.h> -#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> +#include <sound/pxa2xx-lib.h> -#include <asm/irq.h> -#include <linux/mutex.h> #include <mach/hardware.h> #include <mach/pxa-regs.h> -#include <mach/pxa2xx-gpio.h> #include <mach/audio.h> #include "pxa2xx-pcm.h" - -static DEFINE_MUTEX(car_mutex); -static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); -static volatile long gsr_bits; -static struct clk *ac97_clk; -#ifdef CONFIG_PXA27x -static struct clk *ac97conf_clk; -#endif - -/* - * Beware PXA27x bugs: - * - * o Slot 12 read from modem space will hang controller. - * o CDONE, SDONE interrupt fails after any slot 12 IO. - * - * We therefore have an hybrid approach for waiting on SDONE (interrupt or - * 1 jiffy timeout if interrupt never comes). - */ - -static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) -{ - unsigned short val = -1; - volatile u32 *reg_addr; - - mutex_lock(&car_mutex); - - /* set up primary or secondary codec space */ - reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; - reg_addr += (reg >> 1); - - /* start read access across the ac97 link */ - GSR = GSR_CDONE | GSR_SDONE; - gsr_bits = 0; - val = *reg_addr; - if (reg == AC97_GPIO_STATUS) - goto out; - if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1) <= 0 && - !((GSR | gsr_bits) & GSR_SDONE)) { - printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n", - __func__, reg, GSR | gsr_bits); - val = -1; - goto out; - } - - /* valid data now */ - GSR = GSR_CDONE | GSR_SDONE; - gsr_bits = 0; - val = *reg_addr; - /* but we've just started another cycle... */ - wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); - -out: mutex_unlock(&car_mutex); - return val; -} - -static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) -{ - volatile u32 *reg_addr; - - mutex_lock(&car_mutex); - - /* set up primary or secondary codec space */ - reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; - reg_addr += (reg >> 1); - - GSR = GSR_CDONE | GSR_SDONE; - gsr_bits = 0; - *reg_addr = val; - if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1) <= 0 && - !((GSR | gsr_bits) & GSR_CDONE)) - printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", - __func__, reg, GSR | gsr_bits); - - mutex_unlock(&car_mutex); -} - static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) { - /* First, try cold reset */ -#ifdef CONFIG_PXA3xx - int timeout; - - /* Hold CLKBPB for 100us */ - GCR = 0; - GCR = GCR_CLKBPB; - udelay(100); - GCR = 0; -#endif - - GCR &= GCR_COLD_RST; /* clear everything but nCRST */ - GCR &= ~GCR_COLD_RST; /* then assert nCRST */ - - gsr_bits = 0; -#ifdef CONFIG_PXA27x - /* PXA27x Developers Manual section 13.5.2.2.1 */ - clk_enable(ac97conf_clk); - udelay(5); - clk_disable(ac97conf_clk); - GCR = GCR_COLD_RST; - udelay(50); -#elif defined(CONFIG_PXA3xx) - timeout = 1000; - /* Can't use interrupts on PXA3xx */ - GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); - - GCR = GCR_WARM_RST | GCR_COLD_RST; - while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(10); -#else - GCR = GCR_COLD_RST; - GCR |= GCR_CDONE_IE|GCR_SDONE_IE; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); -#endif - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { - printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); - - /* let's try warm reset */ - gsr_bits = 0; -#ifdef CONFIG_PXA27x - /* warm reset broken on Bulverde, - so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); - udelay(10); - GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); - udelay(500); -#elif defined(CONFIG_PXA3xx) - timeout = 100; - /* Can't use interrupts */ - GCR |= GCR_WARM_RST; - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); -#else - GCR |= GCR_WARM_RST|GCR_PRIRDY_IEN|GCR_SECRDY_IEN; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); -#endif - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) - printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); - } - - GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); - GCR |= GCR_SDONE_IE|GCR_CDONE_IE; -} - -static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) -{ - long status; - - status = GSR; - if (status) { - GSR = status; - gsr_bits |= status; - wake_up(&gsr_wq); - -#ifdef CONFIG_PXA27x - /* Although we don't use those we still need to clear them - since they tend to spuriously trigger when MMC is used - (hardware bug? go figure)... */ - MISR = MISR_EOC; - PISR = PISR_EOC; - MCSR = MCSR_EOC; -#endif - - return IRQ_HANDLED; + if (!pxa2xx_ac97_try_cold_reset(ac97)) { + pxa2xx_ac97_try_warm_reset(ac97); } - return IRQ_NONE; + pxa2xx_ac97_finish_reset(ac97); } static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { @@ -288,17 +117,19 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) snd_ac97_suspend(pxa2xx_ac97_ac97); if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); - GCR |= GCR_ACLINK_OFF; - clk_disable(ac97_clk); - return 0; + return pxa2xx_ac97_hw_suspend(); } static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; + int rc; + + rc = pxa2xx_ac97_hw_resume(); + if (rc) + return rc; - clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -354,40 +185,17 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) if (ret) goto err; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); - if (ret < 0) - goto err; - - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); - pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); - pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); - pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); -#ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); - ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); - if (IS_ERR(ac97conf_clk)) { - ret = PTR_ERR(ac97conf_clk); - ac97conf_clk = NULL; - goto err; - } -#endif - - ac97_clk = clk_get(&dev->dev, "AC97CLK"); - if (IS_ERR(ac97_clk)) { - ret = PTR_ERR(ac97_clk); - ac97_clk = NULL; + ret = pxa2xx_ac97_hw_probe(dev); + if (ret) goto err; - } - clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) - goto err; + goto err_remove; memset(&ac97_template, 0, sizeof(ac97_template)); ret = snd_ac97_mixer(ac97_bus, &ac97_template, &pxa2xx_ac97_ac97); if (ret) - goto err; + goto err_remove; snprintf(card->shortname, sizeof(card->shortname), "%s", snd_ac97_get_short_name(pxa2xx_ac97_ac97)); @@ -401,22 +209,11 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) return 0; } - err: +err_remove: + pxa2xx_ac97_hw_remove(dev); +err: if (card) snd_card_free(card); - if (ac97_clk) { - GCR |= GCR_ACLINK_OFF; - free_irq(IRQ_AC97, NULL); - clk_disable(ac97_clk); - clk_put(ac97_clk); - ac97_clk = NULL; - } -#ifdef CONFIG_PXA27x - if (ac97conf_clk) { - clk_put(ac97conf_clk); - ac97conf_clk = NULL; - } -#endif return ret; } @@ -427,15 +224,7 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) if (card) { snd_card_free(card); platform_set_drvdata(dev, NULL); - GCR |= GCR_ACLINK_OFF; - free_irq(IRQ_AC97, NULL); - clk_disable(ac97_clk); - clk_put(ac97_clk); - ac97_clk = NULL; -#ifdef CONFIG_PXA27x - clk_put(ac97conf_clk); - ac97conf_clk = NULL; -#endif + pxa2xx_ac97_hw_remove(dev); } return 0; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 9212c37..d1ccbdc 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -13,6 +13,8 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_ARM + select SND_PXA2XX_LIB select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index d94a495..a80ae07 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -13,225 +13,30 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/wait.h> -#include <linux/clk.h> -#include <linux/delay.h> #include <sound/core.h> -#include <sound/pcm.h> #include <sound/ac97_codec.h> -#include <sound/initval.h> #include <sound/soc.h> +#include <sound/pxa2xx-lib.h> -#include <asm/irq.h> -#include <linux/mutex.h> #include <mach/hardware.h> #include <mach/pxa-regs.h> -#include <mach/pxa2xx-gpio.h> -#include <mach/audio.h> #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static DEFINE_MUTEX(car_mutex); -static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); -static volatile long gsr_bits; -static struct clk *ac97_clk; -#ifdef CONFIG_PXA27x -static struct clk *ac97conf_clk; -#endif - -/* - * Beware PXA27x bugs: - * - * o Slot 12 read from modem space will hang controller. - * o CDONE, SDONE interrupt fails after any slot 12 IO. - * - * We therefore have an hybrid approach for waiting on SDONE (interrupt or - * 1 jiffy timeout if interrupt never comes). - */ - -static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - unsigned short val = -1; - volatile u32 *reg_addr; - - mutex_lock(&car_mutex); - - /* set up primary or secondary codec/modem space */ -#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) - reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -#else - if (reg == AC97_GPIO_STATUS) - reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; - else - reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -#endif - reg_addr += (reg >> 1); - -#ifndef CONFIG_PXA27x - if (reg == AC97_GPIO_STATUS) { - /* read from controller cache */ - val = *reg_addr; - goto out; - } -#endif - - /* start read access across the ac97 link */ - GSR = GSR_CDONE | GSR_SDONE; - gsr_bits = 0; - val = *reg_addr; - - wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); - if (!((GSR | gsr_bits) & GSR_SDONE)) { - printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n", - __func__, reg, GSR | gsr_bits); - val = -1; - goto out; - } - - /* valid data now */ - GSR = GSR_CDONE | GSR_SDONE; - gsr_bits = 0; - val = *reg_addr; - /* but we've just started another cycle... */ - wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); - -out: mutex_unlock(&car_mutex); - return val; -} - -static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - volatile u32 *reg_addr; - - mutex_lock(&car_mutex); - - /* set up primary or secondary codec/modem space */ -#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) - reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -#else - if (reg == AC97_GPIO_STATUS) - reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; - else - reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -#endif - reg_addr += (reg >> 1); - - GSR = GSR_CDONE | GSR_SDONE; - gsr_bits = 0; - *reg_addr = val; - wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1); - if (!((GSR | gsr_bits) & GSR_CDONE)) - printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n", - __func__, reg, GSR | gsr_bits); - - mutex_unlock(&car_mutex); -} - static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { -#ifdef CONFIG_PXA3xx - int timeout = 100; -#endif - gsr_bits = 0; - -#ifdef CONFIG_PXA27x - /* warm reset broken on Bulverde, - so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); - udelay(10); - GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); - udelay(500); -#elif defined(CONFIG_PXA3xx) - /* Can't use interrupts */ - GCR |= GCR_WARM_RST; - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); -#else - GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); -#endif - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) - printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + pxa2xx_ac97_try_warm_reset(ac97); - GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); - GCR |= GCR_SDONE_IE|GCR_CDONE_IE; + pxa2xx_ac97_finish_reset(ac97); } static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) { -#ifdef CONFIG_PXA3xx - int timeout = 1000; - - /* Hold CLKBPB for 100us */ - GCR = 0; - GCR = GCR_CLKBPB; - udelay(100); - GCR = 0; -#endif + pxa2xx_ac97_try_cold_reset(ac97); - GCR &= GCR_COLD_RST; /* clear everything but nCRST */ - GCR &= ~GCR_COLD_RST; /* then assert nCRST */ - - gsr_bits = 0; -#ifdef CONFIG_PXA27x - /* PXA27x Developers Manual section 13.5.2.2.1 */ - clk_enable(ac97conf_clk); - udelay(5); - clk_disable(ac97conf_clk); - GCR = GCR_COLD_RST; - udelay(50); -#elif defined(CONFIG_PXA3xx) - /* Can't use interrupts on PXA3xx */ - GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); - - GCR = GCR_WARM_RST | GCR_COLD_RST; - while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(10); -#else - GCR = GCR_COLD_RST; - GCR |= GCR_CDONE_IE|GCR_SDONE_IE; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); -#endif - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) - printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); - - GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); - GCR |= GCR_SDONE_IE|GCR_CDONE_IE; -} - -static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) -{ - long status; - - status = GSR; - if (status) { - GSR = status; - gsr_bits |= status; - wake_up(&gsr_wq); - -#ifdef CONFIG_PXA27x - /* Although we don't use those we still need to clear them - since they tend to spuriously trigger when MMC is used - (hardware bug? go figure)... */ - MISR = MISR_EOC; - PISR = PISR_EOC; - MCSR = MCSR_EOC; -#endif - - return IRQ_HANDLED; - } - - return IRQ_NONE; + pxa2xx_ac97_finish_reset(ac97); } struct snd_ac97_bus_ops soc_ac97_ops = { @@ -285,24 +90,13 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { static int pxa2xx_ac97_suspend(struct platform_device *pdev, struct snd_soc_dai *dai) { - GCR |= GCR_ACLINK_OFF; - clk_disable(ac97_clk); - return 0; + return pxa2xx_ac97_hw_suspend(); } static int pxa2xx_ac97_resume(struct platform_device *pdev, struct snd_soc_dai *dai) { - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); - pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); - pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); - pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); -#ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); -#endif - clk_enable(ac97_clk); - return 0; + return pxa2xx_ac97_hw_resume(); } #else @@ -313,61 +107,13 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, static int pxa2xx_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - int ret; - - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); - if (ret < 0) - goto err; - - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); - pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); - pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); - pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); -#ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); - - ac97conf_clk = clk_get(&pdev->dev, "AC97CONFCLK"); - if (IS_ERR(ac97conf_clk)) { - ret = PTR_ERR(ac97conf_clk); - ac97conf_clk = NULL; - goto err_irq; - } -#endif - ac97_clk = clk_get(&pdev->dev, "AC97CLK"); - if (IS_ERR(ac97_clk)) { - ret = PTR_ERR(ac97_clk); - ac97_clk = NULL; - goto err_irq; - } - clk_enable(ac97_clk); - return 0; - - err_irq: - GCR |= GCR_ACLINK_OFF; -#ifdef CONFIG_PXA27x - if (ac97conf_clk) { - clk_put(ac97conf_clk); - ac97conf_clk = NULL; - } -#endif - free_irq(IRQ_AC97, NULL); - err: - return ret; + return pxa2xx_ac97_hw_probe(pdev); } static void pxa2xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { - GCR |= GCR_ACLINK_OFF; - free_irq(IRQ_AC97, NULL); -#ifdef CONFIG_PXA27x - clk_put(ac97conf_clk); - ac97conf_clk = NULL; -#endif - clk_disable(ac97_clk); - clk_put(ac97_clk); - ac97_clk = NULL; + pxa2xx_ac97_hw_remove(pdev); } static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, -- 1.5.6.5

4 12

[alsa-devel] Buffer underrun
by Lopez Cruz, Misael 12 Sep '08

12 Sep '08

Hi, I'm observing an issue with my audio driver in omap platform. When playing back audio files several 'buffer underrun' messages appear, but using -M option of 'aplay', audio is played fine. Since the driver is for omap, I'm using the alsa driver for that platform with minor changes (sound/arm/omap/omap-alsa.c from omap-kernel)? Any idea what could be wrong? Regards, Misael

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 19, Issue 97
by vasudha rao 11 Sep '08

11 Sep '08

Since I got like, $cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0] : control I created nodes according to this like, $ls -l /dev/snd/ total 0 crw-r--r-- 1 root root 116, 5 Jun 14 18:58 controlC0 crw-r--r-- 1 root root 116, 24 Jun 14 18:55 pcmC0D0c crw-r--r-- 1 root root 116, 4 Jun 14 18:59 pcmC0D0p crw-r--r-- 1 root root 116, 2 Jun 14 18:59 timer Now when I played aplay, $aplay alarm.wav Playing WAVE '/home/alarm.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono <3>Bad pte = 5fc2d05f, process = aplay, vm_flags = 800fb, vaddr = 4001c000 Unable to handle kernel NULL pointer dereference at virtual address 00000003 pgd = c2a70000 [00000003] *pgd=22a30031, *pte=00000000, *ppte=00000000 Internal error: Oops: 1 [#2] Modules linked in: CPU: 0 PC is at snd_pcm_mmap_data_nopage+0xc4/0xe8 LR is at 0x4001d000 pc : [<c01bf854>] lr : [<4001d000>] Not tainted sp : c2a4fe6c ip : ffffffff fp : c2a4fe80 r10: 4001d000 r9 : c002bba0 r8 : c2a4e000 r7 : c2e364c4 r6 : 00000000 r5 : c2a4fea0 r4 : c2a7ac00 r3 : 00000093 r2 : 00000013 r1 : 00002000 r0 : c0afa5c0 Flags: nzcv IRQs off FIQs on Mode SVC_32 Segment user Control: 5317F Table: 22A70000 DAC: 00000015 Process aplay (pid: 912, stack limit = 0xc2a4e250) Stack: (0xc2a4fe6c to 0xc2a50000) fe60: 00000000 c29b3544 c2a4fecc c2a4fe84 c0079c58 fe80: c01bf7a0 c2a71000 c002bbe0 5fc2d05f 00000000 00000000 c2a71000 00000000 fea0: 00000002 ffffffeb c03d0040 c2e364c4 c002bba0 c002bbd4 c2a4ffb0 4001d000 fec0: c2a4ff04 c2a4fed0 c003b3a4 c0079b2c 000227d0 00000000 00000017 ffffffff fee0: c02cd04c 00000017 c2a4ffb0 4001d000 4032e000 400cb5fc c2a4ffac c2a4ff08 ff00: c003b5c0 c003b2dc c2a4ff14 c2faddc0 ffffffe7 c0844123 00000000 c0035ec4 ff20: 400cb5fc c2a4ff40 c2a4ff34 c01c15b8 c01c10e4 c2a4ff5c c2a4ff44 c00943d0 ff40: c01c1594 c2faddc0 000227d0 00000004 c2a4ff84 c2a4ff60 c00946ac c00943a4 ff60: 000003d8 00000000 c2faddc0 fffffff7 c0844123 00000036 c2a4ffa4 c2a4ff88 ff80: c009470c c0094428 00000000 ffffffff 4032e320 00000008 00000004 00000004 ffa0: 00000000 c2a4ffb0 c0035cc8 c003b594 00000000 4001d000 4032c000 ffffff00 ffc0: 00000400 4032e320 00000008 00000004 00000004 4032e000 400cb5fc 000004c8 ffe0: ffffff00 bebb29e0 ffffff00 4008ce20 20000010 ffffffff 4009e42c 4009e464 Backtrace: [<c01bf790>] (snd_pcm_mmap_data_nopage+0x0/0xe8) from [<c0079c58>] (__handle_mm_fault+0x13c/0x760) r5 = C29B3544 r4 = 00000000 [<c0079b1c>] (__handle_mm_fault+0x0/0x760) from [<c003b3a4>] (do_page_fault+0xd8/0x20c) [<c003b2cc>] (do_page_fault+0x0/0x20c) from [<c003b5c0>] (do_DataAbort+0x3c/0xa0) [<c003b584>] (do_DataAbort+0x0/0xa0) from [<c0035cc8>] (ret_from_exception+0x0/0x10) r8 = 00000004 r7 = 00000004 r6 = 00000008 r5 = 4032E320 r4 = FFFFFFFF Code: 1590c00c e10f2000 e3823080 e121f003 (e59c3004) Segmentation fault No sound ...:-(( Where I am doing wrong..?? On Thu, Sep 11, 2008 at 6:12 PM, Prithwee Narayan N S < prithweens(a)sanyo.co.in> wrote: > Hi, > > > > Try the following nodes. > > > > # cat /proc/asound/devices > > 0: [ 0] : control > > 16: [ 0- 0]: digital audio playback > > 24: [ 0- 0]: digital audio capture > > 33: : timer > > > > / # ls -l /dev/snd/ > > crw-rw---- 1 0 101 116, 0 Jan 1 00:00 controlC0 > > crw-rw---- 1 0 101 116, 24 Jan 1 00:00 pcmC0D0c > > crw-rw---- 1 0 101 116, 16 Jan 1 00:00 pcmC0D0p > > crw-rw---- 1 0 101 116, 33 Jan 1 00:00 timer > > > > *aplay** -D hw:**0**,0 file.wav* or *aplay** -D plughw:0,0 file.wav*should play your WAV file provided you have the correct alsa.conf file. > > > > Regards, > > Prithwee Narayan. > > > > On Wed, Sep 10, 2008 at 6:16 PM, vasudha rao <vasudha.jrao(a)gmail.com> > wrote: > > > > > > Hi , > > > > > > I want to play and record audio file in my custom atmel 9263 board > > > running with linux(2.6.20). > > > when I tried to play aplay for a wave file, I got > > > > > > $aplay /home/alarm.wav > > > ALSA lib pcm.c:1707:(snd_pcm_open_noupdate) Unknown PCM default > > > aplay: main:446: audio open error: No such file or directory > > > > > > $aplay -l > > > aplay: device_list:187: no soundcards found... > > > > > > $cat /proc/asound/devices > > > 2: : timer > > > 3: : sequencer > > > 4: [ 0- 0]: digital audio playback > > > 5: [ 0] : control > > > > > > Can you tel me where I am doing wrong..?? > > > > > > Thanks in advance.. >

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Re: [alsa-devel] aplay not working
by vasudha rao 11 Sep '08

11 Sep '08

Sir, If you have, can you please send me the o/p of this comand, $ls -lh /dev/snd/ Thank you.. On Thu, Sep 11, 2008 at 2:00 PM, dinesh <dinesh.dua(a)coraltele.com> wrote: > Hi, > As per ur proc enteries you havn't registered any capture device so u can > not do recording with ur device. > > > > -----Original Message----- > *From*: vasudha rao <vasudha.jrao(a)gmail.com<vasudha%20rao%20%3cvasudha.jrao(a)gmail.com%3e> > > > *To*: aggarwal <anuj.aggarwal(a)gmail.com<aggarwal%20%3canuj.aggarwal(a)gmail.com%3e> > > > *Cc*: alsa-devel(a)alsa-project.org > *Subject*: [alsa-devel] aplay not working > *Date*: Thu, 11 Sep 2008 13:33:10 +0530 > > Hi , > > I am not able to do audio record also.. > > I tried following commands.. > > $arecord > RIFF$��.WAVEfmt ...@.@.data��.Recording WAVE 'stdout' : Unsigned 8 bit, Rate > 8000 Hz, Mono > arecord: xrun:1037: read/write error, state = PREPARED > > $aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > $aplay > aplay: main:508: audio open error: Inappropriate ioctl for device > > $cat /proc/asound/cards > 0 [AC97 ]: ac97c - Atmel AC97 > Atmel AC97 Controller at 0xfffa0000, irq 18 > > > $cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.14rc1 (Tue Jan 09 > 09:56:17 2007 UTC). > > $cat /proc/asound/devices > 2: : timer > 3: : sequencer > 4: [ 0- 0]: digital audio playback > 5: [ 0] : control > > > Where I am doing wrong.. > > > > Thank you.. > _______________________________________________ > Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

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[alsa-devel] aplay not working
by vasudha rao 11 Sep '08

11 Sep '08

Hi , I am not able to do audio record also.. I tried following commands.. $arecord RIFF$��.WAVEfmt ...@.@.data��.Recording WAVE 'stdout' : Unsigned 8 bit, Rate 8000 Hz, Mono arecord: xrun:1037: read/write error, state = PREPARED $aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] Subdevices: 1/1 Subdevice #0: subdevice #0 $aplay aplay: main:508: audio open error: Inappropriate ioctl for device $cat /proc/asound/cards 0 [AC97 ]: ac97c - Atmel AC97 Atmel AC97 Controller at 0xfffa0000, irq 18 $cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc1 (Tue Jan 09 09:56:17 2007 UTC). $cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0] : control Where I am doing wrong.. Thank you..

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[alsa-devel] bug in snd_device_name_hint
by Andreas Messer 10 Sep '08

10 Sep '08

Hello, at first I'm sorry for not using the bug-tracking system to report this bug. But I'm not going to create one account for each os-project only to submit a bug report. While playing arround with mumble, I discovered a bug in the alsa library (1.0.17a) function 'snd_device_name_hint': Calling this function leads to a partial destroyed snd_config for my configuration. In my case, after calling this function, an open request for certain pcm devices fails. If you need detailed information, feel free to contact me using my email adress. (I'm not subscribing the mailing list!) I omitted my config here, to keep this mail short... greetings Andreas

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[alsa-devel] [PATCH 01/24] sound/pci: coding style fixes: ac97
by Alexander Beregalov 10 Sep '08

10 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 192 ++++++++++++++++---------------- sound/pci/ac97/ac97_patch.c | 252 +++++++++++++++++++++--------------------- sound/pci/ac97/ac97_pcm.c | 10 +- sound/pci/ac97/ac97_proc.c | 24 ++-- 4 files changed, 239 insertions(+), 239 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acb..0cf819a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -141,33 +141,33 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x43525958, 0xfffffff8, "CS4205", patch_cirrus_spdif, NULL }, { 0x43525960, 0xfffffff8, "CS4291", NULL, NULL }, { 0x43525970, 0xfffffff8, "CS4202", NULL, NULL }, -{ 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II -{ 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different +{ 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, /* SmartMC II */ +{ 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, /* SmartAMC fixme: the mask might be different */ { 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, { 0x44543031, 0xfffffff0, "DT0398", NULL, NULL }, -{ 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028? +{ 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, /* same as TR28028? */ { 0x45838308, 0xffffffff, "ESS1988", NULL, NULL }, { 0x48525300, 0xffffff00, "HMP9701", NULL, NULL }, { 0x49434501, 0xffffffff, "ICE1230", NULL, NULL }, -{ 0x49434511, 0xffffffff, "ICE1232", NULL, NULL }, // alias VIA VT1611A? +{ 0x49434511, 0xffffffff, "ICE1232", NULL, NULL }, /* alias VIA VT1611A? */ { 0x49434514, 0xffffffff, "ICE1232A", NULL, NULL }, -{ 0x49434551, 0xffffffff, "VT1616", patch_vt1616, NULL }, -{ 0x49434552, 0xffffffff, "VT1616i", patch_vt1616, NULL }, // VT1616 compatible (chipset integrated) +{ 0x49434551, 0xffffffff, "VT1616", patch_vt1616, NULL }, +{ 0x49434552, 0xffffffff, "VT1616i", patch_vt1616, NULL }, /* VT1616 compatible (chipset integrated) */ { 0x49544520, 0xffffffff, "IT2226E", NULL, NULL }, { 0x49544561, 0xffffffff, "IT2646E", patch_it2646, NULL }, -{ 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, // only guess --jk +{ 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, /* only guess --jk */ { 0x4e534331, 0xffffffff, "LM4549", NULL, NULL }, -{ 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix +{ 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, /* volume wrap fix */ { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, { 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, -{ 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] -{ 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, /* added by xin jin [07/09/99] */ +{ 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, /* only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] */ { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, -{ 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF -{ 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF +{ 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, /* modified ICE1232 with S/PDIF */ +{ 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, /* modified VT1616 with S/PDIF */ { 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, @@ -186,11 +186,11 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x83847608, 0xffffffff, "STAC9708,11", patch_sigmatel_stac9708, NULL }, { 0x83847609, 0xffffffff, "STAC9721,23", patch_sigmatel_stac9721, NULL }, { 0x83847644, 0xffffffff, "STAC9744", patch_sigmatel_stac9744, NULL }, -{ 0x83847650, 0xffffffff, "STAC9750,51", NULL, NULL }, // patch? -{ 0x83847652, 0xffffffff, "STAC9752,53", NULL, NULL }, // patch? +{ 0x83847650, 0xffffffff, "STAC9750,51", NULL, NULL }, /* patch? */ +{ 0x83847652, 0xffffffff, "STAC9752,53", NULL, NULL }, /* patch? */ { 0x83847656, 0xffffffff, "STAC9756,57", patch_sigmatel_stac9756, NULL }, { 0x83847658, 0xffffffff, "STAC9758,59", patch_sigmatel_stac9758, NULL }, -{ 0x83847666, 0xffffffff, "STAC9766,67", NULL, NULL }, // patch? +{ 0x83847666, 0xffffffff, "STAC9766,67", NULL, NULL }, /* patch? */ { 0, 0, NULL, NULL, NULL } }; @@ -274,7 +274,7 @@ EXPORT_SYMBOL(snd_ac97_write); /** * snd_ac97_read - read a value from the given register - * + * * @ac97: the ac97 instance * @reg: the register to read * @@ -293,9 +293,9 @@ unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg) /* read a register - return the cached value if already read */ static inline unsigned short snd_ac97_read_cache(struct snd_ac97 *ac97, unsigned short reg) { - if (! test_bit(reg, ac97->reg_accessed)) { + if (!test_bit(reg, ac97->reg_accessed)) { ac97->regs[reg] = ac97->bus->ops->read(ac97, reg); - // set_bit(reg, ac97->reg_accessed); + /* set_bit(reg, ac97->reg_accessed); */ } return ac97->regs[reg]; } @@ -437,11 +437,11 @@ static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = e->shift_l == e->shift_r ? 1 : 2; uinfo->value.enumerated.items = e->mask; - + if (uinfo->value.enumerated.item > e->mask - 1) uinfo->value.enumerated.item = e->mask - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); @@ -454,7 +454,7 @@ static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value; unsigned short val, bitmask; - + for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ; val = snd_ac97_read_cache(ac97, e->reg); @@ -472,7 +472,7 @@ static int snd_ac97_put_enum_double(struct snd_kcontrol *kcontrol, struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value; unsigned short val; unsigned short mask, bitmask; - + for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ; if (ucontrol->value.enumerated.item[0] > e->mask - 1) @@ -561,7 +561,7 @@ static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, int invert = (kcontrol->private_value >> 24) & 0x01; int err, page_save; unsigned short val, val2, val_mask; - + page_save = snd_ac97_page_save(ac97, reg, kcontrol); val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -622,8 +622,8 @@ AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, std_mix), AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, std_mic), }; -static const struct snd_kcontrol_new snd_ac97_control_capture_src = -AC97_ENUM("Capture Source", std_enum[0]); +static const struct snd_kcontrol_new snd_ac97_control_capture_src = +AC97_ENUM("Capture Source", std_enum[0]); static const struct snd_kcontrol_new snd_ac97_control_capture_vol = AC97_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0); @@ -690,7 +690,7 @@ static int snd_ac97_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ct uinfo->count = 1; return 0; } - + static int snd_ac97_spdif_cmask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | @@ -702,7 +702,7 @@ static int snd_ac97_spdif_cmask_get(struct snd_kcontrol *kcontrol, struct snd_ct ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS; return 0; } - + static int snd_ac97_spdif_pmask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { /* FIXME: AC'97 spec doesn't say which bits are used for what */ @@ -725,7 +725,7 @@ static int snd_ac97_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_ mutex_unlock(&ac97->reg_mutex); return 0; } - + static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); @@ -752,7 +752,7 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ val |= 1<<3; if (!(new & IEC958_AES0_CON_NOT_COPYRIGHT)) val |= 1<<2; - val |= ((new >> 8) & 0xff) << 4; // category + original + val |= ((new >> 8) & 0xff) << 4; /* category + original */ switch ((new >> 24) & 0xff) { case IEC958_AES3_CON_FS_44100: val |= 0<<12; break; case IEC958_AES3_CON_FS_48000: val |= 2<<12; break; @@ -768,16 +768,16 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ if (ac97->flags & AC97_CS_SPDIF) { int x = (val >> 12) & 0x03; switch (x) { - case 0: x = 1; break; // 44.1 - case 2: x = 0; break; // 48.0 - default: x = 0; break; // illegal. + case 0: x = 1; break; /* 44.1 */ + case 2: x = 0; break; /* 48.0 */ + default: x = 0; break; /* illegal. */ } change |= snd_ac97_update_bits_nolock(ac97, AC97_CSR_SPDIF, 0x3fff, ((val & 0xcfff) | (x << 12))); } else if (ac97->flags & AC97_CX_SPDIF) { int v; v = new & (IEC958_AES0_CON_EMPHASIS_5015|IEC958_AES0_CON_NOT_COPYRIGHT) ? 0 : AC97_CXR_COPYRGT; v |= new & IEC958_AES0_NONAUDIO ? AC97_CXR_SPDIF_AC3 : AC97_CXR_SPDIF_PCM; - change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC, + change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC, AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT, v); } else if (ac97->id == AC97_ID_YMF743) { @@ -806,7 +806,7 @@ static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - // int invert = (kcontrol->private_value >> 24) & 0xff; + /* int invert = (kcontrol->private_value >> 24) & 0xff; */ unsigned short value, old, new; int change; @@ -893,7 +893,7 @@ static int snd_ac97_ad18xx_pcm_get_bits(struct snd_kcontrol *kcontrol, struct sn int lshift = (kcontrol->private_value >> 8) & 0x0f; int rshift = (kcontrol->private_value >> 12) & 0x0f; int mask = (kcontrol->private_value >> 16) & 0xff; - + ucontrol->value.integer.value[0] = mask - ((ac97->spec.ad18xx.pcmreg[codec] >> lshift) & mask); if (lshift != rshift && (ac97->flags & AC97_STEREO_MUTES)) ucontrol->value.integer.value[1] = mask - ((ac97->spec.ad18xx.pcmreg[codec] >> rshift) & mask); @@ -908,7 +908,7 @@ static int snd_ac97_ad18xx_pcm_put_bits(struct snd_kcontrol *kcontrol, struct sn int rshift = (kcontrol->private_value >> 12) & 0x0f; int mask = (kcontrol->private_value >> 16) & 0xff; unsigned short val, valmask; - + val = (mask - (ucontrol->value.integer.value[0] & mask)) << lshift; valmask = mask << lshift; if (lshift != rshift && (ac97->flags & AC97_STEREO_MUTES)) { @@ -936,7 +936,7 @@ static int snd_ac97_ad18xx_pcm_get_volume(struct snd_kcontrol *kcontrol, struct { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); int codec = kcontrol->private_value & 3; - + mutex_lock(&ac97->page_mutex); ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31); ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31); @@ -949,7 +949,7 @@ static int snd_ac97_ad18xx_pcm_put_volume(struct snd_kcontrol *kcontrol, struct struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); int codec = kcontrol->private_value & 3; unsigned short val1, val2; - + val1 = 31 - (ucontrol->value.integer.value[0] & 31); val2 = 31 - (ucontrol->value.integer.value[1] & 31); return snd_ac97_ad18xx_update_pcm_bits(ac97, codec, 0x1f1f, (val1 << 8) | val2); @@ -1023,11 +1023,11 @@ static int snd_ac97_dev_free(struct snd_device *device) return snd_ac97_free(ac97); } -static int snd_ac97_try_volume_mix(struct snd_ac97 * ac97, int reg) +static int snd_ac97_try_volume_mix(struct snd_ac97 *ac97, int reg) { unsigned short val, mask = 0x8000; - if (! snd_ac97_valid_reg(ac97, reg)) + if (!snd_ac97_valid_reg(ac97, reg)) return 0; switch (reg) { @@ -1092,7 +1092,7 @@ static void check_volume_resolution(struct snd_ac97 *ac97, int reg, unsigned cha } *lo_max = *hi_max = 0; - for (i = 0 ; i < ARRAY_SIZE(cbit); i++) { + for (i = 0; i < ARRAY_SIZE(cbit); i++) { unsigned short val; snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8)); /* Do the read twice due to buffers on some ac97 codecs. @@ -1110,7 +1110,7 @@ static void check_volume_resolution(struct snd_ac97 *ac97, int reg, unsigned cha } } -static int snd_ac97_try_bit(struct snd_ac97 * ac97, int reg, int bit) +static int snd_ac97_try_bit(struct snd_ac97 *ac97, int reg, int bit) { unsigned short mask, val, orig, res; @@ -1124,7 +1124,7 @@ static int snd_ac97_try_bit(struct snd_ac97 * ac97, int reg, int bit) } /* check the volume resolution of center/lfe */ -static void snd_ac97_change_volume_params2(struct snd_ac97 * ac97, int reg, int shift, unsigned char *max) +static void snd_ac97_change_volume_params2(struct snd_ac97 *ac97, int reg, int shift, unsigned char *max) { unsigned short val, val1; @@ -1151,7 +1151,7 @@ static inline int printable(unsigned int x) } static struct snd_kcontrol *snd_ac97_cnew(const struct snd_kcontrol_new *_template, - struct snd_ac97 * ac97) + struct snd_ac97 *ac97) { struct snd_kcontrol_new template; memcpy(&template, _template, sizeof(template)); @@ -1170,7 +1170,7 @@ static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, int err; unsigned short val, val1, mute_mask; - if (! snd_ac97_valid_reg(ac97, reg)) + if (!snd_ac97_valid_reg(ac97, reg)) return 0; mute_mask = 0x8000; @@ -1238,7 +1238,7 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne int err; struct snd_kcontrol *kctl; - if (! snd_ac97_valid_reg(ac97, reg)) + if (!snd_ac97_valid_reg(ac97, reg)) return 0; if (hi_max) { /* invert */ @@ -1275,7 +1275,7 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, char name[44]; unsigned char lo_max, hi_max; - if (! snd_ac97_valid_reg(ac97, reg)) + if (!snd_ac97_valid_reg(ac97, reg)) return 0; if (snd_ac97_try_bit(ac97, reg, 15)) { @@ -1301,7 +1301,7 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, static unsigned int snd_ac97_determine_spdif_rates(struct snd_ac97 *ac97); -static int snd_ac97_mixer_build(struct snd_ac97 * ac97) +static int snd_ac97_mixer_build(struct snd_ac97 *ac97) { struct snd_card *card = ac97->bus->card; struct snd_kcontrol *kctl; @@ -1325,7 +1325,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080; /* build center controls */ - if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) + if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) && !(ac97->flags & AC97_AD_MULTI)) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_center[0], ac97))) < 0) return err; @@ -1353,7 +1353,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build surround controls */ - if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) + if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) && !(ac97->flags & AC97_AD_MULTI)) { /* Surround Master (0x38) is with stereo mutes */ if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", @@ -1368,14 +1368,14 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) AC97_HEADPHONE, 0, ac97)) < 0) return err; } - + /* build master mono controls */ if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_MONO)) { if ((err = snd_ac97_cmix_new(card, "Master Mono Playback", AC97_MASTER_MONO, 0, ac97)) < 0) return err; } - + /* build master tone controls */ if (!(ac97->flags & AC97_HAS_NO_TONE)) { if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_TONE)) { @@ -1391,9 +1391,9 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) snd_ac97_write_cache(ac97, AC97_MASTER_TONE, 0x0f0f); } } - + /* build PC Speaker controls */ - if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) && + if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) && ((ac97->flags & AC97_HAS_PC_BEEP) || snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) { for (idx = 0; idx < 2; idx++) @@ -1403,7 +1403,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) snd_ac97_write_cache(ac97, AC97_PC_BEEP, snd_ac97_read(ac97, AC97_PC_BEEP) | 0x801e); } - + /* build Phone controls */ if (!(ac97->flags & AC97_HAS_NO_PHONE)) { if (snd_ac97_try_volume_mix(ac97, AC97_PHONE)) { @@ -1412,7 +1412,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) return err; } } - + /* build MIC controls */ if (!(ac97->flags & AC97_HAS_NO_MIC)) { if (snd_ac97_try_volume_mix(ac97, AC97_MIC)) { @@ -1430,7 +1430,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) AC97_LINE, 1, ac97)) < 0) return err; } - + /* build CD controls */ if (!(ac97->flags & AC97_HAS_NO_CD)) { if (snd_ac97_try_volume_mix(ac97, AC97_CD)) { @@ -1439,7 +1439,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) return err; } } - + /* build Video controls */ if (!(ac97->flags & AC97_HAS_NO_VIDEO)) { if (snd_ac97_try_volume_mix(ac97, AC97_VIDEO)) { @@ -1620,7 +1620,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } ac97->spdif_status = SNDRV_PCM_DEFAULT_CON_SPDIF; } - + /* build chip specific controls */ if (ac97->build_ops->build_specific) if ((err = ac97->build_ops->build_specific(ac97)) < 0) @@ -1628,7 +1628,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) if (snd_ac97_try_bit(ac97, AC97_POWERDOWN, 15)) { kctl = snd_ac97_cnew(&snd_ac97_control_eapd, ac97); - if (! kctl) + if (!kctl) return -ENOMEM; if (ac97->scaps & AC97_SCAP_INV_EAPD) set_inv_eapd(ac97, kctl); @@ -1639,11 +1639,11 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) return 0; } -static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97) +static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 *ac97) { int err, idx; - //printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG)); + /* printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG)); */ snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff); @@ -1776,15 +1776,15 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m printable(id >> 16), printable(id >> 8)); pid = look_for_codec_id(snd_ac97_codec_id_vendors, id); - if (! pid) + if (!pid) return; strcpy(name, pid->name); if (ac97 && pid->patch) { if ((modem && (pid->flags & AC97_MODEM_PATCH)) || - (! modem && ! (pid->flags & AC97_MODEM_PATCH))) + (!modem && !(pid->flags & AC97_MODEM_PATCH))) pid->patch(ac97); - } + } pid = look_for_codec_id(snd_ac97_codec_ids, id); if (pid) { @@ -1794,7 +1794,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m sprintf(name + strlen(name), " rev %d", id & ~pid->mask); if (ac97 && pid->patch) { if ((modem && (pid->flags & AC97_MODEM_PATCH)) || - (! modem && ! (pid->flags & AC97_MODEM_PATCH))) + (!modem && !(pid->flags & AC97_MODEM_PATCH))) pid->patch(ac97); } } else @@ -1829,7 +1829,7 @@ static int ac97_reset_wait(struct snd_ac97 *ac97, int timeout, int with_modem) end_time = jiffies + timeout; do { - + /* use preliminary reads to settle the communication */ snd_ac97_read(ac97, AC97_RESET); snd_ac97_read(ac97, AC97_VENDOR_ID1); @@ -1872,7 +1872,7 @@ static int ac97_reset_wait(struct snd_ac97 *ac97, int timeout, int with_modem) * * The ops table must include valid callbacks (at least read and * write). The other callbacks, wait and reset, are not mandatory. - * + * * The clock is set to 48000. If another clock is needed, set * (*rbus)->clock manually. * @@ -1971,7 +1971,7 @@ static void do_update_power(struct work_struct *work) * * The template must include the codec number (num) and address (addr), * and the private data (private_data). - * + * * The ac97 instance is registered as a low-level device, so you don't * have to release it manually. * @@ -2061,10 +2061,10 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, /* proceed anyway - it's often non-critical */ } } - __access_ok: +__access_ok: ac97->id = snd_ac97_read(ac97, AC97_VENDOR_ID1) << 16; ac97->id |= snd_ac97_read(ac97, AC97_VENDOR_ID2); - if (! (ac97->scaps & AC97_SCAP_DETECT_BY_VENDOR) && + if (!(ac97->scaps & AC97_SCAP_DETECT_BY_VENDOR) && (ac97->id == 0x00000000 || ac97->id == 0xffffffff)) { snd_printk(KERN_ERR "AC'97 %d access is not valid [0x%x], removing mixer.\n", ac97->num, ac97->id); snd_ac97_free(ac97); @@ -2073,7 +2073,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, pid = look_for_codec_id(snd_ac97_codec_ids, ac97->id); if (pid) ac97->flags |= pid->flags; - + /* test for AC'97 */ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO) && !(ac97->scaps & AC97_SCAP_AUDIO)) { /* test if we can write to the record gain volume register */ @@ -2104,14 +2104,14 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, return -EACCES; } - if (bus->ops->reset) // FIXME: always skipping? + if (bus->ops->reset) /* FIXME: always skipping? */ goto __ready_ok; /* FIXME: add powerdown control */ if (ac97_is_audio(ac97)) { /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0); - if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) { + if (!(ac97->flags & AC97_DEFAULT_POWER_OFF)) { snd_ac97_write_cache(ac97, AC97_RESET, 0); /* reset to defaults */ udelay(100); snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0); @@ -2158,8 +2158,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } while (time_after_eq(end_time, jiffies)); snd_printk(KERN_WARNING "MC'97 %d converters and GPIO not ready (0x%x)\n", ac97->num, snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS)); } - - __ready_ok: + +__ready_ok: if (ac97_is_audio(ac97)) ac97->addr = (ac97->ext_id & AC97_EI_ADDR_MASK) >> AC97_EI_ADDR_SHIFT; else @@ -2167,7 +2167,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, if (ac97->ext_id & 0x01c9) { /* L/R, MIC, SDAC, LDAC VRA support */ reg = snd_ac97_read(ac97, AC97_EXTENDED_STATUS); reg |= ac97->ext_id & 0x01c0; /* LDAC/SDAC/CDAC */ - if (! bus->no_vra) + if (!bus->no_vra) reg |= ac97->ext_id & 0x0009; /* VRA/VRM */ snd_ac97_write_cache(ac97, AC97_EXTENDED_STATUS, reg); } @@ -2210,8 +2210,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, if (bus->ops->init) bus->ops->init(ac97); snd_ac97_get_name(ac97, ac97->id, name, !ac97_is_audio(ac97)); - snd_ac97_get_name(NULL, ac97->id, name, !ac97_is_audio(ac97)); // ac97->id might be changed in the special setup code - if (! ac97->build_ops) + snd_ac97_get_name(NULL, ac97->id, name, !ac97_is_audio(ac97)); /* ac97->id might be changed in the special setup code */ + if (!ac97->build_ops) ac97->build_ops = &null_build_ops; if (ac97_is_audio(ac97)) { @@ -2295,7 +2295,7 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97) /* powerdown external amplifier */ if (ac97->scaps & AC97_SCAP_INV_EAPD) power = ac97->regs[AC97_POWERDOWN] & ~AC97_PD_EAPD; - else if (! (ac97->scaps & AC97_SCAP_EAPD_LED)) + else if (!(ac97->scaps & AC97_SCAP_EAPD_LED)) power = ac97->regs[AC97_POWERDOWN] | AC97_PD_EAPD; power |= AC97_PD_PR6; /* Headphone amplifier powerdown */ power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */ @@ -2347,7 +2347,7 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) { int i; - if (! ac97) + if (!ac97) return 0; if (reg) { @@ -2416,8 +2416,8 @@ static void update_power_regs(struct snd_ac97 *ac97) snd_ac97_update_bits(ac97, power_regs[i].power_reg, power_regs[i].mask, bits); } - if (! power_up) { - if (! (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2)) { + if (!power_up) { + if (!(ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2)) { /* power down analog mix and vref */ snd_ac97_update_bits(ac97, AC97_POWERDOWN, AC97_PD_PR2, AC97_PD_PR2); @@ -2437,7 +2437,7 @@ static void update_power_regs(struct snd_ac97 *ac97) */ void snd_ac97_suspend(struct snd_ac97 *ac97) { - if (! ac97) + if (!ac97) return; if (ac97->build_ops->suspend) ac97->build_ops->suspend(ac97); @@ -2457,7 +2457,7 @@ static void snd_ac97_restore_status(struct snd_ac97 *ac97) { int i; - for (i = 2; i < 0x7c ; i += 2) { + for (i = 2; i < 0x7c; i += 2) { if (i == AC97_POWERDOWN || i == AC97_EXTENDED_ID) continue; /* restore only accessible registers @@ -2501,7 +2501,7 @@ void snd_ac97_resume(struct snd_ac97 *ac97) { unsigned long end_time; - if (! ac97) + if (!ac97) return; if (ac97->bus->ops->reset) { @@ -2510,7 +2510,7 @@ void snd_ac97_resume(struct snd_ac97 *ac97) } snd_ac97_write(ac97, AC97_POWERDOWN, 0); - if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) { + if (!(ac97->flags & AC97_DEFAULT_POWER_OFF)) { if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO)) snd_ac97_write(ac97, AC97_RESET, 0); else if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM)) @@ -2568,7 +2568,7 @@ static void set_ctl_name(char *dst, const char *src, const char *suffix) sprintf(dst, "%s %s", src, suffix); else strcpy(dst, src); -} +} /* remove the control with the given name and optional suffix */ static int snd_ac97_remove_ctl(struct snd_ac97 *ac97, const char *name, @@ -2644,7 +2644,7 @@ static int tune_hp_only(struct snd_ac97 *ac97) { struct snd_kcontrol *msw = ctl_find(ac97, "Master Playback Switch", NULL); struct snd_kcontrol *mvol = ctl_find(ac97, "Master Playback Volume", NULL); - if (! msw || ! mvol) + if (!msw || !mvol) return -ENOENT; msw->put = bind_hp_volsw_put; mvol->put = bind_hp_volsw_put; @@ -2699,7 +2699,7 @@ static int tune_ad_sharing(struct snd_ac97 *ac97) return 0; } -static const struct snd_kcontrol_new snd_ac97_alc_jack_detect = +static const struct snd_kcontrol_new snd_ac97_alc_jack_detect = AC97_SINGLE("Jack Detect", AC97_ALC650_CLOCK, 5, 1, 0); /* ac97 tune: set up ALC jack-select */ @@ -2720,7 +2720,7 @@ static int tune_alc_jack(struct snd_ac97 *ac97) static int tune_inv_eapd(struct snd_ac97 *ac97) { struct snd_kcontrol *kctl = ctl_find(ac97, "External Amplifier", NULL); - if (! kctl) + if (!kctl) return -ENOENT; set_inv_eapd(ac97, kctl); return 0; @@ -2749,7 +2749,7 @@ static int master_mute_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int tune_mute_led(struct snd_ac97 *ac97) { struct snd_kcontrol *msw = ctl_find(ac97, "Master Playback Switch", NULL); - if (! msw) + if (!msw) return -ENOENT; msw->put = master_mute_sw_put; snd_ac97_remove_ctl(ac97, "External Amplifier", NULL); @@ -2782,7 +2782,7 @@ static int tune_hp_mute_led(struct snd_ac97 *ac97) { struct snd_kcontrol *msw = ctl_find(ac97, "Master Playback Switch", NULL); struct snd_kcontrol *mvol = ctl_find(ac97, "Master Playback Volume", NULL); - if (! msw || ! mvol) + if (!msw || !mvol) return -ENOENT; msw->put = hp_master_mute_sw_put; mvol->put = bind_hp_volsw_put; @@ -2830,7 +2830,7 @@ static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr) for (i = 0; i < ARRAY_SIZE(applicable_quirks); i++) { q = &applicable_quirks[i]; - if (q->name && ! strcmp(typestr, q->name)) + if (q->name && !strcmp(typestr, q->name)) return apply_quirk(ac97, i); } /* for compatibility, accept the numbers, too */ @@ -2864,13 +2864,13 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, cons return result; } - if (! quirk) + if (!quirk) return -EINVAL; for (; quirk->subvendor; quirk++) { if (quirk->subvendor != ac97->subsystem_vendor) continue; - if ((! quirk->mask && quirk->subdevice == ac97->subsystem_device) || + if ((!quirk->mask && quirk->subdevice == ac97->subsystem_device) || quirk->subdevice == (quirk->mask & ac97->subsystem_device)) { if (quirk->codec_id && quirk->codec_id != ac97->id) continue; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe..19e8572 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -30,7 +30,7 @@ * Chip specific initialization */ -static int patch_build_controls(struct snd_ac97 * ac97, const struct snd_kcontrol_new *controls, int count) +static int patch_build_controls(struct snd_ac97 *ac97, const struct snd_kcontrol_new *controls, int count) { int idx, err; @@ -446,7 +446,7 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = { AC97_YMF7X3_DIT_CTRL, 2, 1, 1) }; -static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) +static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 *ac97) { int err; @@ -460,7 +460,7 @@ static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { .build_post_spdif = patch_yamaha_ymf753_post_spdif }; -static int patch_yamaha_ymf753(struct snd_ac97 * ac97) +static int patch_yamaha_ymf753(struct snd_ac97 *ac97) { /* Patch for Yamaha YMF753, Copyright (c) by David Shust, dshust(a)shustring.com. This chip has nonstandard and extended behaviour with regard to its S/PDIF output. @@ -486,14 +486,14 @@ AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1), AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1), }; -static int patch_wolfson_wm9703_specific(struct snd_ac97 * ac97) +static int patch_wolfson_wm9703_specific(struct snd_ac97 *ac97) { /* This is known to work for the ViewSonic ViewPad 1000 * Randolph Bentson <bentson(a)holmsjoen.com> - * WM9703/9707/9708/9717 + * WM9703/9707/9708/9717 */ int err, i; - + for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) return err; @@ -506,7 +506,7 @@ static struct snd_ac97_build_ops patch_wolfson_wm9703_ops = { .build_specific = patch_wolfson_wm9703_specific, }; -static int patch_wolfson03(struct snd_ac97 * ac97) +static int patch_wolfson03(struct snd_ac97 *ac97) { ac97->build_ops = &patch_wolfson_wm9703_ops; return 0; @@ -521,7 +521,7 @@ AC97_DOUBLE("Rear DAC Volume", AC97_WM9704_RPCM_VOL, 8, 0, 31, 1), AC97_DOUBLE("Surround Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), }; -static int patch_wolfson_wm9704_specific(struct snd_ac97 * ac97) +static int patch_wolfson_wm9704_specific(struct snd_ac97 *ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm9704_snd_ac97_controls); i++) { @@ -537,14 +537,14 @@ static struct snd_ac97_build_ops patch_wolfson_wm9704_ops = { .build_specific = patch_wolfson_wm9704_specific, }; -static int patch_wolfson04(struct snd_ac97 * ac97) +static int patch_wolfson04(struct snd_ac97 *ac97) { /* WM9704M/9704Q */ ac97->build_ops = &patch_wolfson_wm9704_ops; return 0; } -static int patch_wolfson_wm9705_specific(struct snd_ac97 * ac97) +static int patch_wolfson_wm9705_specific(struct snd_ac97 *ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { @@ -559,7 +559,7 @@ static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = { .build_specific = patch_wolfson_wm9705_specific, }; -static int patch_wolfson05(struct snd_ac97 * ac97) +static int patch_wolfson05(struct snd_ac97 *ac97) { /* WM9705, WM9710 */ ac97->build_ops = &patch_wolfson_wm9705_ops; @@ -578,7 +578,7 @@ static const char* wm9711_rec_adc[] = {"Stereo", "Left", "Right", "Mute"}; static const char* wm9711_base[] = {"Linear Control", "Adaptive Boost"}; static const char* wm9711_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; static const char* wm9711_mic[] = {"Mic 1", "Differential", "Mic 2", "Stereo"}; -static const char* wm9711_rec_sel[] = +static const char* wm9711_rec_sel[] = {"Mic 1", "NC", "NC", "Master Mix", "Line", "Headphone Mix", "Phone Mix", "Phone"}; static const char* wm9711_ng_type[] = {"Constant Gain", "Mute"}; @@ -675,10 +675,10 @@ AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), }; -static int patch_wolfson_wm9711_specific(struct snd_ac97 * ac97) +static int patch_wolfson_wm9711_specific(struct snd_ac97 *ac97) { int err, i; - + for (i = 0; i < ARRAY_SIZE(wm9711_snd_ac97_controls); i++) { if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm9711_snd_ac97_controls[i], ac97))) < 0) return err; @@ -696,32 +696,32 @@ static struct snd_ac97_build_ops patch_wolfson_wm9711_ops = { .build_specific = patch_wolfson_wm9711_specific, }; -static int patch_wolfson11(struct snd_ac97 * ac97) +static int patch_wolfson11(struct snd_ac97 *ac97) { /* WM9711, WM9712 */ ac97->build_ops = &patch_wolfson_wm9711_ops; ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_MIC | AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD; - + return 0; } static const char* wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; static const char* wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; -static const char* wm9713_rec_src[] = - {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone Mix", "Master Mix", +static const char* wm9713_rec_src[] = + {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone Mix", "Master Mix", "Mono Mix", "Zh"}; static const char* wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; static const char* wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; static const char* wm9713_mono_pga[] = {"Vmid", "Zh", "Mono Mix", "Inv 1"}; -static const char* wm9713_spk_pga[] = +static const char* wm9713_spk_pga[] = {"Vmid", "Zh", "Headphone Mix", "Master Mix", "Inv", "NC", "NC", "NC"}; static const char* wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone Mix", "NC"}; static const char* wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "NC"}; static const char* wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "NC"}; -static const char* wm9713_dac_inv[] = - {"Off", "Mono Mix", "Master Mix", "Headphone Mix L", "Headphone Mix R", +static const char* wm9713_dac_inv[] = + {"Off", "Mono Mix", "Master Mix", "Headphone Mix L", "Headphone Mix R", "Headphone Mix Mono", "NC", "Vmid"}; static const char* wm9713_base[] = {"Linear Control", "Adaptive Boost"}; static const char* wm9713_ng_type[] = {"Constant Gain", "Mute"}; @@ -842,10 +842,10 @@ AC97_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), AC97_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; -static int patch_wolfson_wm9713_3d (struct snd_ac97 * ac97) +static int patch_wolfson_wm9713_3d(struct snd_ac97 *ac97) { int err, i; - + for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_3d); i++) { if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_3d[i], ac97))) < 0) return err; @@ -853,10 +853,10 @@ static int patch_wolfson_wm9713_3d (struct snd_ac97 * ac97) return 0; } -static int patch_wolfson_wm9713_specific(struct snd_ac97 * ac97) +static int patch_wolfson_wm9713_specific(struct snd_ac97 *ac97) { int err, i; - + for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls); i++) { if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls[i], ac97))) < 0) return err; @@ -872,13 +872,13 @@ static int patch_wolfson_wm9713_specific(struct snd_ac97 * ac97) } #ifdef CONFIG_PM -static void patch_wolfson_wm9713_suspend (struct snd_ac97 * ac97) +static void patch_wolfson_wm9713_suspend(struct snd_ac97 *ac97) { snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xfeff); snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0xffff); } -static void patch_wolfson_wm9713_resume (struct snd_ac97 * ac97) +static void patch_wolfson_wm9713_resume(struct snd_ac97 *ac97) { snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xda00); snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0x3810); @@ -889,13 +889,13 @@ static void patch_wolfson_wm9713_resume (struct snd_ac97 * ac97) static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { .build_specific = patch_wolfson_wm9713_specific, .build_3d = patch_wolfson_wm9713_3d, -#ifdef CONFIG_PM +#ifdef CONFIG_PM .suspend = patch_wolfson_wm9713_suspend, .resume = patch_wolfson_wm9713_resume #endif }; -static int patch_wolfson13(struct snd_ac97 * ac97) +static int patch_wolfson13(struct snd_ac97 *ac97) { /* WM9713, WM9714 */ ac97->build_ops = &patch_wolfson_wm9713_ops; @@ -915,7 +915,7 @@ static int patch_wolfson13(struct snd_ac97 * ac97) /* * Tritech codec */ -static int patch_tritech_tr28028(struct snd_ac97 * ac97) +static int patch_tritech_tr28028(struct snd_ac97 *ac97) { snd_ac97_write_cache(ac97, 0x26, 0x0300); snd_ac97_write_cache(ac97, 0x26, 0x0000); @@ -927,7 +927,7 @@ static int patch_tritech_tr28028(struct snd_ac97 * ac97) /* * Sigmatel STAC97xx codecs */ -static int patch_sigmatel_stac9700_3d(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9700_3d(struct snd_ac97 *ac97) { struct snd_kcontrol *kctl; int err; @@ -940,7 +940,7 @@ static int patch_sigmatel_stac9700_3d(struct snd_ac97 * ac97) return 0; } -static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9708_3d(struct snd_ac97 *ac97) { struct snd_kcontrol *kctl; int err; @@ -968,7 +968,7 @@ AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0), AC97_SINGLE("Sigmatel ADC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 0, 1, 0) }; -static int patch_sigmatel_stac97xx_specific(struct snd_ac97 * ac97) +static int patch_sigmatel_stac97xx_specific(struct snd_ac97 *ac97) { int err; @@ -993,7 +993,7 @@ static struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = { .build_specific = patch_sigmatel_stac97xx_specific }; -static int patch_sigmatel_stac9700(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9700(struct snd_ac97 *ac97) { ac97->build_ops = &patch_sigmatel_stac9700_ops; return 0; @@ -1040,7 +1040,7 @@ static struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = { .build_specific = patch_sigmatel_stac9708_specific }; -static int patch_sigmatel_stac9708(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9708(struct snd_ac97 *ac97) { unsigned int codec72, codec6c; @@ -1066,11 +1066,11 @@ static int patch_sigmatel_stac9708(struct snd_ac97 * ac97) return 0; } -static int patch_sigmatel_stac9721(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9721(struct snd_ac97 *ac97) { ac97->build_ops = &patch_sigmatel_stac9700_ops; if (snd_ac97_read(ac97, AC97_SIGMATEL_ANALOG) == 0) { - // patch for SigmaTel + /* patch for SigmaTel */ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba); snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x4000); snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS1, 0xabba); @@ -1080,9 +1080,9 @@ static int patch_sigmatel_stac9721(struct snd_ac97 * ac97) return 0; } -static int patch_sigmatel_stac9744(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9744(struct snd_ac97 *ac97) { - // patch for SigmaTel + /* patch for SigmaTel */ ac97->build_ops = &patch_sigmatel_stac9700_ops; snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba); snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x0000); /* is this correct? --jk */ @@ -1092,9 +1092,9 @@ static int patch_sigmatel_stac9744(struct snd_ac97 * ac97) return 0; } -static int patch_sigmatel_stac9756(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9756(struct snd_ac97 *ac97) { - // patch for SigmaTel + /* patch for SigmaTel */ ac97->build_ops = &patch_sigmatel_stac9700_ops; snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba); snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x0000); /* is this correct? --jk */ @@ -1269,7 +1269,7 @@ static struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = { .build_specific = patch_sigmatel_stac9758_specific }; -static int patch_sigmatel_stac9758(struct snd_ac97 * ac97) +static int patch_sigmatel_stac9758(struct snd_ac97 *ac97) { static unsigned short regs[4] = { AC97_SIGMATEL_OUTSEL, @@ -1293,12 +1293,12 @@ static int patch_sigmatel_stac9758(struct snd_ac97 * ac97) int i; /* Gateway M675 notebook */ - if (ac97->pci && + if (ac97->pci && ac97->subsystem_vendor == 0x107b && ac97->subsystem_device == 0x0601) pregs = m675_regs; - // patch for SigmaTel + /* patch for SigmaTel */ ac97->build_ops = &patch_sigmatel_stac9758_ops; /* FIXME: assume only page 0 for writing cache */ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR); @@ -1317,7 +1317,7 @@ static const struct snd_kcontrol_new snd_ac97_cirrus_controls_spdif[2] = { AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "AC97-SPSA", AC97_CSR_ACMODE, 0, 3, 0) }; -static int patch_cirrus_build_spdif(struct snd_ac97 * ac97) +static int patch_cirrus_build_spdif(struct snd_ac97 *ac97) { int err; @@ -1343,7 +1343,7 @@ static struct snd_ac97_build_ops patch_cirrus_ops = { .build_spdif = patch_cirrus_build_spdif }; -static int patch_cirrus_spdif(struct snd_ac97 * ac97) +static int patch_cirrus_spdif(struct snd_ac97 *ac97) { /* Basically, the cs4201/cs4205/cs4297a has non-standard sp/dif registers. WHY CAN'T ANYONE FOLLOW THE BLOODY SPEC? *sigh* @@ -1357,18 +1357,18 @@ static int patch_cirrus_spdif(struct snd_ac97 * ac97) */ ac97->build_ops = &patch_cirrus_ops; - ac97->flags |= AC97_CS_SPDIF; + ac97->flags |= AC97_CS_SPDIF; ac97->rates[AC97_RATES_SPDIF] &= ~SNDRV_PCM_RATE_32000; ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ snd_ac97_write_cache(ac97, AC97_CSR_ACMODE, 0x0080); return 0; } -static int patch_cirrus_cs4299(struct snd_ac97 * ac97) +static int patch_cirrus_cs4299(struct snd_ac97 *ac97) { /* force the detection of PC Beep */ ac97->flags |= AC97_HAS_PC_BEEP; - + return patch_cirrus_spdif(ac97); } @@ -1379,7 +1379,7 @@ static const struct snd_kcontrol_new snd_ac97_conexant_controls_spdif[1] = { AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), AC97_CXR_AUDIO_MISC, 3, 1, 0), }; -static int patch_conexant_build_spdif(struct snd_ac97 * ac97) +static int patch_conexant_build_spdif(struct snd_ac97 *ac97) { int err; @@ -1400,7 +1400,7 @@ static struct snd_ac97_build_ops patch_conexant_ops = { .build_spdif = patch_conexant_build_spdif }; -static int patch_conexant(struct snd_ac97 * ac97) +static int patch_conexant(struct snd_ac97 *ac97) { ac97->build_ops = &patch_conexant_ops; ac97->flags |= AC97_CX_SPDIF; @@ -1434,13 +1434,13 @@ static void ad18xx_resume(struct snd_ac97 *ac97) } } - if (! (ac97->flags & AC97_AD_MULTI)) + if (!(ac97->flags & AC97_AD_MULTI)) /* normal restore */ snd_ac97_restore_status(ac97); else { /* restore the AD18xx codec configurations */ for (codec = 0; codec < 3; codec++) { - if (! ac97->spec.ad18xx.id[codec]) + if (!ac97->spec.ad18xx.id[codec]) continue; /* select single codec */ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, @@ -1451,14 +1451,14 @@ static void ad18xx_resume(struct snd_ac97 *ac97) snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000); /* restore status */ - for (i = 2; i < 0x7c ; i += 2) { + for (i = 2; i < 0x7c; i += 2) { if (i == AC97_POWERDOWN || i == AC97_EXTENDED_ID) continue; if (test_bit(i, ac97->reg_accessed)) { /* handle multi codecs for AD18xx */ if (i == AC97_PCM) { for (codec = 0; codec < 3; codec++) { - if (! ac97->spec.ad18xx.id[codec]) + if (!ac97->spec.ad18xx.id[codec]) continue; /* select single codec */ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, @@ -1501,22 +1501,22 @@ static const struct snd_ac97_res_table ad1819_restbl[] = { { } /* terminator */ }; -static int patch_ad1819(struct snd_ac97 * ac97) +static int patch_ad1819(struct snd_ac97 *ac97) { unsigned short scfg; - // patch for Analog Devices + /* patch for Analog Devices */ scfg = snd_ac97_read(ac97, AC97_AD_SERIAL_CFG); snd_ac97_write_cache(ac97, AC97_AD_SERIAL_CFG, scfg | 0x7000); /* select all codecs */ ac97->res_table = ad1819_restbl; return 0; } -static unsigned short patch_ad1881_unchained(struct snd_ac97 * ac97, int idx, unsigned short mask) +static unsigned short patch_ad1881_unchained(struct snd_ac97 *ac97, int idx, unsigned short mask) { unsigned short val; - // test for unchained codec + /* test for unchained codec */ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, mask); snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0000); /* ID0C, ID1C, SDIE = off */ val = snd_ac97_read(ac97, AC97_VENDOR_ID2); @@ -1528,13 +1528,13 @@ static unsigned short patch_ad1881_unchained(struct snd_ac97 * ac97, int idx, un return mask; } -static int patch_ad1881_chained1(struct snd_ac97 * ac97, int idx, unsigned short codec_bits) +static int patch_ad1881_chained1(struct snd_ac97 *ac97, int idx, unsigned short codec_bits) { static int cfg_bits[3] = { 1<<12, 1<<14, 1<<13 }; unsigned short val; - + snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, cfg_bits[idx]); - snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0004); // SDIE + snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0004); /* SDIE */ val = snd_ac97_read(ac97, AC97_VENDOR_ID2); if ((val & 0xff40) != 0x5340) return 0; @@ -1546,26 +1546,26 @@ static int patch_ad1881_chained1(struct snd_ac97 * ac97, int idx, unsigned short return 1; } -static void patch_ad1881_chained(struct snd_ac97 * ac97, int unchained_idx, int cidx1, int cidx2) +static void patch_ad1881_chained(struct snd_ac97 *ac97, int unchained_idx, int cidx1, int cidx2) { - // already detected? + /* already detected? */ if (ac97->spec.ad18xx.unchained[cidx1] || ac97->spec.ad18xx.chained[cidx1]) cidx1 = -1; if (ac97->spec.ad18xx.unchained[cidx2] || ac97->spec.ad18xx.chained[cidx2]) cidx2 = -1; if (cidx1 < 0 && cidx2 < 0) return; - // test for chained codecs + /* test for chained codecs */ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, ac97->spec.ad18xx.unchained[unchained_idx]); - snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0002); // ID1C + snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0002); /* ID1C */ ac97->spec.ad18xx.codec_cfg[unchained_idx] = 0x0002; if (cidx1 >= 0) { if (cidx2 < 0) patch_ad1881_chained1(ac97, cidx1, 0); - else if (patch_ad1881_chained1(ac97, cidx1, 0x0006)) // SDIE | ID1C + else if (patch_ad1881_chained1(ac97, cidx1, 0x0006)) /* SDIE | ID1C */ patch_ad1881_chained1(ac97, cidx2, 0); - else if (patch_ad1881_chained1(ac97, cidx2, 0x0006)) // SDIE | ID1C + else if (patch_ad1881_chained1(ac97, cidx2, 0x0006)) /* SDIE | ID1C */ patch_ad1881_chained1(ac97, cidx1, 0); } else if (cidx2 >= 0) { patch_ad1881_chained1(ac97, cidx2, 0); @@ -1578,15 +1578,15 @@ static struct snd_ac97_build_ops patch_ad1881_build_ops = { #endif }; -static int patch_ad1881(struct snd_ac97 * ac97) +static int patch_ad1881(struct snd_ac97 *ac97) { static const char cfg_idxs[3][2] = { {2, 1}, {0, 2}, {0, 1} }; - - // patch for Analog Devices + + /* patch for Analog Devices */ unsigned short codecs[3]; unsigned short val; int idx, num; @@ -1597,7 +1597,7 @@ static int patch_ad1881(struct snd_ac97 * ac97) codecs[1] = patch_ad1881_unchained(ac97, 1, (1<<14)); codecs[2] = patch_ad1881_unchained(ac97, 2, (1<<13)); - if (! (codecs[0] || codecs[1] || codecs[2])) + if (!(codecs[0] || codecs[1] || codecs[2])) goto __end; for (idx = 0; idx < 3; idx++) @@ -1613,7 +1613,7 @@ static int patch_ad1881(struct snd_ac97 * ac97) ac97->scaps |= AC97_SCAP_CENTER_LFE_DAC; } - __end: +__end: /* select all codecs */ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000); /* check if only one codec is present */ @@ -1623,8 +1623,8 @@ static int patch_ad1881(struct snd_ac97 * ac97) if (num == 1) { /* ok, deselect all ID bits */ snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0000); - ac97->spec.ad18xx.codec_cfg[0] = - ac97->spec.ad18xx.codec_cfg[1] = + ac97->spec.ad18xx.codec_cfg[0] = + ac97->spec.ad18xx.codec_cfg[1] = ac97->spec.ad18xx.codec_cfg[2] = 0x0000; } /* required for AD1886/AD1885 combination */ @@ -1648,7 +1648,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_ad1885[] = { static const DECLARE_TLV_DB_SCALE(db_scale_6bit_6db_max, -8850, 150, 0); -static int patch_ad1885_specific(struct snd_ac97 * ac97) +static int patch_ad1885_specific(struct snd_ac97 *ac97) { int err; @@ -1666,7 +1666,7 @@ static struct snd_ac97_build_ops patch_ad1885_build_ops = { #endif }; -static int patch_ad1885(struct snd_ac97 * ac97) +static int patch_ad1885(struct snd_ac97 *ac97) { patch_ad1881(ac97); /* This is required to deal with the Intel D815EEAL2 */ @@ -1679,7 +1679,7 @@ static int patch_ad1885(struct snd_ac97 * ac97) return 0; } -static int patch_ad1886_specific(struct snd_ac97 * ac97) +static int patch_ad1886_specific(struct snd_ac97 *ac97) { reset_tlv(ac97, "Headphone Playback Volume", db_scale_6bit_6db_max); @@ -1693,7 +1693,7 @@ static struct snd_ac97_build_ops patch_ad1886_build_ops = { #endif }; -static int patch_ad1886(struct snd_ac97 * ac97) +static int patch_ad1886(struct snd_ac97 *ac97) { patch_ad1881(ac97); /* Presario700 workaround */ @@ -1847,7 +1847,7 @@ static const struct snd_kcontrol_new snd_ac97_ad198x_spdif_source = { .put = snd_ac97_ad198x_spdif_source_put, }; -static int patch_ad198x_post_spdif(struct snd_ac97 * ac97) +static int patch_ad198x_post_spdif(struct snd_ac97 *ac97) { return patch_build_controls(ac97, &snd_ac97_ad198x_spdif_source, 1); } @@ -1882,7 +1882,7 @@ static int check_list(struct snd_ac97 *ac97, const unsigned int *list) return 0; } -static int patch_ad1981a_specific(struct snd_ac97 * ac97) +static int patch_ad1981a_specific(struct snd_ac97 *ac97) { if (check_list(ac97, ad1981_jacks_blacklist)) return 0; @@ -2092,10 +2092,10 @@ static struct snd_ac97_build_ops patch_ad1888_build_ops = { .update_jacks = ad1888_update_jacks, }; -static int patch_ad1888(struct snd_ac97 * ac97) +static int patch_ad1888(struct snd_ac97 *ac97) { unsigned short misc; - + patch_ad1881(ac97); ac97->build_ops = &patch_ad1888_build_ops; @@ -2141,7 +2141,7 @@ static struct snd_ac97_build_ops patch_ad1980_build_ops = { .update_jacks = ad1888_update_jacks, }; -static int patch_ad1980(struct snd_ac97 * ac97) +static int patch_ad1980(struct snd_ac97 *ac97) { patch_ad1888(ac97); ac97->build_ops = &patch_ad1980_build_ops; @@ -2175,7 +2175,7 @@ static int snd_ac97_ad1985_vrefout_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_ac97_ad1985_vrefout_put(struct snd_kcontrol *kcontrol, +static int snd_ac97_ad1985_vrefout_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { static const int ctrl2reg[4] = {1, 2, 0, 3}; @@ -2256,10 +2256,10 @@ static struct snd_ac97_build_ops patch_ad1985_build_ops = { .update_jacks = ad1985_update_jacks, }; -static int patch_ad1985(struct snd_ac97 * ac97) +static int patch_ad1985(struct snd_ac97 *ac97) { unsigned short misc; - + patch_ad1881(ac97); ac97->build_ops = &patch_ad1985_build_ops; misc = snd_ac97_read(ac97, AC97_AD_MISC); @@ -2548,7 +2548,7 @@ static struct snd_ac97_build_ops patch_ad1986_build_ops = { .update_jacks = ad1986_update_jacks, }; -static int patch_ad1986(struct snd_ac97 * ac97) +static int patch_ad1986(struct snd_ac97 *ac97) { patch_ad1881(ac97); ac97->build_ops = &patch_ad1986_build_ops; @@ -2575,7 +2575,7 @@ static int patch_alc203(struct snd_ac97 *ac97) static void alc650_update_jacks(struct snd_ac97 *ac97) { int shared; - + /* shared Line-In / Surround Out */ shared = is_shared_surrout(ac97); snd_ac97_update_bits(ac97, AC97_ALC650_MULTICH, 1 << 9, @@ -2628,7 +2628,7 @@ static const struct snd_kcontrol_new snd_ac97_spdif_controls_alc650[] = { static const DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_max, -4350, 150, 0); -static int patch_alc650_specific(struct snd_ac97 * ac97) +static int patch_alc650_specific(struct snd_ac97 *ac97) { int err; @@ -2649,7 +2649,7 @@ static struct snd_ac97_build_ops patch_alc650_ops = { .update_jacks = alc650_update_jacks }; -static int patch_alc650(struct snd_ac97 * ac97) +static int patch_alc650(struct snd_ac97 *ac97) { unsigned short val; @@ -2672,7 +2672,7 @@ static int patch_alc650(struct snd_ac97 * ac97) ac97->id == 0x414c4723); /* enable AC97_ALC650_GPIO_SETUP, AC97_ALC650_CLOCK for R/W */ - snd_ac97_write_cache(ac97, AC97_ALC650_GPIO_STATUS, + snd_ac97_write_cache(ac97, AC97_ALC650_GPIO_STATUS, snd_ac97_read(ac97, AC97_ALC650_GPIO_STATUS) | 0x8000); /* Enable SPDIF-IN only on Rev.E and above */ @@ -2680,7 +2680,7 @@ static int patch_alc650(struct snd_ac97 * ac97) /* SPDIF IN with pin 47 */ if (ac97->spec.dev_flags && /* ASUS A6KM requires EAPD */ - ! (ac97->subsystem_vendor == 0x1043 && + !(ac97->subsystem_vendor == 0x1043 && ac97->subsystem_device == 0x1103)) val |= 0x03; /* enable */ else @@ -2710,7 +2710,7 @@ static int patch_alc650(struct snd_ac97 * ac97) static void alc655_update_jacks(struct snd_ac97 *ac97) { int shared; - + /* shared Line-In / Surround Out */ shared = is_shared_surrout(ac97); ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 1 << 9, @@ -2783,7 +2783,7 @@ static const struct snd_kcontrol_new snd_ac97_spdif_controls_alc655[] = { }, }; -static int patch_alc655_specific(struct snd_ac97 * ac97) +static int patch_alc655_specific(struct snd_ac97 *ac97) { int err; @@ -2801,7 +2801,7 @@ static struct snd_ac97_build_ops patch_alc655_ops = { .update_jacks = alc655_update_jacks }; -static int patch_alc655(struct snd_ac97 * ac97) +static int patch_alc655(struct snd_ac97 *ac97) { unsigned int val; @@ -2862,7 +2862,7 @@ static void alc850_update_jacks(struct snd_ac97 *ac97) { int shared; int aux_is_back_surround; - + /* shared Line-In / Surround Out */ shared = is_shared_surrout(ac97); /* SURR 1kOhm (bit4), Amp (bit5) */ @@ -2963,7 +2963,7 @@ static const struct snd_kcontrol_new snd_ac97_cm9738_controls[] = { AC97_CHANNEL_MODE_4CH_CTL, }; -static int patch_cm9738_specific(struct snd_ac97 * ac97) +static int patch_cm9738_specific(struct snd_ac97 *ac97) { return patch_build_controls(ac97, snd_ac97_cm9738_controls, ARRAY_SIZE(snd_ac97_cm9738_controls)); } @@ -2973,7 +2973,7 @@ static struct snd_ac97_build_ops patch_cm9738_ops = { .update_jacks = cm9738_update_jacks }; -static int patch_cm9738(struct snd_ac97 * ac97) +static int patch_cm9738(struct snd_ac97 *ac97) { ac97->build_ops = &patch_cm9738_ops; /* FIXME: can anyone confirm below? */ @@ -3012,7 +3012,7 @@ static int snd_ac97_cmedia_spdif_playback_source_put(struct snd_kcontrol *kcontr struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); return snd_ac97_update_bits(ac97, AC97_CM9739_SPDIF_CTRL, - 0x01 << 1, + 0x01 << 1, (ucontrol->value.enumerated.item[0] & 0x01) << 1); } @@ -3028,7 +3028,7 @@ static const struct snd_kcontrol_new snd_ac97_cm9739_controls_spdif[] = { /* BIT 2: IG_SPIV */ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Valid Switch", AC97_CM9739_SPDIF_CTRL, 2, 1, 0), /* BIT 3: SPI2F */ - AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9739_SPDIF_CTRL, 3, 1, 0), + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9739_SPDIF_CTRL, 3, 1, 0), /* BIT 4: SPI2SDI */ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_CM9739_SPDIF_CTRL, 4, 1, 0), /* BIT 8: SPD32 - 32bit SPDIF - not supported yet */ @@ -3049,12 +3049,12 @@ static const struct snd_kcontrol_new snd_ac97_cm9739_controls[] = { AC97_CHANNEL_MODE_CTL, }; -static int patch_cm9739_specific(struct snd_ac97 * ac97) +static int patch_cm9739_specific(struct snd_ac97 *ac97) { return patch_build_controls(ac97, snd_ac97_cm9739_controls, ARRAY_SIZE(snd_ac97_cm9739_controls)); } -static int patch_cm9739_post_spdif(struct snd_ac97 * ac97) +static int patch_cm9739_post_spdif(struct snd_ac97 *ac97) { return patch_build_controls(ac97, snd_ac97_cm9739_controls_spdif, ARRAY_SIZE(snd_ac97_cm9739_controls_spdif)); } @@ -3065,7 +3065,7 @@ static struct snd_ac97_build_ops patch_cm9739_ops = { .update_jacks = cm9739_update_jacks }; -static int patch_cm9739(struct snd_ac97 * ac97) +static int patch_cm9739(struct snd_ac97 *ac97) { unsigned short val; @@ -3100,7 +3100,7 @@ static int patch_cm9739(struct snd_ac97 * ac97) val = snd_ac97_read(ac97, AC97_CM9739_MULTI_CHAN) & (1 << 4); val |= (1 << 3); val |= (1 << 13); - if (! (ac97->ext_id & AC97_EI_SPDIF)) + if (!(ac97->ext_id & AC97_EI_SPDIF)) val |= (1 << 14); snd_ac97_write_cache(ac97, AC97_CM9739_MULTI_CHAN, val); @@ -3216,19 +3216,19 @@ static const struct snd_kcontrol_new snd_ac97_cm9761_controls_spdif[] = { /* BIT 2: IG_SPIV */ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Valid Switch", AC97_CM9761_SPDIF_CTRL, 2, 1, 0), /* BIT 3: SPI2F */ - AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9761_SPDIF_CTRL, 3, 1, 0), + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9761_SPDIF_CTRL, 3, 1, 0), /* BIT 4: SPI2SDI */ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_CM9761_SPDIF_CTRL, 4, 1, 0), /* BIT 9-10: DAC_CTL */ AC97_ENUM("DAC Clock Source", cm9761_dac_clock_enum), }; -static int patch_cm9761_post_spdif(struct snd_ac97 * ac97) +static int patch_cm9761_post_spdif(struct snd_ac97 *ac97) { return patch_build_controls(ac97, snd_ac97_cm9761_controls_spdif, ARRAY_SIZE(snd_ac97_cm9761_controls_spdif)); } -static int patch_cm9761_specific(struct snd_ac97 * ac97) +static int patch_cm9761_specific(struct snd_ac97 *ac97) { return patch_build_controls(ac97, snd_ac97_cm9761_controls, ARRAY_SIZE(snd_ac97_cm9761_controls)); } @@ -3308,7 +3308,7 @@ static int patch_cm9761(struct snd_ac97 *ac97) return 0; } - + #define AC97_CM9780_SIDE 0x60 #define AC97_CM9780_JACK 0x62 #define AC97_CM9780_MIXER 0x64 @@ -3418,7 +3418,7 @@ static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, return 0; } -static int patch_vt1616_specific(struct snd_ac97 * ac97) +static int patch_vt1616_specific(struct snd_ac97 *ac97) { struct snd_kcontrol *kctl; int err; @@ -3453,7 +3453,7 @@ static struct snd_ac97_build_ops patch_vt1616_ops = { .build_specific = patch_vt1616_specific }; -static int patch_vt1616(struct snd_ac97 * ac97) +static int patch_vt1616(struct snd_ac97 *ac97) { ac97->build_ops = &patch_vt1616_ops; return 0; @@ -3480,7 +3480,7 @@ static int snd_ac97_vt1617a_smart51_info(struct snd_kcontrol *kcontrol, /* ordering in this list reflects vt1617a docs for Reg 20 and * 7a and Table 6 that lays out the matrix NB WRT Table6: SM51 * is SM51EN *AND* it's Bit14, not Bit15 so the table is very - * counter-intuitive */ + * counter-intuitive */ static const char* texts[] = { "LineIn Mic1", "LineIn Mic1 Mic3", "Surr LFE/C Mic3", "LineIn LFE/C Mic3", @@ -3492,18 +3492,18 @@ static int snd_ac97_vt1617a_smart51_info(struct snd_kcontrol *kcontrol, static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ushort usSM51, usMS; + ushort usSM51, usMS; struct snd_ac97 *pac97; - + pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */ /* grab our desired bits, then mash them together in a manner * consistent with Table 6 on page 17 in the 1617a docs */ - + usSM51 = snd_ac97_read(pac97, 0x7a) >> 14; usMS = snd_ac97_read(pac97, 0x20) >> 8; - + ucontrol->value.enumerated.item[0] = (usSM51 << 1) + usMS; return 0; @@ -3512,7 +3512,7 @@ static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, static int snd_ac97_vt1617a_smart51_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ushort usSM51, usMS, usReg; + ushort usSM51, usMS, usReg; struct snd_ac97 *pac97; @@ -3548,7 +3548,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_vt1617a[] = { }, }; -static int patch_vt1617a(struct snd_ac97 * ac97) +static int patch_vt1617a(struct snd_ac97 *ac97) { int err = 0; int val; @@ -3793,7 +3793,7 @@ static const struct snd_kcontrol_new snd_ac97_spdif_controls_it2646[] = { AC97_SINGLE("IEC958 Input Monitor", 0x76, 13, 1, 0), }; -static int patch_it2646_specific(struct snd_ac97 * ac97) +static int patch_it2646_specific(struct snd_ac97 *ac97) { int err; if ((err = patch_build_controls(ac97, snd_ac97_controls_it2646, ARRAY_SIZE(snd_ac97_controls_it2646))) < 0) @@ -3808,7 +3808,7 @@ static struct snd_ac97_build_ops patch_it2646_ops = { .update_jacks = it2646_update_jacks }; -static int patch_it2646(struct snd_ac97 * ac97) +static int patch_it2646(struct snd_ac97 *ac97) { ac97->build_ops = &patch_it2646_ops; /* full DAC volume */ @@ -3828,7 +3828,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_si3036[] = { AC97_DOUBLE("Modem Speaker Volume", 0x5c, 14, 12, 3, 1) }; -static int patch_si3036_specific(struct snd_ac97 * ac97) +static int patch_si3036_specific(struct snd_ac97 *ac97) { int idx, err; for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_si3036); idx++) @@ -3841,10 +3841,10 @@ static struct snd_ac97_build_ops patch_si3036_ops = { .build_specific = patch_si3036_specific, }; -static int mpatch_si3036(struct snd_ac97 * ac97) +static int mpatch_si3036(struct snd_ac97 *ac97) { ac97->build_ops = &patch_si3036_ops; - snd_ac97_write_cache(ac97, 0x5c, 0xf210 ); + snd_ac97_write_cache(ac97, 0x5c, 0xf210); snd_ac97_write_cache(ac97, 0x68, 0); return 0; } @@ -3879,7 +3879,7 @@ static int patch_lm4550(struct snd_ac97 *ac97) return 0; } -/* +/* * UCB1400 codec (http://www.semiconductors.philips.com/acrobat_download/datasheets/UCB1400-0…) */ static const struct snd_kcontrol_new snd_ac97_controls_ucb1400[] = { @@ -3895,7 +3895,7 @@ AC97_SINGLE("DC Filter", 0x6a, 4, 1, 0), AC97_SINGLE("Smart Low Power Mode", 0x6c, 4, 3, 0), }; -static int patch_ucb1400_specific(struct snd_ac97 * ac97) +static int patch_ucb1400_specific(struct snd_ac97 *ac97) { int idx, err; for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_ucb1400); idx++) @@ -3908,7 +3908,7 @@ static struct snd_ac97_build_ops patch_ucb1400_ops = { .build_specific = patch_ucb1400_specific, }; -static int patch_ucb1400(struct snd_ac97 * ac97) +static int patch_ucb1400(struct snd_ac97 *ac97) { ac97->build_ops = &patch_ucb1400_ops; /* enable headphone driver and smart low power mode by default */ diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 48cbda9..cb86273 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -175,7 +175,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate) unsigned short old, bits, reg, mask; unsigned int sbits; - if (! (ac97->ext_id & AC97_EI_SPDIF)) + if (!(ac97->ext_id & AC97_EI_SPDIF)) return -ENODEV; /* TODO: double rate support */ @@ -258,7 +258,7 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate) { int dbl; unsigned int tmp; - + dbl = rate > 48000; if (dbl) { if (!(ac97->flags & AC97_DOUBLE_RATE)) @@ -281,11 +281,11 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate) return -EINVAL; break; case AC97_PCM_SURR_DAC_RATE: - if (! (ac97->scaps & AC97_SCAP_SURROUND_DAC)) + if (!(ac97->scaps & AC97_SCAP_SURROUND_DAC)) return -EINVAL; break; case AC97_PCM_LFE_DAC_RATE: - if (! (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)) + if (!(ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)) return -EINVAL; break; case AC97_SPDIF: @@ -618,7 +618,7 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, } if (reg_ok[cidx] & (1 << (reg - AC97_PCM_FRONT_DAC_RATE))) continue; - //printk(KERN_DEBUG "setting ac97 reg 0x%x to rate %d\n", reg, rate); + /* printk(KERN_DEBUG "setting ac97 reg 0x%x to rate %d\n", reg, rate); */ err = snd_ac97_set_rate(pcm->r[r].codec[cidx], reg, rate); if (err < 0) snd_printk(KERN_ERR "error in snd_ac97_set_rate: cidx=%d, reg=0x%x, rate=%d, err=%d\n", cidx, reg, rate, err); diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 060ea59..11f7e50 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -140,7 +140,7 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe AC97_PAGE_MASK, val & AC97_PAGE_MASK); } - // val = snd_ac97_read(ac97, AC97_RESET); + /* val = snd_ac97_read(ac97, AC97_RESET); */ val = ac97->caps; snd_iprintf(buffer, "Capabilities :%s%s%s%s%s%s\n", val & AC97_BC_DEDICATED_MIC ? " -dedicated MIC PCM IN channel-" : "", @@ -188,7 +188,7 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe ext = snd_ac97_read(ac97, AC97_EXTENDED_ID); if (ext == 0) goto __modem; - + snd_iprintf(buffer, "Extended ID : codec=%i rev=%i%s%s%s%s DSA=%i%s%s%s%s\n", (ext & AC97_EI_ADDR_MASK) >> AC97_EI_ADDR_SHIFT, (ext & AC97_EI_REV_MASK) >> AC97_EI_REV_SHIFT, @@ -295,11 +295,11 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe } - __modem: +__modem: mext = snd_ac97_read(ac97, AC97_EXTENDED_MID); if (mext == 0) return; - + snd_iprintf(buffer, "Extended modem ID: codec=%i%s%s%s%s%s\n", (mext & AC97_MEI_ADDR_MASK) >> AC97_MEI_ADDR_SHIFT, mext & AC97_MEI_CID2 ? " CID2" : "", @@ -342,9 +342,9 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe static void snd_ac97_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ac97 *ac97 = entry->private_data; - + mutex_lock(&ac97->page_mutex); - if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { // Analog Devices AD1881/85/86 + if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { /* Analog Devices AD1881/85/86 */ int idx; for (idx = 0; idx < 3; idx++) if (ac97->spec.ad18xx.id[idx]) { @@ -356,7 +356,7 @@ static void snd_ac97_proc_read(struct snd_info_entry *entry, struct snd_info_buf } /* select all codecs */ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000); - + snd_iprintf(buffer, "\nAD18XX configuration\n"); snd_iprintf(buffer, "Unchained : 0x%04x,0x%04x,0x%04x\n", ac97->spec.ad18xx.unchained[0], @@ -401,13 +401,13 @@ static void snd_ac97_proc_regs_read_main(struct snd_ac97 *ac97, struct snd_info_ } } -static void snd_ac97_proc_regs_read(struct snd_info_entry *entry, +static void snd_ac97_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ac97 *ac97 = entry->private_data; mutex_lock(&ac97->page_mutex); - if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { // Analog Devices AD1881/85/86 + if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { /* Analog Devices AD1881/85/86 */ int idx; for (idx = 0; idx < 3; idx++) @@ -421,11 +421,11 @@ static void snd_ac97_proc_regs_read(struct snd_info_entry *entry, snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000); } else { snd_ac97_proc_regs_read_main(ac97, buffer, 0); - } + } mutex_unlock(&ac97->page_mutex); } -void snd_ac97_proc_init(struct snd_ac97 * ac97) +void snd_ac97_proc_init(struct snd_ac97 *ac97) { struct snd_info_entry *entry; char name[32]; @@ -458,7 +458,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) ac97->proc_regs = entry; } -void snd_ac97_proc_done(struct snd_ac97 * ac97) +void snd_ac97_proc_done(struct snd_ac97 *ac97) { snd_info_free_entry(ac97->proc_regs); ac97->proc_regs = NULL;

2 4

[alsa-devel] Where to report ALSA bugs?
by William Estrada 10 Sep '08

10 Sep '08

Hi group, I am making progress with my ALSA apps but in the process I have found a few bugs. One of them is in the documentation. In http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#g692ad9e59… Under Parameters=> 'Size frames to be written' should be 'frames to be read'. Just FYI. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

4 3

[alsa-devel] Request - emu20k1 support
by Maximilian Güntner 10 Sep '08

10 Sep '08

Hi, what is the current status in the development of the native emu20k1, ( for the low-end X-fis , Gamer, Music... ) driver? thanks - CU beomuex

3 2

[alsa-devel] [PATCH 1/2] ca0106 playback 44100Hz support to SPDIF and playback format & rate constraints
by Ben Stanley 10 Sep '08

10 Sep '08

Dear list, This patch provides the ability to play back digital audio via SPDIF at a sampling rate of 44100Hz. It also implements rate constraints to ensure that the hardware restrictions of the card are not violated. Note that if 44100Hz is used, then all used channels must use that frequency. For example, opening hw:0,0 at 48000 and then opening hw:0,1 at 44100 will fail due to constraints. However, you may successfully open both hw:0,0 and hw:0,1 at 44100 (presume no other channels open). Further details of the restriction may be found in the thread http://thread.gmane.org/gmane.linux.alsa.devel/52149/focus=52345 This patch also implements playback format restrictions, such that only one playback format may be in use at any time. I tested driver playback using the attached script (Test_ca0106_driver.sh). I get good audio output for all combinations of sampling rates and formats except for 192kHz S16 (as noted at the end of the script). Sometimes it works and sometimes it doesn't. The reason for the failure is not apparent at this time. It could have something to do with my receiver. I don't know. The attached script also checks and validates that the rate and format constraints operate as required. I periodically have trouble with this driver not producing sound output. This has been investigated and the cause is explained here http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 I intend to submit a new patch to address this in the future. This patch has been in use for three months on my MythTV system (Ubuntu 8.04). I find it works reliably for me. I also use it with xine, aplay, mplayer, mame, and childsplay. I have checked with checkpatch.pl . Patch is against v1.0.18rc1. Signed-off-by: Ben Stanley Ben.Stanley(a)exemail.com.au This is the patch against alsa-kmirror. --- pci/ca0106/ca0106.h | 7 +- pci/ca0106/ca0106_main.c | 266 ++++++++++++++++++++++++++++++++++++++-------- 2 files changed, 229 insertions(+), 44 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..376fa9c 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.22 + * Version: 0.0.23 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -49,6 +49,8 @@ * Implement support for Line-in capture on SB Live 24bit. * 0.0.22 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.23 + * Add support for playback sampling rate and format constraints. * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -644,6 +646,8 @@ #include "ca_midi.h" +#define DRVNAME "snd-ca0106" + struct snd_ca0106; struct snd_ca0106_channel { @@ -659,6 +663,7 @@ struct snd_ca0106_pcm { struct snd_pcm_substream *substream; int channel_id; unsigned short running; + unsigned short hw_reserved; }; struct snd_ca0106_details { diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 2f8b28a..33f992b 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.25 + * Version: 0.0.26 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -83,9 +83,14 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * 0.0.25 * Powerdown SPI DAC channels when not in use + * 0.0.26 Ben Stanley + * Added support for output at 44100Hz rate (SPDIF only). + * Implemented constraints system for output rate and format. * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. + * Some programs fail to produce sound output (tested on SPDIF). See + * http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 * -- * * TODO: @@ -145,6 +150,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/ac97_codec.h> #include <sound/info.h> @@ -284,9 +290,9 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000), - .rate_min = 48000, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, @@ -318,6 +324,111 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; +static unsigned int all_spdif_playback_rates[] = + {44100, 48000, 96000, 192000}; + +static int hw_rule_playback_rate( + struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int chi, any_44100 = 0, any_non_44100 = 0, mask = 0; + struct snd_ca0106_channel *chp = 0; + struct snd_pcm_runtime *runtime; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (chip->spdif_enable) { + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printd("snd_hw_rule_playback_rate: ch=%d, " + "rate=%d.\n", chi, runtime->rate); + any_44100 += runtime->rate == 44100; + any_non_44100 += runtime->rate != 44100; + } + if (snd_BUG_ON(any_44100 && any_non_44100)) + return -EINVAL; + if (any_44100) + mask = 0x1; + else if (any_non_44100) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC (FIXME Why?) */ + mask = 0xE; + } + snd_printd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + any_44100, any_non_44100, mask, chip->spdif_enable); + return snd_interval_list( + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE), + ARRAY_SIZE(all_spdif_playback_rates), + all_spdif_playback_rates, mask); +} + +static int hw_rule_playback_format( + struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int chi, any_S16 = 0, any_S32 = 0; + struct snd_ca0106_channel *chp = 0; + struct snd_pcm_runtime *runtime; + struct snd_mask fmt, *f = hw_param_mask( + params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printd("snd_hw_rule_playback_format: ch=%d, format=%d.\n", + chi, runtime->format); + any_S16 += runtime->format == SNDRV_PCM_FORMAT_S16_LE; + any_S32 += runtime->format == SNDRV_PCM_FORMAT_S32_LE; + } + if (snd_BUG_ON(any_S16 && any_S32)) + return -EINVAL; + if (any_S16) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (any_S32) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + else { + /* No format yet chosen, so both formats are available. */ + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + } + result = snd_mask_refine(f, &fmt); + snd_printd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + any_S16, any_S32, f->bits[0], fmt.bits[0], result); + return result; +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -508,10 +619,24 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + err = snd_pcm_hw_constraint_integer( + runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + err = snd_pcm_hw_constraint_step( + runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_playback_rate, (void *)chip, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_playback_format, (void *)chip, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -646,6 +771,9 @@ static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream /* hw_free callback */ static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + epcm->hw_reserved = 0; return snd_pcm_lib_free_pages(substream); } @@ -667,53 +795,103 @@ static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime, *runtimei = 0; struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; + struct snd_ca0106_channel *chp = 0; + int channel = epcm->channel_id, chi, any_44100 = 0, any_non_44100 = 0; u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; u32 hcfg_set = 0x00000000; u32 hcfg; - u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_mask = 0xFF0000; u32 reg40_set = 0; u32 reg40; - /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask; + u32 reg71_shift; u32 reg71_set = 0; u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); - /* Rate can be set per channel. */ - /* reg40 control host to fifo */ - /* reg71 controls DAC rate. */ - switch (runtime->rate) { - case 44100: - reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; - break; - case 48000: - reg40_set = 0; - reg71_set = 0; - break; - case 96000: - reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; - break; - case 192000: - reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; - break; - default: - reg40_set = 0; - reg71_set = 0; - break; + epcm->hw_reserved = 1; + /* FIXME CLEAN UP IF spdif_enable IS CHANGED WHILE CHANNELS ARE OPENED + * OR PREVENT THIS FROM HAPPENING. */ + if (emu->spdif_enable) + reg71_shift = 24; /* SPDIF Output Rate */ + else + reg71_shift = 16; /* I2S Output Rate */ + reg71_mask = 0x3 << reg71_shift; + + printk(KERN_DEBUG DRVNAME ": prepare_playback: " + "channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld,period_size=%ld, periods=%u, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + /*printk("dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr,runtime->dma_area, table_base);*/ + /*printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr,emu->buffer.area, emu->buffer.bytes);*/ + /* We are forced to build the settings for all the channels. */ + for (chi = 0; chi < 4; ++chi) { + chp = &(emu->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (chi != channel && !chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtimei = chp->epcm->substream->runtime; + any_44100 += runtimei->rate == 44100; + any_non_44100 += runtimei->rate != 44100; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* We only support 44100 to spdif, not to DAC. + (FIXME WHY?)*/ + if (emu->spdif_enable) { + /* When using 44100, *all* channels + must be set to that rate. */ + reg40_set |= 0x550000; + reg71_set |= 0x1 << reg71_shift; + break; + } else { + printk(KERN_ERR DRVNAME + "prepare_playback: " + "44100Hz is invalid for DAC.\n"); + } + case 48000: + /* reg40_set &= !(0x1 << (chi<<1)); */ + /* reg71_set &= !(0x1 << reg71_shift); */ + break; + case 96000: + reg40_set |= 0x20000 << (chi<<1); + reg71_set |= 0x2 << reg71_shift; + break; + case 192000: + reg40_set |= 0x30000 << (chi<<1); + reg71_set |= 0x3 << reg71_shift; + break; + default: + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %d.\n", + runtimei->rate); + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + printk(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + any_44100, any_non_44100, emu->spdif_enable); + /* Format is a global setting. */ + /* Only the first channel accessed can set this + (enforced by constraints). */ switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: hcfg_set = 0; @@ -1363,8 +1541,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); @@ -1378,7 +1556,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } chip->details = c; if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + printk(KERN_INFO DRVNAME + ": Sound card name=%s, subsystem=0x%x. " + "Forced to subsystem=0x%x\n", c->name, chip->serial, subsystem[dev]); } -- 1.5.4.3

5 15

Re: [alsa-devel] [Alsa-user] X-Fi support ?
by Vedran Miletić 10 Sep '08

10 Sep '08

Titanium is second generation X-Fi, and ALSA has yet to support the 1st generation. It's very unlikely that it will happen anytime soon. 2008/9/10 William Zumwalt <wizumwalt(a)gmail.com>: > Hi all, > I have an X-Fi Titanium Fatal1ty card and have been having a really hard > time trying to get the driver for it installed and was wondering if anyone > had an idea when this card would be supported by alsa. > Thanks, > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > > -- Vedran Miletić

1 0

[alsa-devel] wm8731 capture -dapm failure
by Alan Horstmann 10 Sep '08

10 Sep '08

I've started a new thread to avoid confusion, and copied bits here: On Monday 08 September 2008 21:51, Mark Brown wrote: > On Mon, Sep 08, 2008 at 09:27:13PM +0100, Alan Horstmann wrote: > > With just this one change we have been successful in running the codec > > over spi. At present we do have an issue with dapm during capture > > -nothing is being turned on! Any pointers would be appreciated. > > If you define DEBUG when building soc-dapm.c (a define at the top of the > source file will do the trick) then it will be very verbose about the > decisions it's making - that will normally point out the problem. > Probably either one of the input controls is not configured to make a > path from the input pins to the ADC or the input has been mistakenly > marked as disabled. That gets us lots of info to digest, and it does look like there are no dapm capture paths -listing below. What we cannot fathom is why! Can users of wm8731 confirm successful capture with dapm on any platform? It is not clear at this stage whether we are looking for a problem on our platform, or a general bug in the driver dapm code. Any assistance would be appreciated. Alan widget ADC HiFi Capture stream Capture event 1 widget DAC HiFi Playback stream Capture event 1 reg old 5f new 5f change 0 reg old 5f new 5f change 0 DAPM WM8731 snd_soc_dapm_stream_event Ext Spk: Off in 0 out 1 in static LHPOUT Int Mic: Off in 1 out 0 out static Mic Bias LLINEIN: Off in 0 out 1 out static Line Input RLINEIN: Off in 0 out 1 out static Line Input MICIN: Off in 1 out 0 in static Mic Bias out static Mic Bias Mic Bias: Off in 1 out 0 in static Int Mic in static MICIN out static MICIN Line Input: Off in 0 out 1 in static RLINEIN in static LLINEIN out Line In Input Mux Input Mux: Off in 0 out 1 in Line In Line Input out static ADC ADC: Off in 0 out 1 in static Input Mux RHPOUT: Off in 0 out 1 in static Output Mixer ROUT: Off in 0 out 1 in static Output Mixer LHPOUT: Off in 0 out 1 in static Output Mixer out static Ext Spk LOUT: Off in 0 out 1 in static Output Mixer DAC: Off in 0 out 4 out HiFi Playback Switch Output Mixer Output Mixer: Off in 0 out 4 in HiFi Playback Switch DAC out static LOUT out static LHPOUT out static ROUT out static RHPOUT

2 2

[alsa-devel] Problems with pulseaudio / asla on PS3
by Kevin Gilbert 09 Sep '08

09 Sep '08

Hi, I'm trying to get an jukebox application going on a Sony PlayStation3 using Qt-Embedded, Phonon and PulseAudio. The application is working well on my KDE- based laptop where it appears that the application is using Phonon -> ALSA -> PulseAudio. But there is no sound from the application when run on the PS3. I initially reported this problem to the PulseAudio mailing list. The first post on this topic is at https://tango.0pointer.de/pipermail/pulseaudio- discuss/2008-August/002249.html . In pacticular, see these posts: 1) https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008- September/002346.html where I detail the results of a final test using the latest development version of PulseAudio; and 2) https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008- September/002347.html where a PulseAudio developer diagnoses the problem as being within ALSA. For convenience, the following is from the latter post: "So, what happens here is this: your device can only do non-interleaved audio and is hence configured for it (as we can see from that "slave setup" listing above). Because your hardware lacks a hardware volume control ALSA puts "softvol" on top of the hw device: the "Soft Volume PCM" you can see in the first part. That device is initialized for interleaved audio as you can see. Which is what PA supports and PA asked for. Now, if softval takes non-interleaved from below and gives interleaved to the layer above than it would need to rearrange those samples -- which it however doesn't actually. PA hence checks with those asserts if the audio data is properly arranged -- which it is not as we can see. This is a bug in ALSA's softvol module. Please report this on alsa-devel. softvol falsely claims to be able to rearrange non-interleaved audio to interleaved." Not sure what other information I should post here so I'll wait for any feedback. Thanks in advance, Kevin

4 7

[alsa-devel] [PATCH 1/8] ASoC: Make all codecs depend on rather than selecting I2C
by Mark Brown 09 Sep '08

09 Sep '08

The I2C layer uses I/O operations that aren't available on all architectures and since select bypasses Kconfig dependency checking selecting I2C breaks builds like allmodconfig on some architectures. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5d77dc3..13ae4fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" - select I2C + depends on I2C select SPI select SND_SOC_AK4535 select SND_SOC_UDA1380 -- 1.5.6.5

6 20

[alsa-devel] how is 'dir' in snd_pcm_hw_params_set_periods_near() supposed to work?
by Lennart Poettering 09 Sep '08

09 Sep '08

Heya! How is the 'dir' argument in snd_pcm_hw_params_set_periods_near() supposed to work? I'd assume that passing dir=1 would mean that it is looked for a number of periods that is greater *or equal* to what we request. If we pass dir=-1 we'd be looking for a number of periods that is smaller *or equal* to what we request. However, what I am experiencing is that there is no "or equal". On my softvol on hda setup here if I ask for 5 periods I will always get 6 if I pass dir=1 -- even if the could give me 5 and will give me five if I pass dir=0. The docs don't really elaborate on this much. I'd certainly think that having ">=" and "<=" here is much more useful and natural than ">" and "<", and hence I was wondering: is this just a bug in the software or is it really intended to be this way? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] [PATCH 09/24] sound/pci: coding style fixes: sound/pci/e*
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ens1370.c | 60 ++++++++++++++++++++-------------------- sound/pci/es1938.c | 68 +++++++++++++++++++++++----------------------- sound/pci/es1968.c | 74 +++++++++++++++++++++++++------------------------- 3 files changed, 101 insertions(+), 101 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 9bf9536..56a8f7c 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -26,7 +26,7 @@ * by Kurt J. Bosch */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -467,13 +467,13 @@ MODULE_DEVICE_TABLE(pci, snd_audiopci_ids); static unsigned int snd_es1370_fixed_rates[] = {5512, 11025, 22050, 44100}; static struct snd_pcm_hw_constraint_list snd_es1370_hw_constraints_rates = { - .count = 4, + .count = 4, .list = snd_es1370_fixed_rates, .mask = 0, }; static struct snd_ratnum es1370_clock = { .num = ES_1370_SRCLOCK, - .den_min = 29, + .den_min = 29, .den_max = 353, .den_step = 1, }; @@ -494,7 +494,7 @@ static struct snd_pcm_hw_constraint_ratdens snd_es1371_hw_constraints_dac_clock }; static struct snd_ratnum es1371_adc_clock = { .num = 48000 << 15, - .den_min = 32768, + .den_min = 32768, .den_max = 393216, .den_step = 1, }; @@ -542,7 +542,7 @@ static unsigned int snd_es1371_src_read(struct ensoniq * ensoniq, unsigned short /* now, wait for busy and the correct time to read */ temp = snd_es1371_wait_src_ready(ensoniq); - + if ((temp & 0x00870000) != 0x00010000) { /* wait for the right state */ for (i = 0; i < POLL_COUNT; i++) { @@ -552,12 +552,12 @@ static unsigned int snd_es1371_src_read(struct ensoniq * ensoniq, unsigned short } } - /* hide the state bits */ + /* hide the state bits */ r = orig & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | ES_1371_DIS_P2 | ES_1371_DIS_R1); r |= ES_1371_SRC_RAM_ADDRO(reg); outl(r, ES_REG(ensoniq, 1371_SMPRATE)); - + return temp; } @@ -648,7 +648,7 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, struct ensoniq *ensoniq = ac97->private_data; unsigned int t, x, fail = 0; - __again: +__again: mutex_lock(&ensoniq->src_mutex); for (t = 0; t < POLL_COUNT; t++) { if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) { @@ -677,7 +677,7 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, /* wait for WIP again */ for (t = 0; t < POLL_COUNT; t++) { if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) - break; + break; } /* now wait for the stinkin' data (RDY) */ for (t = 0; t < POLL_COUNT; t++) { @@ -1053,7 +1053,7 @@ static struct snd_pcm_hardware snd_ensoniq_playback1 = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, #else (SNDRV_PCM_RATE_KNOT | /* 5512Hz rate */ - SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050 | + SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100), #endif .rate_min = 4000, @@ -1072,7 +1072,7 @@ static struct snd_pcm_hardware snd_ensoniq_playback2 = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START), .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, @@ -1424,7 +1424,7 @@ static int snd_es1371_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&ensoniq->reg_lock); ucontrol->value.integer.value[0] = ensoniq->ctrl & ES_1373_SPDIF_THRU ? 1 : 0; spin_unlock_irq(&ensoniq->reg_lock); @@ -1437,7 +1437,7 @@ static int snd_es1371_spdif_put(struct snd_kcontrol *kcontrol, struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); unsigned int nval1, nval2; int change; - + nval1 = ucontrol->value.integer.value[0] ? ES_1373_SPDIF_THRU : 0; nval2 = ucontrol->value.integer.value[0] ? ES_1373_SPDIF_EN : 0; spin_lock_irq(&ensoniq->reg_lock); @@ -1487,7 +1487,7 @@ static int snd_es1373_rear_get(struct snd_kcontrol *kcontrol, { struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); int val = 0; - + spin_lock_irq(&ensoniq->reg_lock); if ((ensoniq->cssr & (ES_1373_REAR_BIT27|ES_1373_REAR_BIT26| ES_1373_REAR_BIT24)) == ES_1373_REAR_BIT26) @@ -1503,7 +1503,7 @@ static int snd_es1373_rear_put(struct snd_kcontrol *kcontrol, struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); unsigned int nval1; int change; - + nval1 = ucontrol->value.integer.value[0] ? ES_1373_REAR_BIT26 : (ES_1373_REAR_BIT27|ES_1373_REAR_BIT24); spin_lock_irq(&ensoniq->reg_lock); @@ -1532,7 +1532,7 @@ static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, { struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); int val = 0; - + spin_lock_irq(&ensoniq->reg_lock); if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4) val = 1; @@ -1547,7 +1547,7 @@ static int snd_es1373_line_put(struct snd_kcontrol *kcontrol, struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); int changed; unsigned int ctrl; - + spin_lock_irq(&ensoniq->reg_lock); ctrl = ensoniq->ctrl; if (ucontrol->value.integer.value[0]) @@ -1690,7 +1690,7 @@ static int snd_ensoniq_control_get(struct snd_kcontrol *kcontrol, { struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol); int mask = kcontrol->private_value; - + spin_lock_irq(&ensoniq->reg_lock); ucontrol->value.integer.value[0] = ensoniq->ctrl & mask ? 1 : 0; spin_unlock_irq(&ensoniq->reg_lock); @@ -1704,7 +1704,7 @@ static int snd_ensoniq_control_put(struct snd_kcontrol *kcontrol, int mask = kcontrol->private_value; unsigned int nval; int change; - + nval = ucontrol->value.integer.value[0] ? mask : 0; spin_lock_irq(&ensoniq->reg_lock); change = (ensoniq->ctrl & mask) != nval; @@ -1864,7 +1864,7 @@ static inline void snd_ensoniq_free_gameport(struct ensoniq *ensoniq) { } */ -static void snd_ensoniq_proc_read(struct snd_info_entry *entry, +static void snd_ensoniq_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct ensoniq *ensoniq = entry->private_data; @@ -1891,7 +1891,7 @@ static void __devinit snd_ensoniq_proc_init(struct ensoniq * ensoniq) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ensoniq->card, "audiopci", &entry)) + if (!snd_card_proc_new(ensoniq->card, "audiopci", &entry)) snd_info_set_text_ops(entry, ensoniq, snd_ensoniq_proc_read); } @@ -1914,7 +1914,7 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) if (ensoniq->irq >= 0) synchronize_irq(ensoniq->irq); pci_set_power_state(ensoniq->pci, 3); - __hw_end: +__hw_end: #ifdef CHIP1370 if (ensoniq->dma_bug.area) snd_dma_free_pages(&ensoniq->dma_bug); @@ -1982,7 +1982,7 @@ static void snd_ensoniq_chip_init(struct ensoniq *ensoniq) udelay(20); outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL)); /* Init the sample rate converter */ - snd_es1371_wait_src_ready(ensoniq); + snd_es1371_wait_src_ready(ensoniq); outl(ES_1371_SRC_DISABLE, ES_REG(ensoniq, 1371_SMPRATE)); for (idx = 0; idx < 0x80; idx++) snd_es1371_src_write(ensoniq, idx, 0); @@ -2021,13 +2021,13 @@ static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct ensoniq *ensoniq = card->private_data; - + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(ensoniq->pcm1); snd_pcm_suspend_all(ensoniq->pcm2); - -#ifdef CHIP1371 + +#ifdef CHIP1371 snd_ac97_suspend(ensoniq->u.es1371.ac97); #else /* try to reset AK4531 */ @@ -2038,7 +2038,7 @@ static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) inw(ES_REG(ensoniq, 1370_CODEC)); udelay(100); snd_ak4531_suspend(ensoniq->u.es1370.ak4531); -#endif +#endif pci_disable_device(pci); pci_save_state(pci); @@ -2063,11 +2063,11 @@ static int snd_ensoniq_resume(struct pci_dev *pci) snd_ensoniq_chip_init(ensoniq); -#ifdef CHIP1371 +#ifdef CHIP1371 snd_ac97_resume(ensoniq->u.es1371.ac97); #else snd_ak4531_resume(ensoniq->u.es1370.ak4531); -#endif +#endif snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2482,7 +2482,7 @@ static struct pci_driver driver = { .resume = snd_ensoniq_resume, #endif }; - + static int __init alsa_card_ens137x_init(void) { return pci_register_driver(&driver); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 4cd9a1f..87effdc 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -5,7 +5,7 @@ * Thomas Sailer <sailer(a)ife.ee.ethz.ch>, * Abramo Bagnara <abramo(a)alsa-project.org>, * Markus Gruber <gruber(a)eikon.tum.de> - * + * * Rewritten from sonicvibes.c source. * * TODO: @@ -63,7 +63,7 @@ #include <sound/initval.h> #include <sound/tlv.h> -#include <asm/io.h> +#include <linux/io.h> MODULE_AUTHOR("Jaromir Koutek <miri(a)punknet.cz>"); MODULE_DESCRIPTION("ESS Solo-1"); @@ -167,7 +167,7 @@ MODULE_PARM_DESC(enable, "Enable ESS Solo-1 soundcard."); #define ESS_CMD_SETFORMAT2 0xb7 #define ESS_CMD_DMACONTROL 0xb8 #define ESS_CMD_DMATYPE 0xb9 -#define ESS_CMD_OFFSETLEFT 0xba +#define ESS_CMD_OFFSETLEFT 0xba #define ESS_CMD_OFFSETRIGHT 0xbb #define ESS_CMD_READREG 0xc0 #define ESS_CMD_ENABLEEXT 0xc6 @@ -418,7 +418,7 @@ static void snd_es1938_reset(struct es1938 *chip) } snd_printk(KERN_ERR "ESS Solo-1 reset failed\n"); - __next: +__next: snd_es1938_write_cmd(chip, ESS_CMD_ENABLEEXT); /* Demand transfer DMA: 4 bytes per DMA request */ @@ -426,7 +426,7 @@ static void snd_es1938_reset(struct es1938 *chip) /* Change behaviour of register A1 4x oversampling - 2nd channel DAC asynchronous */ + 2nd channel DAC asynchronous */ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2MODE, 0x32); /* enable/select DMA channel and IRQ channel */ snd_es1938_bits(chip, ESS_CMD_IRQCONTROL, 0xf0, 0x50); @@ -469,7 +469,7 @@ static struct snd_pcm_hw_constraint_ratnums hw_constraints_clocks = { }; -static void snd_es1938_rate_set(struct es1938 *chip, +static void snd_es1938_rate_set(struct es1938 *chip, struct snd_pcm_substream *substream, int mode) { @@ -482,7 +482,7 @@ static void snd_es1938_rate_set(struct es1938 *chip, /* set filter register */ div0 = 256 - 7160000*20/(8*82*runtime->rate); - + if (mode == DAC2) { snd_es1938_mixer_write(chip, 0x70, bits); snd_es1938_mixer_write(chip, 0x72, div0); @@ -650,7 +650,7 @@ static int snd_es1938_capture_prepare(struct snd_pcm_substream *substream) chip->dma1_shift = 2 - mono - is8; snd_es1938_reset_fifo(chip); - + /* program type */ snd_es1938_bits(chip, ESS_CMD_ANALOGCONTROL, 0x03, (mono ? 2 : 1)); @@ -663,12 +663,12 @@ static int snd_es1938_capture_prepare(struct snd_pcm_substream *substream) /* initialize and configure ADC */ snd_es1938_write(chip, ESS_CMD_SETFORMAT2, u ? 0x51 : 0x71); - snd_es1938_write(chip, ESS_CMD_SETFORMAT2, 0x90 | - (u ? 0x00 : 0x20) | - (is8 ? 0x00 : 0x04) | + snd_es1938_write(chip, ESS_CMD_SETFORMAT2, 0x90 | + (u ? 0x00 : 0x20) | + (is8 ? 0x00 : 0x04) | (mono ? 0x40 : 0x08)); - // snd_es1938_reset_fifo(chip); + /* snd_es1938_reset_fifo(chip); */ /* 11. configure system interrupt controller and DMA controller */ snd_es1938_capture_setdma(chip); @@ -713,7 +713,7 @@ static int snd_es1938_playback1_prepare(struct snd_pcm_substream *substream) /* program DMA */ snd_es1938_playback1_setdma(chip); - + return 0; } @@ -735,10 +735,10 @@ static int snd_es1938_playback2_prepare(struct snd_pcm_substream *substream) chip->dma1_shift = 2 - mono - is8; count = 0x10000 - count; - + /* reset */ snd_es1938_reset_fifo(chip); - + snd_es1938_bits(chip, ESS_CMD_ANALOGCONTROL, 0x03, (mono ? 2 : 1)); /* set clock and counters */ @@ -749,13 +749,13 @@ static int snd_es1938_playback2_prepare(struct snd_pcm_substream *substream) /* initialized and configure DAC */ snd_es1938_write(chip, ESS_CMD_SETFORMAT, u ? 0x80 : 0x00); snd_es1938_write(chip, ESS_CMD_SETFORMAT, u ? 0x51 : 0x71); - snd_es1938_write(chip, ESS_CMD_SETFORMAT2, + snd_es1938_write(chip, ESS_CMD_SETFORMAT2, 0x90 | (mono ? 0x40 : 0x08) | (is8 ? 0x00 : 0x04) | (u ? 0x00 : 0x20)); /* program DMA */ snd_es1938_playback2_setdma(chip); - + return 0; } @@ -1036,7 +1036,7 @@ static int __devinit snd_es1938_new_pcm(struct es1938 *chip, int device) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1938_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_es1938_capture_ops); - + pcm->private_data = chip; pcm->info_flags = 0; strcpy(pcm->name, "ESS Solo-1"); @@ -1049,7 +1049,7 @@ static int __devinit snd_es1938_new_pcm(struct es1938 *chip, int device) } /* ------------------------------------------------------------------- - * + * * *** Mixer part *** */ @@ -1083,7 +1083,7 @@ static int snd_es1938_put_mux(struct snd_kcontrol *kcontrol, { struct es1938 *chip = snd_kcontrol_chip(kcontrol); unsigned char val = ucontrol->value.enumerated.item[0]; - + if (val > 7) return -EINVAL; return snd_es1938_mixer_bits(chip, 0x1c, 0x07, val) != val; @@ -1207,7 +1207,7 @@ static int snd_es1938_get_single(struct snd_kcontrol *kcontrol, int mask = (kcontrol->private_value >> 16) & 0xff; int invert = (kcontrol->private_value >> 24) & 0xff; int val; - + val = snd_es1938_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) @@ -1224,7 +1224,7 @@ static int snd_es1938_put_single(struct snd_kcontrol *kcontrol, int mask = (kcontrol->private_value >> 16) & 0xff; int invert = (kcontrol->private_value >> 24) & 0xff; unsigned char val; - + val = (ucontrol->value.integer.value[0] & mask); if (invert) val = mask - val; @@ -1270,7 +1270,7 @@ static int snd_es1938_get_double(struct snd_kcontrol *kcontrol, int mask = (kcontrol->private_value >> 24) & 0xff; int invert = (kcontrol->private_value >> 22) & 1; unsigned char left, right; - + left = snd_es1938_reg_read(chip, left_reg); if (left_reg != right_reg) right = snd_es1938_reg_read(chip, right_reg); @@ -1297,7 +1297,7 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol, int invert = (kcontrol->private_value >> 22) & 1; int change; unsigned char val1, val2, mask1, mask2; - + val1 = ucontrol->value.integer.value[0] & mask; val2 = ucontrol->value.integer.value[1] & mask; if (invert) { @@ -1315,7 +1315,7 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol, if (snd_es1938_reg_bits(chip, right_reg, mask2, val2) != val2) change = 1; } else { - change = (snd_es1938_reg_bits(chip, left_reg, mask1 | mask2, + change = (snd_es1938_reg_bits(chip, left_reg, mask1 | mask2, val1 | val2) != (val1 | val2)); } return change; @@ -1675,7 +1675,7 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) #if 0 printk("Es1938debug - interrupt status: =0x%x\n", status); #endif - + /* AUDIO 1 */ if (status & 0x10) { #if 0 @@ -1695,7 +1695,7 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) else if (chip->active & DAC1) snd_pcm_period_elapsed(chip->playback2_substream); } - + /* AUDIO 2 */ if (status & 0x20) { #if 0 @@ -1731,10 +1731,10 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) /* MPU401 */ if (status & 0x80) { - // the following line is evil! It switches off MIDI interrupt handling after the first interrupt received. - // replacing the last 0 by 0x40 works for ESS-Solo1, but just doing nothing works as well! - // andreas(a)flying-snail.de - // snd_es1938_mixer_bits(chip, ESSSB_IREG_MPU401CONTROL, 0x40, 0); /* ack? */ + /* the following line is evil! It switches off MIDI interrupt handling after the first interrupt received. */ + /* replacing the last 0 by 0x40 works for ESS-Solo1, but just doing nothing works as well! */ + /* andreas(a)flying-snail.de */ + /* snd_es1938_mixer_bits(chip, ESSSB_IREG_MPU401CONTROL, 0x40, 0); ack? */ if (chip->rmidi) { handled = 1; snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); @@ -1781,7 +1781,7 @@ static int __devinit snd_es1938_mixer(struct es1938 *chip) } return 0; } - + static int __devinit snd_es1938_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) @@ -1851,8 +1851,8 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, chip->irq, 0, &chip->rmidi) < 0) { printk(KERN_ERR "es1938: unable to initialize MPU-401\n"); } else { - // this line is vital for MIDI interrupt handling on ess-solo1 - // andreas(a)flying-snail.de + /* this line is vital for MIDI interrupt handling on ess-solo1 */ + /* andreas(a)flying-snail.de */ snd_es1938_mixer_bits(chip, ESSSB_IREG_MPU401CONTROL, 0x40, 0x40); } diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 20ee759..46c383f 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2,7 +2,7 @@ * Driver for ESS Maestro 1/2/2E Sound Card (started 21.8.99) * Copyright (c) by Matze Braun <MatzeBraun(a)gmx.de>. * Takashi Iwai <tiwai(a)suse.de> - * + * * Most of the driver code comes from Zach Brown(zab(a)redhat.com) * Alan Cox OSS Driver * Rewritted from card-es1938.c source. @@ -29,7 +29,7 @@ * * Hardware Description * - * A working Maestro setup contains the Maestro chip wired to a + * A working Maestro setup contains the Maestro chip wired to a * codec or 2. In the Maestro we have the APUs, the ASSP, and the * Wavecache. The APUs can be though of as virtual audio routing * channels. They can take data from a number of sources and perform @@ -38,7 +38,7 @@ * APUs. The ASSP is a wacky DSP like device that ESS is loth * to release docs on. Thankfully it isn't required on the Maestro * until you start doing insane things like FM emulation and surround - * encoding. The codecs are almost always AC-97 compliant codecs, + * encoding. The codecs are almost always AC-97 compliant codecs, * but it appears that early Maestros may have had PT101 (an ESS * part?) wired to them. The only real difference in the Maestro * families is external goop like docking capability, memory for @@ -76,7 +76,7 @@ * on open time. The sonicvibes OSS routines we inherited really want * power of 2 buffers, so we have all those next to each other, then * 512 byte regions for the recording wavecaches. This ends up - * wasting quite a bit of memory. The only fixes I can see would be + * wasting quite a bit of memory. The only fixes I can see would be * getting a kernel allocator that could work in zones, or figuring out * just how to coerce the WP into doing what we want. * @@ -86,7 +86,7 @@ * like the APU interface that is indirect registers gotten at through * the main maestro indirection. Ouch. We spinlock around the actual * ports on a per card basis. This means spinlock activity at each IO - * operation, but the only IO operation clusters are in non critical + * operation, but the only IO operation clusters are in non critical * paths and it makes the code far easier to follow. Interrupts are * blocked while holding the locks because the int handler has to * get at some of them :(. The mixer interface doesn't, however. @@ -94,7 +94,7 @@ * places. */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -896,7 +896,7 @@ snd_es1968_calc_bob_rate(struct es1968 *chip, struct esschan *es, static u32 snd_es1968_compute_rate(struct es1968 *chip, u32 freq) { u32 rate = (freq << 16) / chip->clock; -#if 0 /* XXX: do we need this? */ +#if 0 /* XXX: do we need this? */ if (rate > 0x10000) rate = 0x10000; #endif @@ -971,7 +971,7 @@ static void snd_es1968_program_wavecache(struct es1968 *chip, struct esschan *es { u32 tmpval = (addr - 0x10) & 0xFFF8; - if (! capture) { + if (!capture) { if (!(es->fmt & ESS_FMT_16BIT)) tmpval |= 4; /* 8bit */ if (es->fmt & ESS_FMT_STEREO) @@ -1059,7 +1059,7 @@ static void snd_es1968_playback_setup(struct es1968 *chip, struct esschan *es, and not the APU Regs 4-5. */ apu_set_register(chip, apu, 10, 0x8F00 | (channel ? 0 : 0x10)); - es->apu_mode[channel] += 1; /* stereo */ + es->apu_mode[channel]++; /* stereo */ } else apu_set_register(chip, apu, 10, 0x8F08); } @@ -1255,7 +1255,7 @@ static int snd_es1968_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - if (! es->running) + if (!es->running) break; snd_es1968_pcm_stop(chip, es); es->running = 0; @@ -1273,7 +1273,7 @@ static snd_pcm_uframes_t snd_es1968_pcm_pointer(struct snd_pcm_substream *substr unsigned int ptr; ptr = snd_es1968_get_dma_ptr(chip, es) << es->wav_shift; - + return bytes_to_frames(substream->runtime, ptr % es->dma_size); } @@ -1407,7 +1407,7 @@ static void snd_es1968_free_dmabuf(struct es1968 *chip) { struct list_head *p; - if (! chip->dma.area) + if (!chip->dma.area) return; snd_dma_reserve_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci)); while ((p = chip->buf_list.next) != &chip->buf_list) { @@ -1425,11 +1425,11 @@ snd_es1968_init_dmabuf(struct es1968 *chip) chip->dma.dev.type = SNDRV_DMA_TYPE_DEV; chip->dma.dev.dev = snd_dma_pci_data(chip->pci); - if (! snd_dma_get_reserved_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci))) { + if (!snd_dma_get_reserved_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci))) { err = snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), chip->total_bufsize, &chip->dma); - if (err < 0 || ! chip->dma.area) { + if (err < 0 || !chip->dma.area) { snd_printk(KERN_ERR "es1968: can't allocate dma pages for size %d\n", chip->total_bufsize); return -ENOMEM; @@ -1478,7 +1478,7 @@ static int snd_es1968_hw_params(struct snd_pcm_substream *substream, } chan->memory = snd_es1968_new_memory(chip, size); if (chan->memory == NULL) { - // snd_printd("cannot allocate dma buffer: size = %d\n", size); + /* snd_printd("cannot allocate dma buffer: size = %d\n", size); */ return -ENOMEM; } snd_pcm_set_runtime_buffer(substream, &chan->memory->buf); @@ -1491,7 +1491,7 @@ static int snd_es1968_hw_free(struct snd_pcm_substream *substream) struct es1968 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct esschan *chan; - + if (runtime->private_data == NULL) return 0; chan = runtime->private_data; @@ -1586,7 +1586,7 @@ static int snd_es1968_capture_open(struct snd_pcm_substream *substream) snd_es1968_free_apu_pair(chip, apu1); return apu2; } - + es = kzalloc(sizeof(*es), GFP_KERNEL); if (!es) { snd_es1968_free_apu_pair(chip, apu1); @@ -1854,7 +1854,7 @@ static void snd_es1968_update_pcm(struct es1968 *chip, struct esschan *es) unsigned int hwptr; unsigned int diff; struct snd_pcm_substream *subs = es->substream; - + if (subs == NULL || !es->running) return; @@ -1895,7 +1895,7 @@ static void es1968_update_hw_volume(unsigned long private_data) if (chip->in_suspend) return; - if (! chip->master_switch || ! chip->master_volume) + if (!chip->master_switch || ! chip->master_volume) return; /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */ @@ -2038,7 +2038,7 @@ static void snd_es1968_ac97_reset(struct es1968 *chip) /* save configuration */ save_ringbus_a = inw(ioaddr + 0x36); - //outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38); /* clear second codec id? */ + /* outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38); clear second codec id? */ /* set command/status address i/o to 1st codec */ outw(inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a); outw(inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c); @@ -2072,7 +2072,7 @@ static void snd_es1968_ac97_reset(struct es1968 *chip) udelay(20); outw(0x0009, ioaddr + 0x60); /* write 9 to gpio */ msleep(500); - //outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38); + /* outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38); */ outw(inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a); outw(inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c); @@ -2154,15 +2154,15 @@ static void snd_es1968_chip_init(struct es1968 *chip) /* We used to muck around with pci config space that * we had no business messing with. We don't know enough - * about the machine to know which DMA mode is appropriate, + * about the machine to know which DMA mode is appropriate, * etc. We were guessing wrong on some machines and making * them unhappy. We now trust in the BIOS to do things right, * which almost certainly means a new host of problems will - * arise with broken BIOS implementations. screw 'em. + * arise with broken BIOS implementations. screw 'em. * We're already intolerant of machines that don't assign * IRQs. */ - + /* Config Reg A */ pci_read_config_word(pci, ESM_CONFIG_A, &w); @@ -2172,7 +2172,7 @@ static void snd_es1968_chip_init(struct es1968 *chip) w |= POST_WRITE; /* Posted write */ w |= PCI_TIMING; /* PCI timing on */ /* XXX huh? claims to be reserved.. */ - w &= ~SWAP_LR; /* swap left/right + w &= ~SWAP_LR; /* swap left/right seems to only have effect on SB Emulation */ w &= ~SUBTR_DECODE; /* Subtractive decode off */ @@ -2223,7 +2223,7 @@ static void snd_es1968_chip_init(struct es1968 *chip) w|=1<<3; /* Turn on 978 hardware volume control. */ w&=~(1<<11); /* Turn on 978 mixer volume control. */ pci_write_config_word(pci, 0x58, w); - + /* Sound Reset */ snd_es1968_reset(chip); @@ -2241,14 +2241,14 @@ static void snd_es1968_chip_init(struct es1968 *chip) /* * Reset the CODEC */ - + snd_es1968_ac97_reset(chip); /* Ring Bus Control B */ n = inl(iobase + ESM_RING_BUS_CONTR_B); n &= ~RINGB_EN_SPDIF; /* SPDIF off */ - //w |= RINGB_EN_2CODEC; /* enable 2nd codec */ + /* w |= RINGB_EN_2CODEC; enable 2nd codec */ outl(n, iobase + ESM_RING_BUS_CONTR_B); /* Set hardware volume control registers to midpoints. @@ -2306,7 +2306,7 @@ static void snd_es1968_chip_init(struct es1968 *chip) maestro_write(chip, 0x0D, 0x7632); - /* Wave cache control on - test off, sg off, + /* Wave cache control on - test off, sg off, enable, enable extra chans 1Mb */ w = inw(iobase + WC_CONTROL); @@ -2353,7 +2353,7 @@ static int es1968_suspend(struct pci_dev *pci, pm_message_t state) struct snd_card *card = pci_get_drvdata(pci); struct es1968 *chip = card->private_data; - if (! chip->do_pm) + if (!chip->do_pm) return 0; chip->in_suspend = 1; @@ -2374,7 +2374,7 @@ static int es1968_resume(struct pci_dev *pci) struct es1968 *chip = card->private_data; struct esschan *es; - if (! chip->do_pm) + if (!chip->do_pm) return 0; /* restore all our config */ @@ -2390,7 +2390,7 @@ static int es1968_resume(struct pci_dev *pci) snd_es1968_chip_init(chip); - /* need to restore the base pointers.. */ + /* need to restore the base pointers.. */ if (chip->dma.addr) { /* set PCMBAR */ wave_set_register(chip, 0x01FC, chip->dma.addr >> 12); @@ -2547,7 +2547,7 @@ static int __devinit snd_es1968_create(struct snd_card *card, } chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { + if (!chip) { pci_disable_device(pci); return -ENOMEM; } @@ -2581,7 +2581,7 @@ static int __devinit snd_es1968_create(struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - + /* Clear Maestro_map */ for (i = 0; i < 32; i++) chip->maestro_map[i] = 0; @@ -2648,14 +2648,14 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (!card) return -ENOMEM; - + if (total_bufsize[dev] < 128) total_bufsize[dev] = 128; if (total_bufsize[dev] > 4096) total_bufsize[dev] = 4096; if ((err = snd_es1968_create(card, pci, total_bufsize[dev] * 1024, /* in bytes */ - pcm_substreams_p[dev], + pcm_substreams_p[dev], pcm_substreams_c[dev], pci_id->driver_data, use_pm[dev], @@ -2716,7 +2716,7 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, snd_es1968_start_irq(chip); chip->clock = clock[dev]; - if (! chip->clock) + if (!chip->clock) es1968_measure_clock(chip); sprintf(card->longname, "%s at 0x%lx, irq %i",

1 0

[alsa-devel] [PATCH 24/24] sound/pci: coding style fixes: ymfpci
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ymfpci/ymfpci_main.c | 98 ++++++++++++++++++++-------------------- 1 files changed, 49 insertions(+), 49 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 90d0d62..baf64b9 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -36,7 +36,7 @@ #include <sound/asoundef.h> #include <sound/mpu401.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/byteorder.h> /* @@ -79,7 +79,7 @@ static int snd_ymfpci_codec_ready(struct snd_ymfpci *chip, int secondary) { unsigned long end_time; u32 reg = secondary ? YDSXGR_SECSTATUSADR : YDSXGR_PRISTATUSADR; - + end_time = jiffies + msecs_to_jiffies(750); do { if ((snd_ymfpci_readw(chip, reg) & 0x8000) == 0) @@ -94,7 +94,7 @@ static void snd_ymfpci_codec_write(struct snd_ac97 *ac97, u16 reg, u16 val) { struct snd_ymfpci *chip = ac97->private_data; u32 cmd; - + snd_ymfpci_codec_ready(chip, 0); cmd = ((YDSXG_AC97WRITECMD | reg) << 16) | val; snd_ymfpci_writel(chip, YDSXGR_AC97CMDDATA, cmd); @@ -145,7 +145,7 @@ static u32 snd_ymfpci_calc_lpfK(u32 rate) 0x00570000, 0x06AA0000, 0x18B20000, 0x20930000, 0x2B9A0000, 0x35A10000, 0x3EAA0000, 0x40000000 }; - + if (rate == 44100) return 0x40000000; /* FIXME: What's the right value? */ for (i = 0; i < 8; i++) @@ -161,7 +161,7 @@ static u32 snd_ymfpci_calc_lpfQ(u32 rate) 0x35280000, 0x34A70000, 0x32020000, 0x31770000, 0x31390000, 0x31C90000, 0x33D00000, 0x40000000 }; - + if (rate == 44100) return 0x370A0000; for (i = 0; i < 8; i++) @@ -184,7 +184,7 @@ static void snd_ymfpci_hw_start(struct snd_ymfpci *chip) snd_ymfpci_writel(chip, YDSXGR_MODE, snd_ymfpci_readl(chip, YDSXGR_MODE) | 3); chip->active_bank = snd_ymfpci_readl(chip, YDSXGR_CTRLSELECT) & 1; - __end: +__end: spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -206,7 +206,7 @@ static void snd_ymfpci_hw_stop(struct snd_ymfpci *chip) atomic_set(&chip->interrupt_sleep_count, 0); wake_up(&chip->interrupt_sleep); } - __end: +__end: spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -220,7 +220,7 @@ static int voice_alloc(struct snd_ymfpci *chip, { struct snd_ymfpci_voice *voice, *voice2; int idx; - + *rvoice = NULL; for (idx = 0; idx < YDSXG_PLAYBACK_VOICES; idx += pair ? 2 : 1) { voice = &chip->voices[idx]; @@ -258,12 +258,12 @@ static int snd_ymfpci_voice_alloc(struct snd_ymfpci *chip, { unsigned long flags; int result; - + if (snd_BUG_ON(!rvoice)) return -EINVAL; if (snd_BUG_ON(pair && type != YMFPCI_PCM)) return -EINVAL; - + spin_lock_irqsave(&chip->voice_lock, flags); for (;;) { result = voice_alloc(chip, type, pair, rvoice); @@ -272,14 +272,14 @@ static int snd_ymfpci_voice_alloc(struct snd_ymfpci *chip, /* TODO: synth/midi voice deallocation */ break; } - spin_unlock_irqrestore(&chip->voice_lock, flags); - return result; + spin_unlock_irqrestore(&chip->voice_lock, flags); + return result; } static int snd_ymfpci_voice_free(struct snd_ymfpci *chip, struct snd_ymfpci_voice *pvoice) { unsigned long flags; - + if (snd_BUG_ON(!pvoice)) return -EINVAL; snd_ymfpci_hw_stop(chip); @@ -303,7 +303,7 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ { struct snd_ymfpci_pcm *ypcm; u32 pos, delta; - + if ((ypcm = voice->ypcm) == NULL) return; if (ypcm->substream == NULL) @@ -318,7 +318,7 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ ypcm->period_pos += delta; ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); */ ypcm->period_pos %= ypcm->period_size; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(ypcm->substream); @@ -330,7 +330,7 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ unsigned int next_bank = 1 - chip->active_bank; struct snd_ymfpci_playback_bank *bank; u32 volume; - + bank = &voice->bank[next_bank]; volume = cpu_to_le32(chip->pcm_mixer[subs].left << 15); bank->left_gain_end = volume; @@ -354,7 +354,7 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream struct snd_ymfpci_pcm *ypcm = runtime->private_data; struct snd_ymfpci *chip = ypcm->chip; u32 pos, delta; - + spin_lock(&chip->reg_lock); if (ypcm->running) { pos = le32_to_cpu(chip->bank_capture[ypcm->capture_bank_number][chip->active_bank]->start) >> ypcm->shift; @@ -366,7 +366,7 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { ypcm->period_pos %= ypcm->period_size; - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); */ spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(substream); spin_lock(&chip->reg_lock); @@ -414,7 +414,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream, result = -EINVAL; break; } - __unlock: +__unlock: spin_unlock(&chip->reg_lock); if (kctl) snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id); @@ -468,7 +468,7 @@ static int snd_ymfpci_pcm_voice_alloc(struct snd_ymfpci_pcm *ypcm, int voices) if (ypcm->voices[0] != NULL && ypcm->voices[1] == NULL) { snd_ymfpci_voice_free(ypcm->chip, ypcm->voices[0]); ypcm->voices[0] = NULL; - } + } } err = snd_ymfpci_voice_alloc(ypcm->chip, YMFPCI_PCM, voices > 1, &ypcm->voices[0]); if (err < 0) @@ -615,7 +615,7 @@ static int snd_ymfpci_ac3_done(struct snd_ymfpci *chip) snd_ymfpci_writel(chip, YDSXGR_MAPOFEFFECT, snd_ymfpci_readl(chip, YDSXGR_MAPOFEFFECT) & ~(3 << 3)); spin_unlock_irq(&chip->reg_lock); - // snd_ymfpci_irq_wait(chip); + /* snd_ymfpci_irq_wait(chip); */ if (chip->ac3_tmp_base.area) { snd_dma_free_pages(&chip->ac3_tmp_base); chip->ac3_tmp_base.area = NULL; @@ -642,7 +642,7 @@ static int snd_ymfpci_playback_hw_free(struct snd_pcm_substream *substream) struct snd_ymfpci *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; - + if (runtime->private_data == NULL) return 0; ypcm = runtime->private_data; @@ -836,7 +836,7 @@ static irqreturn_t snd_ymfpci_interrupt(int irq, void *dev_id) static struct snd_pcm_hardware snd_ymfpci_playback = { .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | @@ -905,8 +905,8 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) /* call with spinlock held */ static void ymfpci_open_extension(struct snd_ymfpci *chip) { - if (! chip->rear_opened) { - if (! chip->spdif_opened) /* set AC3 */ + if (!chip->rear_opened) { + if (!chip->spdif_opened) /* set AC3 */ snd_ymfpci_writel(chip, YDSXGR_MODE, snd_ymfpci_readl(chip, YDSXGR_MODE) | (1 << 30)); /* enable second codec (4CHEN) */ @@ -918,8 +918,8 @@ static void ymfpci_open_extension(struct snd_ymfpci *chip) /* call with spinlock held */ static void ymfpci_close_extension(struct snd_ymfpci *chip) { - if (! chip->rear_opened) { - if (! chip->spdif_opened) + if (!chip->rear_opened) { + if (!chip->spdif_opened) snd_ymfpci_writel(chip, YDSXGR_MODE, snd_ymfpci_readl(chip, YDSXGR_MODE) & ~(1 << 30)); snd_ymfpci_writew(chip, YDSXGR_SECCONFIG, @@ -933,7 +933,7 @@ static int snd_ymfpci_playback_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; int err; - + if ((err = snd_ymfpci_playback_open_1(substream)) < 0) return err; ypcm = runtime->private_data; @@ -955,7 +955,7 @@ static int snd_ymfpci_playback_spdif_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; int err; - + if ((err = snd_ymfpci_playback_open_1(substream)) < 0) return err; ypcm = runtime->private_data; @@ -983,7 +983,7 @@ static int snd_ymfpci_playback_4ch_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; int err; - + if ((err = snd_ymfpci_playback_open_1(substream)) < 0) return err; ypcm = runtime->private_data; @@ -1009,7 +1009,7 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, return -ENOMEM; ypcm->chip = chip; ypcm->type = capture_bank_number + CAPTURE_REC; - ypcm->substream = substream; + ypcm->substream = substream; ypcm->capture_bank_number = capture_bank_number; chip->capture_substream[capture_bank_number] = substream; runtime->hw = snd_ymfpci_capture; @@ -1449,7 +1449,7 @@ static int snd_ymfpci_get_single(struct snd_kcontrol *kcontrol, int reg = kcontrol->private_value & 0xffff; unsigned int shift = (kcontrol->private_value >> 16) & 0xff; unsigned int mask = 1; - + switch (reg) { case YDSXGR_SPDIFOUTCTRL: break; case YDSXGR_SPDIFINCTRL: break; @@ -1469,7 +1469,7 @@ static int snd_ymfpci_put_single(struct snd_kcontrol *kcontrol, unsigned int mask = 1; int change; unsigned int val, oval; - + switch (reg) { case YDSXGR_SPDIFOUTCTRL: break; case YDSXGR_SPDIFINCTRL: break; @@ -1515,7 +1515,7 @@ static int snd_ymfpci_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e unsigned int reg = kcontrol->private_value; unsigned int shift_left = 0, shift_right = 16, mask = 16383; unsigned int val; - + if (reg < 0x80 || reg >= 0xc0) return -EINVAL; spin_lock_irq(&chip->reg_lock); @@ -1533,7 +1533,7 @@ static int snd_ymfpci_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e unsigned int shift_left = 0, shift_right = 16, mask = 16383; int change; unsigned int val1, val2, oval; - + if (reg < 0x80 || reg >= 0xc0) return -EINVAL; val1 = ucontrol->value.integer.value[0] & mask; @@ -1557,7 +1557,7 @@ static int snd_ymfpci_put_nativedacvol(struct snd_kcontrol *kcontrol, unsigned int reg2 = YDSXGR_BUF441OUTVOL; int change; unsigned int value, oval; - + value = ucontrol->value.integer.value[0] & 0x3fff; value |= (ucontrol->value.integer.value[1] & 0x3fff) << 16; spin_lock_irq(&chip->reg_lock); @@ -1937,12 +1937,12 @@ int __devinit snd_ymfpci_timer(struct snd_ymfpci *chip, int device) * proc interface */ -static void snd_ymfpci_proc_read(struct snd_info_entry *entry, +static void snd_ymfpci_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ymfpci *chip = entry->private_data; int i; - + snd_iprintf(buffer, "YMFPCI\n\n"); for (i = 0; i <= YDSXGR_WORKBASE; i += 4) snd_iprintf(buffer, "%04x: %04x\n", i, snd_ymfpci_readl(chip, i)); @@ -1951,8 +1951,8 @@ static void snd_ymfpci_proc_read(struct snd_info_entry *entry, static int __devinit snd_ymfpci_proc_init(struct snd_card *card, struct snd_ymfpci *chip) { struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "ymfpci", &entry)) + + if (!snd_card_proc_new(card, "ymfpci", &entry)) snd_info_set_text_ops(entry, chip, snd_ymfpci_proc_read); return 0; } @@ -1966,7 +1966,7 @@ static void snd_ymfpci_aclink_reset(struct pci_dev * pci) u8 cmd; pci_read_config_byte(pci, PCIR_DSXG_CTRL, &cmd); -#if 0 // force to reset +#if 0 /* force to reset */ if (cmd & 0x03) { #endif pci_write_config_byte(pci, PCIR_DSXG_CTRL, cmd & 0xfc); @@ -2082,7 +2082,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) chip->bank_size_capture = snd_ymfpci_readl(chip, YDSXGR_RECCTRLSIZE) << 2; chip->bank_size_effect = snd_ymfpci_readl(chip, YDSXGR_EFFCTRLSIZE) << 2; chip->work_size = YDSXG_DEFAULT_WORK_SIZE; - + size = ALIGN(playback_ctrl_size, 0x100) + ALIGN(chip->bank_size_playback * 2 * YDSXG_PLAYBACK_VOICES, 0x100) + ALIGN(chip->bank_size_capture * 2 * YDSXG_CAPTURE_VOICES, 0x100) + @@ -2091,7 +2091,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) /* work_ptr must be aligned to 256 bytes, but it's already covered with the kernel page allocation mechanism */ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - size, &chip->work_ptr) < 0) + size, &chip->work_ptr) < 0) return -ENOMEM; ptr = chip->work_ptr.area; ptr_addr = chip->work_ptr.addr; @@ -2137,7 +2137,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) ptr_addr = ALIGN(ptr_addr, 0x100); chip->work_base = ptr; chip->work_base_addr = ptr_addr; - + snd_BUG_ON(ptr + chip->work_size != chip->work_ptr.area + chip->work_ptr.bytes); @@ -2166,7 +2166,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) snd_ymfpci_writel(chip, YDSXGR_NATIVEDACINVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_PRIADCLOOPVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_LEGACYOUTVOL, 0x3fff3fff); - + return 0; } @@ -2214,11 +2214,11 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) iounmap(chip->reg_area_virt); if (chip->work_ptr.area) snd_dma_free_pages(&chip->work_ptr); - + release_and_free_resource(chip->res_reg_area); pci_write_config_word(chip->pci, 0x40, chip->old_legacy_ctrl); - + pci_disable_device(chip->pci); release_firmware(chip->dsp_microcode); release_firmware(chip->controller_microcode); @@ -2267,7 +2267,7 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) struct snd_card *card = pci_get_drvdata(pci); struct snd_ymfpci *chip = card->private_data; unsigned int i; - + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(chip->pcm); snd_pcm_suspend_all(chip->pcm2); @@ -2333,7 +2333,7 @@ int __devinit snd_ymfpci_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_ymfpci_dev_free, }; - + *rchip = NULL; /* enable PCI device */

1 0

[alsa-devel] [PATCH 23/24] sound/pci: coding style fixes: vx222
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/vx222/vx222.c | 2 +- sound/pci/vx222/vx222_ops.c | 330 +++++++++++++++++++++--------------------- 2 files changed, 166 insertions(+), 166 deletions(-) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index acc352f..ec7c743 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -154,7 +154,7 @@ static int __devinit snd_vx222_create(struct snd_card *card, struct pci_dev *pci vx_ops = hw->type == VX_TYPE_BOARD ? &vx222_old_ops : &vx222_ops; chip = snd_vx_create(card, hw, vx_ops, sizeof(struct snd_vx222) - sizeof(struct vx_core)); - if (! chip) { + if (!chip) { pci_disable_device(pci); return -ENOMEM; } diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7e87f39..7d1350a 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/tlv.h> -#include <asm/io.h> +#include <linux/io.h> #include "vx222.h" @@ -50,12 +50,12 @@ static int vx2_reg_offset[VX_REG_MAX] = { [VX_HIFREQ] = 0x38, [VX_CSUER] = 0x3c, [VX_SELMIC] = 0x40, - [VX_COMPOT] = 0x44, // Write: POTENTIOMETER ; Read: COMPRESSION LEVEL activate - [VX_SCOMPR] = 0x48, // Read: COMPRESSION THRESHOLD activate - [VX_GLIMIT] = 0x4c, // Read: LEVEL LIMITATION activate - [VX_INTCSR] = 0x4c, // VX_INTCSR_REGISTER_OFFSET - [VX_CNTRL] = 0x50, // VX_CNTRL_REGISTER_OFFSET - [VX_GPIOC] = 0x54, // VX_GPIOC (new with PLX9030) + [VX_COMPOT] = 0x44, /* Write: POTENTIOMETER; Read: COMPRESSION LEVEL activate */ + [VX_SCOMPR] = 0x48, /* Read: COMPRESSION THRESHOLD activate */ + [VX_GLIMIT] = 0x4c, /* Read: LEVEL LIMITATION activate */ + [VX_INTCSR] = 0x4c, /* VX_INTCSR_REGISTER_OFFSET */ + [VX_CNTRL] = 0x50, /* VX_CNTRL_REGISTER_OFFSET */ + [VX_GPIOC] = 0x54, /* VX_GPIOC (new with PLX9030) */ }; static int vx2_reg_index[VX_REG_MAX] = { @@ -107,7 +107,7 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset) static void vx2_outb(struct vx_core *chip, int offset, unsigned char val) { outb(val, vx2_reg_addr(chip, offset)); - //printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); */ } /** @@ -126,7 +126,7 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset) */ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) { - // printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); */ outl(val, vx2_reg_addr(chip, offset)); } @@ -187,7 +187,7 @@ static int vx2_test_xilinx(struct vx_core *_chip) vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_TEST0_MASK); vx_inl(chip, ISR); data = vx_inl(chip, STATUS); - if (! (data & VX_STATUS_VAL_TEST0_MASK)) { + if (!(data & VX_STATUS_VAL_TEST0_MASK)) { snd_printdd("bad! #2\n"); return -ENODEV; } @@ -207,7 +207,7 @@ static int vx2_test_xilinx(struct vx_core *_chip) vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_TEST1_MASK); vx_inl(chip, ISR); data = vx_inl(chip, STATUS); - if (! (data & VX_STATUS_VAL_TEST1_MASK)) { + if (!(data & VX_STATUS_VAL_TEST1_MASK)) { snd_printdd("bad! #4\n"); return -ENODEV; } @@ -400,7 +400,7 @@ static int vx2_load_xilinx_binary(struct vx_core *chip, const struct firmware *x return 0; } - + /* * load the boot/dsp images */ @@ -439,12 +439,12 @@ static int vx2_load_dsp(struct vx_core *vx, int index, const struct firmware *ds static int vx2_test_and_ack(struct vx_core *chip) { /* not booted yet? */ - if (! (chip->chip_status & VX_STAT_XILINX_LOADED)) + if (!(chip->chip_status & VX_STAT_XILINX_LOADED)) return -ENXIO; - if (! (vx_inl(chip, STATUS) & VX_STATUS_MEMIRQ_MASK)) + if (!(vx_inl(chip, STATUS) & VX_STATUS_MEMIRQ_MASK)) return -EIO; - + /* ok, interrupts generated, now ack it */ /* set ACQUIT bit up and down */ vx_outl(chip, STATUS, 0); @@ -512,154 +512,154 @@ static void vx2_write_codec_reg(struct vx_core *chip, unsigned int data) #define AKM_CODEC_RIGHT_LEVEL_CMD 0xA500 static const u8 vx2_akm_gains_lut[VX2_AKM_LEVEL_MAX+1] = { - 0x7f, // [000] = +0.000 dB -> AKM(0x7f) = +0.000 dB error(+0.000 dB) - 0x7d, // [001] = -0.500 dB -> AKM(0x7d) = -0.572 dB error(-0.072 dB) - 0x7c, // [002] = -1.000 dB -> AKM(0x7c) = -0.873 dB error(+0.127 dB) - 0x7a, // [003] = -1.500 dB -> AKM(0x7a) = -1.508 dB error(-0.008 dB) - 0x79, // [004] = -2.000 dB -> AKM(0x79) = -1.844 dB error(+0.156 dB) - 0x77, // [005] = -2.500 dB -> AKM(0x77) = -2.557 dB error(-0.057 dB) - 0x76, // [006] = -3.000 dB -> AKM(0x76) = -2.937 dB error(+0.063 dB) - 0x75, // [007] = -3.500 dB -> AKM(0x75) = -3.334 dB error(+0.166 dB) - 0x73, // [008] = -4.000 dB -> AKM(0x73) = -4.188 dB error(-0.188 dB) - 0x72, // [009] = -4.500 dB -> AKM(0x72) = -4.648 dB error(-0.148 dB) - 0x71, // [010] = -5.000 dB -> AKM(0x71) = -5.134 dB error(-0.134 dB) - 0x70, // [011] = -5.500 dB -> AKM(0x70) = -5.649 dB error(-0.149 dB) - 0x6f, // [012] = -6.000 dB -> AKM(0x6f) = -6.056 dB error(-0.056 dB) - 0x6d, // [013] = -6.500 dB -> AKM(0x6d) = -6.631 dB error(-0.131 dB) - 0x6c, // [014] = -7.000 dB -> AKM(0x6c) = -6.933 dB error(+0.067 dB) - 0x6a, // [015] = -7.500 dB -> AKM(0x6a) = -7.571 dB error(-0.071 dB) - 0x69, // [016] = -8.000 dB -> AKM(0x69) = -7.909 dB error(+0.091 dB) - 0x67, // [017] = -8.500 dB -> AKM(0x67) = -8.626 dB error(-0.126 dB) - 0x66, // [018] = -9.000 dB -> AKM(0x66) = -9.008 dB error(-0.008 dB) - 0x65, // [019] = -9.500 dB -> AKM(0x65) = -9.407 dB error(+0.093 dB) - 0x64, // [020] = -10.000 dB -> AKM(0x64) = -9.826 dB error(+0.174 dB) - 0x62, // [021] = -10.500 dB -> AKM(0x62) = -10.730 dB error(-0.230 dB) - 0x61, // [022] = -11.000 dB -> AKM(0x61) = -11.219 dB error(-0.219 dB) - 0x60, // [023] = -11.500 dB -> AKM(0x60) = -11.738 dB error(-0.238 dB) - 0x5f, // [024] = -12.000 dB -> AKM(0x5f) = -12.149 dB error(-0.149 dB) - 0x5e, // [025] = -12.500 dB -> AKM(0x5e) = -12.434 dB error(+0.066 dB) - 0x5c, // [026] = -13.000 dB -> AKM(0x5c) = -13.033 dB error(-0.033 dB) - 0x5b, // [027] = -13.500 dB -> AKM(0x5b) = -13.350 dB error(+0.150 dB) - 0x59, // [028] = -14.000 dB -> AKM(0x59) = -14.018 dB error(-0.018 dB) - 0x58, // [029] = -14.500 dB -> AKM(0x58) = -14.373 dB error(+0.127 dB) - 0x56, // [030] = -15.000 dB -> AKM(0x56) = -15.130 dB error(-0.130 dB) - 0x55, // [031] = -15.500 dB -> AKM(0x55) = -15.534 dB error(-0.034 dB) - 0x54, // [032] = -16.000 dB -> AKM(0x54) = -15.958 dB error(+0.042 dB) - 0x53, // [033] = -16.500 dB -> AKM(0x53) = -16.404 dB error(+0.096 dB) - 0x52, // [034] = -17.000 dB -> AKM(0x52) = -16.874 dB error(+0.126 dB) - 0x51, // [035] = -17.500 dB -> AKM(0x51) = -17.371 dB error(+0.129 dB) - 0x50, // [036] = -18.000 dB -> AKM(0x50) = -17.898 dB error(+0.102 dB) - 0x4e, // [037] = -18.500 dB -> AKM(0x4e) = -18.605 dB error(-0.105 dB) - 0x4d, // [038] = -19.000 dB -> AKM(0x4d) = -18.905 dB error(+0.095 dB) - 0x4b, // [039] = -19.500 dB -> AKM(0x4b) = -19.538 dB error(-0.038 dB) - 0x4a, // [040] = -20.000 dB -> AKM(0x4a) = -19.872 dB error(+0.128 dB) - 0x48, // [041] = -20.500 dB -> AKM(0x48) = -20.583 dB error(-0.083 dB) - 0x47, // [042] = -21.000 dB -> AKM(0x47) = -20.961 dB error(+0.039 dB) - 0x46, // [043] = -21.500 dB -> AKM(0x46) = -21.356 dB error(+0.144 dB) - 0x44, // [044] = -22.000 dB -> AKM(0x44) = -22.206 dB error(-0.206 dB) - 0x43, // [045] = -22.500 dB -> AKM(0x43) = -22.664 dB error(-0.164 dB) - 0x42, // [046] = -23.000 dB -> AKM(0x42) = -23.147 dB error(-0.147 dB) - 0x41, // [047] = -23.500 dB -> AKM(0x41) = -23.659 dB error(-0.159 dB) - 0x40, // [048] = -24.000 dB -> AKM(0x40) = -24.203 dB error(-0.203 dB) - 0x3f, // [049] = -24.500 dB -> AKM(0x3f) = -24.635 dB error(-0.135 dB) - 0x3e, // [050] = -25.000 dB -> AKM(0x3e) = -24.935 dB error(+0.065 dB) - 0x3c, // [051] = -25.500 dB -> AKM(0x3c) = -25.569 dB error(-0.069 dB) - 0x3b, // [052] = -26.000 dB -> AKM(0x3b) = -25.904 dB error(+0.096 dB) - 0x39, // [053] = -26.500 dB -> AKM(0x39) = -26.615 dB error(-0.115 dB) - 0x38, // [054] = -27.000 dB -> AKM(0x38) = -26.994 dB error(+0.006 dB) - 0x37, // [055] = -27.500 dB -> AKM(0x37) = -27.390 dB error(+0.110 dB) - 0x36, // [056] = -28.000 dB -> AKM(0x36) = -27.804 dB error(+0.196 dB) - 0x34, // [057] = -28.500 dB -> AKM(0x34) = -28.699 dB error(-0.199 dB) - 0x33, // [058] = -29.000 dB -> AKM(0x33) = -29.183 dB error(-0.183 dB) - 0x32, // [059] = -29.500 dB -> AKM(0x32) = -29.696 dB error(-0.196 dB) - 0x31, // [060] = -30.000 dB -> AKM(0x31) = -30.241 dB error(-0.241 dB) - 0x31, // [061] = -30.500 dB -> AKM(0x31) = -30.241 dB error(+0.259 dB) - 0x30, // [062] = -31.000 dB -> AKM(0x30) = -30.823 dB error(+0.177 dB) - 0x2e, // [063] = -31.500 dB -> AKM(0x2e) = -31.610 dB error(-0.110 dB) - 0x2d, // [064] = -32.000 dB -> AKM(0x2d) = -31.945 dB error(+0.055 dB) - 0x2b, // [065] = -32.500 dB -> AKM(0x2b) = -32.659 dB error(-0.159 dB) - 0x2a, // [066] = -33.000 dB -> AKM(0x2a) = -33.038 dB error(-0.038 dB) - 0x29, // [067] = -33.500 dB -> AKM(0x29) = -33.435 dB error(+0.065 dB) - 0x28, // [068] = -34.000 dB -> AKM(0x28) = -33.852 dB error(+0.148 dB) - 0x27, // [069] = -34.500 dB -> AKM(0x27) = -34.289 dB error(+0.211 dB) - 0x25, // [070] = -35.000 dB -> AKM(0x25) = -35.235 dB error(-0.235 dB) - 0x24, // [071] = -35.500 dB -> AKM(0x24) = -35.750 dB error(-0.250 dB) - 0x24, // [072] = -36.000 dB -> AKM(0x24) = -35.750 dB error(+0.250 dB) - 0x23, // [073] = -36.500 dB -> AKM(0x23) = -36.297 dB error(+0.203 dB) - 0x22, // [074] = -37.000 dB -> AKM(0x22) = -36.881 dB error(+0.119 dB) - 0x21, // [075] = -37.500 dB -> AKM(0x21) = -37.508 dB error(-0.008 dB) - 0x20, // [076] = -38.000 dB -> AKM(0x20) = -38.183 dB error(-0.183 dB) - 0x1f, // [077] = -38.500 dB -> AKM(0x1f) = -38.726 dB error(-0.226 dB) - 0x1e, // [078] = -39.000 dB -> AKM(0x1e) = -39.108 dB error(-0.108 dB) - 0x1d, // [079] = -39.500 dB -> AKM(0x1d) = -39.507 dB error(-0.007 dB) - 0x1c, // [080] = -40.000 dB -> AKM(0x1c) = -39.926 dB error(+0.074 dB) - 0x1b, // [081] = -40.500 dB -> AKM(0x1b) = -40.366 dB error(+0.134 dB) - 0x1a, // [082] = -41.000 dB -> AKM(0x1a) = -40.829 dB error(+0.171 dB) - 0x19, // [083] = -41.500 dB -> AKM(0x19) = -41.318 dB error(+0.182 dB) - 0x18, // [084] = -42.000 dB -> AKM(0x18) = -41.837 dB error(+0.163 dB) - 0x17, // [085] = -42.500 dB -> AKM(0x17) = -42.389 dB error(+0.111 dB) - 0x16, // [086] = -43.000 dB -> AKM(0x16) = -42.978 dB error(+0.022 dB) - 0x15, // [087] = -43.500 dB -> AKM(0x15) = -43.610 dB error(-0.110 dB) - 0x14, // [088] = -44.000 dB -> AKM(0x14) = -44.291 dB error(-0.291 dB) - 0x14, // [089] = -44.500 dB -> AKM(0x14) = -44.291 dB error(+0.209 dB) - 0x13, // [090] = -45.000 dB -> AKM(0x13) = -45.031 dB error(-0.031 dB) - 0x12, // [091] = -45.500 dB -> AKM(0x12) = -45.840 dB error(-0.340 dB) - 0x12, // [092] = -46.000 dB -> AKM(0x12) = -45.840 dB error(+0.160 dB) - 0x11, // [093] = -46.500 dB -> AKM(0x11) = -46.731 dB error(-0.231 dB) - 0x11, // [094] = -47.000 dB -> AKM(0x11) = -46.731 dB error(+0.269 dB) - 0x10, // [095] = -47.500 dB -> AKM(0x10) = -47.725 dB error(-0.225 dB) - 0x10, // [096] = -48.000 dB -> AKM(0x10) = -47.725 dB error(+0.275 dB) - 0x0f, // [097] = -48.500 dB -> AKM(0x0f) = -48.553 dB error(-0.053 dB) - 0x0e, // [098] = -49.000 dB -> AKM(0x0e) = -49.152 dB error(-0.152 dB) - 0x0d, // [099] = -49.500 dB -> AKM(0x0d) = -49.796 dB error(-0.296 dB) - 0x0d, // [100] = -50.000 dB -> AKM(0x0d) = -49.796 dB error(+0.204 dB) - 0x0c, // [101] = -50.500 dB -> AKM(0x0c) = -50.491 dB error(+0.009 dB) - 0x0b, // [102] = -51.000 dB -> AKM(0x0b) = -51.247 dB error(-0.247 dB) - 0x0b, // [103] = -51.500 dB -> AKM(0x0b) = -51.247 dB error(+0.253 dB) - 0x0a, // [104] = -52.000 dB -> AKM(0x0a) = -52.075 dB error(-0.075 dB) - 0x0a, // [105] = -52.500 dB -> AKM(0x0a) = -52.075 dB error(+0.425 dB) - 0x09, // [106] = -53.000 dB -> AKM(0x09) = -52.990 dB error(+0.010 dB) - 0x09, // [107] = -53.500 dB -> AKM(0x09) = -52.990 dB error(+0.510 dB) - 0x08, // [108] = -54.000 dB -> AKM(0x08) = -54.013 dB error(-0.013 dB) - 0x08, // [109] = -54.500 dB -> AKM(0x08) = -54.013 dB error(+0.487 dB) - 0x07, // [110] = -55.000 dB -> AKM(0x07) = -55.173 dB error(-0.173 dB) - 0x07, // [111] = -55.500 dB -> AKM(0x07) = -55.173 dB error(+0.327 dB) - 0x06, // [112] = -56.000 dB -> AKM(0x06) = -56.512 dB error(-0.512 dB) - 0x06, // [113] = -56.500 dB -> AKM(0x06) = -56.512 dB error(-0.012 dB) - 0x06, // [114] = -57.000 dB -> AKM(0x06) = -56.512 dB error(+0.488 dB) - 0x05, // [115] = -57.500 dB -> AKM(0x05) = -58.095 dB error(-0.595 dB) - 0x05, // [116] = -58.000 dB -> AKM(0x05) = -58.095 dB error(-0.095 dB) - 0x05, // [117] = -58.500 dB -> AKM(0x05) = -58.095 dB error(+0.405 dB) - 0x05, // [118] = -59.000 dB -> AKM(0x05) = -58.095 dB error(+0.905 dB) - 0x04, // [119] = -59.500 dB -> AKM(0x04) = -60.034 dB error(-0.534 dB) - 0x04, // [120] = -60.000 dB -> AKM(0x04) = -60.034 dB error(-0.034 dB) - 0x04, // [121] = -60.500 dB -> AKM(0x04) = -60.034 dB error(+0.466 dB) - 0x04, // [122] = -61.000 dB -> AKM(0x04) = -60.034 dB error(+0.966 dB) - 0x03, // [123] = -61.500 dB -> AKM(0x03) = -62.532 dB error(-1.032 dB) - 0x03, // [124] = -62.000 dB -> AKM(0x03) = -62.532 dB error(-0.532 dB) - 0x03, // [125] = -62.500 dB -> AKM(0x03) = -62.532 dB error(-0.032 dB) - 0x03, // [126] = -63.000 dB -> AKM(0x03) = -62.532 dB error(+0.468 dB) - 0x03, // [127] = -63.500 dB -> AKM(0x03) = -62.532 dB error(+0.968 dB) - 0x03, // [128] = -64.000 dB -> AKM(0x03) = -62.532 dB error(+1.468 dB) - 0x02, // [129] = -64.500 dB -> AKM(0x02) = -66.054 dB error(-1.554 dB) - 0x02, // [130] = -65.000 dB -> AKM(0x02) = -66.054 dB error(-1.054 dB) - 0x02, // [131] = -65.500 dB -> AKM(0x02) = -66.054 dB error(-0.554 dB) - 0x02, // [132] = -66.000 dB -> AKM(0x02) = -66.054 dB error(-0.054 dB) - 0x02, // [133] = -66.500 dB -> AKM(0x02) = -66.054 dB error(+0.446 dB) - 0x02, // [134] = -67.000 dB -> AKM(0x02) = -66.054 dB error(+0.946 dB) - 0x02, // [135] = -67.500 dB -> AKM(0x02) = -66.054 dB error(+1.446 dB) - 0x02, // [136] = -68.000 dB -> AKM(0x02) = -66.054 dB error(+1.946 dB) - 0x02, // [137] = -68.500 dB -> AKM(0x02) = -66.054 dB error(+2.446 dB) - 0x02, // [138] = -69.000 dB -> AKM(0x02) = -66.054 dB error(+2.946 dB) - 0x01, // [139] = -69.500 dB -> AKM(0x01) = -72.075 dB error(-2.575 dB) - 0x01, // [140] = -70.000 dB -> AKM(0x01) = -72.075 dB error(-2.075 dB) - 0x01, // [141] = -70.500 dB -> AKM(0x01) = -72.075 dB error(-1.575 dB) - 0x01, // [142] = -71.000 dB -> AKM(0x01) = -72.075 dB error(-1.075 dB) - 0x01, // [143] = -71.500 dB -> AKM(0x01) = -72.075 dB error(-0.575 dB) - 0x01, // [144] = -72.000 dB -> AKM(0x01) = -72.075 dB error(-0.075 dB) - 0x01, // [145] = -72.500 dB -> AKM(0x01) = -72.075 dB error(+0.425 dB) - 0x01, // [146] = -73.000 dB -> AKM(0x01) = -72.075 dB error(+0.925 dB) - 0x00}; // [147] = -73.500 dB -> AKM(0x00) = mute error(+infini) + 0x7f, /* [000] = +0.000 dB -> AKM(0x7f) = +0.000 dB error(+0.000 dB) */ + 0x7d, /* [001] = -0.500 dB -> AKM(0x7d) = -0.572 dB error(-0.072 dB) */ + 0x7c, /* [002] = -1.000 dB -> AKM(0x7c) = -0.873 dB error(+0.127 dB) */ + 0x7a, /* [003] = -1.500 dB -> AKM(0x7a) = -1.508 dB error(-0.008 dB) */ + 0x79, /* [004] = -2.000 dB -> AKM(0x79) = -1.844 dB error(+0.156 dB) */ + 0x77, /* [005] = -2.500 dB -> AKM(0x77) = -2.557 dB error(-0.057 dB) */ + 0x76, /* [006] = -3.000 dB -> AKM(0x76) = -2.937 dB error(+0.063 dB) */ + 0x75, /* [007] = -3.500 dB -> AKM(0x75) = -3.334 dB error(+0.166 dB) */ + 0x73, /* [008] = -4.000 dB -> AKM(0x73) = -4.188 dB error(-0.188 dB) */ + 0x72, /* [009] = -4.500 dB -> AKM(0x72) = -4.648 dB error(-0.148 dB) */ + 0x71, /* [010] = -5.000 dB -> AKM(0x71) = -5.134 dB error(-0.134 dB) */ + 0x70, /* [011] = -5.500 dB -> AKM(0x70) = -5.649 dB error(-0.149 dB) */ + 0x6f, /* [012] = -6.000 dB -> AKM(0x6f) = -6.056 dB error(-0.056 dB) */ + 0x6d, /* [013] = -6.500 dB -> AKM(0x6d) = -6.631 dB error(-0.131 dB) */ + 0x6c, /* [014] = -7.000 dB -> AKM(0x6c) = -6.933 dB error(+0.067 dB) */ + 0x6a, /* [015] = -7.500 dB -> AKM(0x6a) = -7.571 dB error(-0.071 dB) */ + 0x69, /* [016] = -8.000 dB -> AKM(0x69) = -7.909 dB error(+0.091 dB) */ + 0x67, /* [017] = -8.500 dB -> AKM(0x67) = -8.626 dB error(-0.126 dB) */ + 0x66, /* [018] = -9.000 dB -> AKM(0x66) = -9.008 dB error(-0.008 dB) */ + 0x65, /* [019] = -9.500 dB -> AKM(0x65) = -9.407 dB error(+0.093 dB) */ + 0x64, /* [020] = -10.000 dB -> AKM(0x64) = -9.826 dB error(+0.174 dB) */ + 0x62, /* [021] = -10.500 dB -> AKM(0x62) = -10.730 dB error(-0.230 dB) */ + 0x61, /* [022] = -11.000 dB -> AKM(0x61) = -11.219 dB error(-0.219 dB) */ + 0x60, /* [023] = -11.500 dB -> AKM(0x60) = -11.738 dB error(-0.238 dB) */ + 0x5f, /* [024] = -12.000 dB -> AKM(0x5f) = -12.149 dB error(-0.149 dB) */ + 0x5e, /* [025] = -12.500 dB -> AKM(0x5e) = -12.434 dB error(+0.066 dB) */ + 0x5c, /* [026] = -13.000 dB -> AKM(0x5c) = -13.033 dB error(-0.033 dB) */ + 0x5b, /* [027] = -13.500 dB -> AKM(0x5b) = -13.350 dB error(+0.150 dB) */ + 0x59, /* [028] = -14.000 dB -> AKM(0x59) = -14.018 dB error(-0.018 dB) */ + 0x58, /* [029] = -14.500 dB -> AKM(0x58) = -14.373 dB error(+0.127 dB) */ + 0x56, /* [030] = -15.000 dB -> AKM(0x56) = -15.130 dB error(-0.130 dB) */ + 0x55, /* [031] = -15.500 dB -> AKM(0x55) = -15.534 dB error(-0.034 dB) */ + 0x54, /* [032] = -16.000 dB -> AKM(0x54) = -15.958 dB error(+0.042 dB) */ + 0x53, /* [033] = -16.500 dB -> AKM(0x53) = -16.404 dB error(+0.096 dB) */ + 0x52, /* [034] = -17.000 dB -> AKM(0x52) = -16.874 dB error(+0.126 dB) */ + 0x51, /* [035] = -17.500 dB -> AKM(0x51) = -17.371 dB error(+0.129 dB) */ + 0x50, /* [036] = -18.000 dB -> AKM(0x50) = -17.898 dB error(+0.102 dB) */ + 0x4e, /* [037] = -18.500 dB -> AKM(0x4e) = -18.605 dB error(-0.105 dB) */ + 0x4d, /* [038] = -19.000 dB -> AKM(0x4d) = -18.905 dB error(+0.095 dB) */ + 0x4b, /* [039] = -19.500 dB -> AKM(0x4b) = -19.538 dB error(-0.038 dB) */ + 0x4a, /* [040] = -20.000 dB -> AKM(0x4a) = -19.872 dB error(+0.128 dB) */ + 0x48, /* [041] = -20.500 dB -> AKM(0x48) = -20.583 dB error(-0.083 dB) */ + 0x47, /* [042] = -21.000 dB -> AKM(0x47) = -20.961 dB error(+0.039 dB) */ + 0x46, /* [043] = -21.500 dB -> AKM(0x46) = -21.356 dB error(+0.144 dB) */ + 0x44, /* [044] = -22.000 dB -> AKM(0x44) = -22.206 dB error(-0.206 dB) */ + 0x43, /* [045] = -22.500 dB -> AKM(0x43) = -22.664 dB error(-0.164 dB) */ + 0x42, /* [046] = -23.000 dB -> AKM(0x42) = -23.147 dB error(-0.147 dB) */ + 0x41, /* [047] = -23.500 dB -> AKM(0x41) = -23.659 dB error(-0.159 dB) */ + 0x40, /* [048] = -24.000 dB -> AKM(0x40) = -24.203 dB error(-0.203 dB) */ + 0x3f, /* [049] = -24.500 dB -> AKM(0x3f) = -24.635 dB error(-0.135 dB) */ + 0x3e, /* [050] = -25.000 dB -> AKM(0x3e) = -24.935 dB error(+0.065 dB) */ + 0x3c, /* [051] = -25.500 dB -> AKM(0x3c) = -25.569 dB error(-0.069 dB) */ + 0x3b, /* [052] = -26.000 dB -> AKM(0x3b) = -25.904 dB error(+0.096 dB) */ + 0x39, /* [053] = -26.500 dB -> AKM(0x39) = -26.615 dB error(-0.115 dB) */ + 0x38, /* [054] = -27.000 dB -> AKM(0x38) = -26.994 dB error(+0.006 dB) */ + 0x37, /* [055] = -27.500 dB -> AKM(0x37) = -27.390 dB error(+0.110 dB) */ + 0x36, /* [056] = -28.000 dB -> AKM(0x36) = -27.804 dB error(+0.196 dB) */ + 0x34, /* [057] = -28.500 dB -> AKM(0x34) = -28.699 dB error(-0.199 dB) */ + 0x33, /* [058] = -29.000 dB -> AKM(0x33) = -29.183 dB error(-0.183 dB) */ + 0x32, /* [059] = -29.500 dB -> AKM(0x32) = -29.696 dB error(-0.196 dB) */ + 0x31, /* [060] = -30.000 dB -> AKM(0x31) = -30.241 dB error(-0.241 dB) */ + 0x31, /* [061] = -30.500 dB -> AKM(0x31) = -30.241 dB error(+0.259 dB) */ + 0x30, /* [062] = -31.000 dB -> AKM(0x30) = -30.823 dB error(+0.177 dB) */ + 0x2e, /* [063] = -31.500 dB -> AKM(0x2e) = -31.610 dB error(-0.110 dB) */ + 0x2d, /* [064] = -32.000 dB -> AKM(0x2d) = -31.945 dB error(+0.055 dB) */ + 0x2b, /* [065] = -32.500 dB -> AKM(0x2b) = -32.659 dB error(-0.159 dB) */ + 0x2a, /* [066] = -33.000 dB -> AKM(0x2a) = -33.038 dB error(-0.038 dB) */ + 0x29, /* [067] = -33.500 dB -> AKM(0x29) = -33.435 dB error(+0.065 dB) */ + 0x28, /* [068] = -34.000 dB -> AKM(0x28) = -33.852 dB error(+0.148 dB) */ + 0x27, /* [069] = -34.500 dB -> AKM(0x27) = -34.289 dB error(+0.211 dB) */ + 0x25, /* [070] = -35.000 dB -> AKM(0x25) = -35.235 dB error(-0.235 dB) */ + 0x24, /* [071] = -35.500 dB -> AKM(0x24) = -35.750 dB error(-0.250 dB) */ + 0x24, /* [072] = -36.000 dB -> AKM(0x24) = -35.750 dB error(+0.250 dB) */ + 0x23, /* [073] = -36.500 dB -> AKM(0x23) = -36.297 dB error(+0.203 dB) */ + 0x22, /* [074] = -37.000 dB -> AKM(0x22) = -36.881 dB error(+0.119 dB) */ + 0x21, /* [075] = -37.500 dB -> AKM(0x21) = -37.508 dB error(-0.008 dB) */ + 0x20, /* [076] = -38.000 dB -> AKM(0x20) = -38.183 dB error(-0.183 dB) */ + 0x1f, /* [077] = -38.500 dB -> AKM(0x1f) = -38.726 dB error(-0.226 dB) */ + 0x1e, /* [078] = -39.000 dB -> AKM(0x1e) = -39.108 dB error(-0.108 dB) */ + 0x1d, /* [079] = -39.500 dB -> AKM(0x1d) = -39.507 dB error(-0.007 dB) */ + 0x1c, /* [080] = -40.000 dB -> AKM(0x1c) = -39.926 dB error(+0.074 dB) */ + 0x1b, /* [081] = -40.500 dB -> AKM(0x1b) = -40.366 dB error(+0.134 dB) */ + 0x1a, /* [082] = -41.000 dB -> AKM(0x1a) = -40.829 dB error(+0.171 dB) */ + 0x19, /* [083] = -41.500 dB -> AKM(0x19) = -41.318 dB error(+0.182 dB) */ + 0x18, /* [084] = -42.000 dB -> AKM(0x18) = -41.837 dB error(+0.163 dB) */ + 0x17, /* [085] = -42.500 dB -> AKM(0x17) = -42.389 dB error(+0.111 dB) */ + 0x16, /* [086] = -43.000 dB -> AKM(0x16) = -42.978 dB error(+0.022 dB) */ + 0x15, /* [087] = -43.500 dB -> AKM(0x15) = -43.610 dB error(-0.110 dB) */ + 0x14, /* [088] = -44.000 dB -> AKM(0x14) = -44.291 dB error(-0.291 dB) */ + 0x14, /* [089] = -44.500 dB -> AKM(0x14) = -44.291 dB error(+0.209 dB) */ + 0x13, /* [090] = -45.000 dB -> AKM(0x13) = -45.031 dB error(-0.031 dB) */ + 0x12, /* [091] = -45.500 dB -> AKM(0x12) = -45.840 dB error(-0.340 dB) */ + 0x12, /* [092] = -46.000 dB -> AKM(0x12) = -45.840 dB error(+0.160 dB) */ + 0x11, /* [093] = -46.500 dB -> AKM(0x11) = -46.731 dB error(-0.231 dB) */ + 0x11, /* [094] = -47.000 dB -> AKM(0x11) = -46.731 dB error(+0.269 dB) */ + 0x10, /* [095] = -47.500 dB -> AKM(0x10) = -47.725 dB error(-0.225 dB) */ + 0x10, /* [096] = -48.000 dB -> AKM(0x10) = -47.725 dB error(+0.275 dB) */ + 0x0f, /* [097] = -48.500 dB -> AKM(0x0f) = -48.553 dB error(-0.053 dB) */ + 0x0e, /* [098] = -49.000 dB -> AKM(0x0e) = -49.152 dB error(-0.152 dB) */ + 0x0d, /* [099] = -49.500 dB -> AKM(0x0d) = -49.796 dB error(-0.296 dB) */ + 0x0d, /* [100] = -50.000 dB -> AKM(0x0d) = -49.796 dB error(+0.204 dB) */ + 0x0c, /* [101] = -50.500 dB -> AKM(0x0c) = -50.491 dB error(+0.009 dB) */ + 0x0b, /* [102] = -51.000 dB -> AKM(0x0b) = -51.247 dB error(-0.247 dB) */ + 0x0b, /* [103] = -51.500 dB -> AKM(0x0b) = -51.247 dB error(+0.253 dB) */ + 0x0a, /* [104] = -52.000 dB -> AKM(0x0a) = -52.075 dB error(-0.075 dB) */ + 0x0a, /* [105] = -52.500 dB -> AKM(0x0a) = -52.075 dB error(+0.425 dB) */ + 0x09, /* [106] = -53.000 dB -> AKM(0x09) = -52.990 dB error(+0.010 dB) */ + 0x09, /* [107] = -53.500 dB -> AKM(0x09) = -52.990 dB error(+0.510 dB) */ + 0x08, /* [108] = -54.000 dB -> AKM(0x08) = -54.013 dB error(-0.013 dB) */ + 0x08, /* [109] = -54.500 dB -> AKM(0x08) = -54.013 dB error(+0.487 dB) */ + 0x07, /* [110] = -55.000 dB -> AKM(0x07) = -55.173 dB error(-0.173 dB) */ + 0x07, /* [111] = -55.500 dB -> AKM(0x07) = -55.173 dB error(+0.327 dB) */ + 0x06, /* [112] = -56.000 dB -> AKM(0x06) = -56.512 dB error(-0.512 dB) */ + 0x06, /* [113] = -56.500 dB -> AKM(0x06) = -56.512 dB error(-0.012 dB) */ + 0x06, /* [114] = -57.000 dB -> AKM(0x06) = -56.512 dB error(+0.488 dB) */ + 0x05, /* [115] = -57.500 dB -> AKM(0x05) = -58.095 dB error(-0.595 dB) */ + 0x05, /* [116] = -58.000 dB -> AKM(0x05) = -58.095 dB error(-0.095 dB) */ + 0x05, /* [117] = -58.500 dB -> AKM(0x05) = -58.095 dB error(+0.405 dB) */ + 0x05, /* [118] = -59.000 dB -> AKM(0x05) = -58.095 dB error(+0.905 dB) */ + 0x04, /* [119] = -59.500 dB -> AKM(0x04) = -60.034 dB error(-0.534 dB) */ + 0x04, /* [120] = -60.000 dB -> AKM(0x04) = -60.034 dB error(-0.034 dB) */ + 0x04, /* [121] = -60.500 dB -> AKM(0x04) = -60.034 dB error(+0.466 dB) */ + 0x04, /* [122] = -61.000 dB -> AKM(0x04) = -60.034 dB error(+0.966 dB) */ + 0x03, /* [123] = -61.500 dB -> AKM(0x03) = -62.532 dB error(-1.032 dB) */ + 0x03, /* [124] = -62.000 dB -> AKM(0x03) = -62.532 dB error(-0.532 dB) */ + 0x03, /* [125] = -62.500 dB -> AKM(0x03) = -62.532 dB error(-0.032 dB) */ + 0x03, /* [126] = -63.000 dB -> AKM(0x03) = -62.532 dB error(+0.468 dB) */ + 0x03, /* [127] = -63.500 dB -> AKM(0x03) = -62.532 dB error(+0.968 dB) */ + 0x03, /* [128] = -64.000 dB -> AKM(0x03) = -62.532 dB error(+1.468 dB) */ + 0x02, /* [129] = -64.500 dB -> AKM(0x02) = -66.054 dB error(-1.554 dB) */ + 0x02, /* [130] = -65.000 dB -> AKM(0x02) = -66.054 dB error(-1.054 dB) */ + 0x02, /* [131] = -65.500 dB -> AKM(0x02) = -66.054 dB error(-0.554 dB) */ + 0x02, /* [132] = -66.000 dB -> AKM(0x02) = -66.054 dB error(-0.054 dB) */ + 0x02, /* [133] = -66.500 dB -> AKM(0x02) = -66.054 dB error(+0.446 dB) */ + 0x02, /* [134] = -67.000 dB -> AKM(0x02) = -66.054 dB error(+0.946 dB) */ + 0x02, /* [135] = -67.500 dB -> AKM(0x02) = -66.054 dB error(+1.446 dB) */ + 0x02, /* [136] = -68.000 dB -> AKM(0x02) = -66.054 dB error(+1.946 dB) */ + 0x02, /* [137] = -68.500 dB -> AKM(0x02) = -66.054 dB error(+2.446 dB) */ + 0x02, /* [138] = -69.000 dB -> AKM(0x02) = -66.054 dB error(+2.946 dB) */ + 0x01, /* [139] = -69.500 dB -> AKM(0x01) = -72.075 dB error(-2.575 dB) */ + 0x01, /* [140] = -70.000 dB -> AKM(0x01) = -72.075 dB error(-2.075 dB) */ + 0x01, /* [141] = -70.500 dB -> AKM(0x01) = -72.075 dB error(-1.575 dB) */ + 0x01, /* [142] = -71.000 dB -> AKM(0x01) = -72.075 dB error(-1.075 dB) */ + 0x01, /* [143] = -71.500 dB -> AKM(0x01) = -72.075 dB error(-0.575 dB) */ + 0x01, /* [144] = -72.000 dB -> AKM(0x01) = -72.075 dB error(-0.075 dB) */ + 0x01, /* [145] = -72.500 dB -> AKM(0x01) = -72.075 dB error(+0.425 dB) */ + 0x01, /* [146] = -73.000 dB -> AKM(0x01) = -72.075 dB error(+0.925 dB) */ + 0x00}; /* [147] = -73.500 dB -> AKM(0x00) = mute error(+infini) */ /* * pseudo-codec write entry @@ -738,7 +738,7 @@ static void vx2_reset_codec(struct vx_core *_chip) msleep(5); /* additionnel wait time for AKM's */ vx2_write_codec_reg(_chip, AKM_CODEC_POWER_CONTROL_CMD); /* DAC power up, ADC power up, Vref power down */ - + vx2_write_codec_reg(_chip, AKM_CODEC_CLOCK_FORMAT_CMD); /* default */ vx2_write_codec_reg(_chip, AKM_CODEC_MUTE_CMD); /* Mute = ON ,Deemphasis = OFF */ vx2_write_codec_reg(_chip, AKM_CODEC_RESET_OFF_CMD); /* DAC and ADC normal operation */ @@ -798,7 +798,7 @@ static void vx2_reset_board(struct vx_core *_chip, int cold_reset) struct snd_vx222 *chip = (struct snd_vx222 *)_chip; /* initialize the register values */ - chip->regCDSP = VX_CDSP_CODEC_RESET_MASK | VX_CDSP_DSP_RESET_MASK ; + chip->regCDSP = VX_CDSP_CODEC_RESET_MASK | VX_CDSP_DSP_RESET_MASK; chip->regCFG = 0; }

1 0

[alsa-devel] [PATCH 22/24] sound/pci: coding style fixes: via82xx
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/via82xx.c | 76 ++++++++++++++++++++++---------------------- sound/pci/via82xx_modem.c | 60 ++++++++++++++++++------------------ 2 files changed, 68 insertions(+), 68 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1aafe95..8a84cb2 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -43,10 +43,10 @@ * routines for each chipset. * * Sep. 26, 2005 Karsten Wiese <annabellesgarden(a)yahoo.de> - * - Optimize position calculation for the 823x chips. + * - Optimize position calculation for the 823x chips. */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -146,7 +146,7 @@ module_param(enable, bool, 0444); #define VIA_REG_CTRL_TERMINATE 0x40 /* WO */ #define VIA_REG_CTRL_AUTOSTART 0x20 #define VIA_REG_CTRL_PAUSE 0x08 /* RW */ -#define VIA_REG_CTRL_INT_STOP 0x04 +#define VIA_REG_CTRL_INT_STOP 0x04 #define VIA_REG_CTRL_INT_EOL 0x02 #define VIA_REG_CTRL_INT_FLAG 0x01 #define VIA_REG_CTRL_RESET 0x01 /* RW - probably reset? undocumented */ @@ -310,7 +310,7 @@ DEFINE_VIA_REGSET(CAPTURE_8233, 0x60); struct snd_via_sg_table { unsigned int offset; unsigned int size; -} ; +}; #define VIA_TABLE_SIZE 255 #define VIA_MAX_BUFSIZE (1<<24) @@ -431,9 +431,9 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre &dev->table) < 0) return -ENOMEM; } - if (! dev->idx_table) { + if (!dev->idx_table) { dev->idx_table = kmalloc(sizeof(*dev->idx_table) * VIA_TABLE_SIZE, GFP_KERNEL); - if (! dev->idx_table) + if (!dev->idx_table) return -ENOMEM; } @@ -459,14 +459,14 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest); rest -= r; - if (! rest) { + if (!rest) { if (i == periods - 1) flag = VIA_TBL_BIT_EOL; /* buffer boundary */ else flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; @@ -502,17 +502,17 @@ static inline unsigned int snd_via82xx_codec_xread(struct via82xx *chip) { return inl(VIAREG(chip, AC97)); } - + static inline void snd_via82xx_codec_xwrite(struct via82xx *chip, unsigned int val) { outl(val, VIAREG(chip, AC97)); } - + static int snd_via82xx_codec_ready(struct via82xx *chip, int secondary) { unsigned int timeout = 1000; /* 1ms */ unsigned int val; - + while (timeout-- > 0) { udelay(1); if (!((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)) @@ -522,14 +522,14 @@ static int snd_via82xx_codec_ready(struct via82xx *chip, int secondary) secondary, snd_via82xx_codec_xread(chip)); return -EIO; } - + static int snd_via82xx_codec_valid(struct via82xx *chip, int secondary) { unsigned int timeout = 1000; /* 1ms */ unsigned int val, val1; unsigned int stat = !secondary ? VIA_REG_AC97_PRIMARY_VALID : VIA_REG_AC97_SECONDARY_VALID; - + while (timeout-- > 0) { val = snd_via82xx_codec_xread(chip); val1 = val & (VIA_REG_AC97_BUSY | stat); @@ -539,7 +539,7 @@ static int snd_via82xx_codec_valid(struct via82xx *chip, int secondary) } return -EIO; } - + static void snd_via82xx_codec_wait(struct snd_ac97 *ac97) { struct via82xx *chip = ac97->private_data; @@ -602,7 +602,7 @@ static void snd_via82xx_channel_reset(struct via82xx *chip, struct viadev *viade /* clear interrupts */ outb(0x03, VIADEV_REG(viadev, OFFSET_STATUS)); outb(0x00, VIADEV_REG(viadev, OFFSET_TYPE)); /* for via686 */ - // outl(0, VIADEV_REG(viadev, OFFSET_CURR_PTR)); + /* outl(0, VIADEV_REG(viadev, OFFSET_CURR_PTR)); */ viadev->lastpos = 0; viadev->hwptr_done = 0; } @@ -619,7 +619,7 @@ static irqreturn_t snd_via686_interrupt(int irq, void *dev_id) unsigned int i; status = inl(VIAREG(chip, SGD_SHADOW)); - if (! (status & chip->intr_mask)) { + if (!(status & chip->intr_mask)) { if (chip->rmidi) /* check mpu401 interrupt */ return snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); @@ -631,12 +631,12 @@ static irqreturn_t snd_via686_interrupt(int irq, void *dev_id) for (i = 0; i < chip->num_devs; i++) { struct viadev *viadev = &chip->devs[i]; unsigned char c_status = inb(VIADEV_REG(viadev, OFFSET_STATUS)); - if (! (c_status & (VIA_REG_STAT_EOL|VIA_REG_STAT_FLAG|VIA_REG_STAT_STOPPED))) + if (!(c_status & (VIA_REG_STAT_EOL|VIA_REG_STAT_FLAG|VIA_REG_STAT_STOPPED))) continue; if (viadev->substream && viadev->running) { /* * Update hwptr_done based on 'period elapsed' - * interrupts. We'll use it, when the chip returns 0 + * interrupts. We'll use it, when the chip returns 0 * for OFFSET_CURR_COUNT. */ if (c_status & VIA_REG_STAT_EOL) @@ -685,7 +685,7 @@ static irqreturn_t snd_via8233_interrupt(int irq, void *dev_id) if (substream && viadev->running) { /* * Update hwptr_done based on 'period elapsed' - * interrupts. We'll use it, when the chip returns 0 + * interrupts. We'll use it, when the chip returns 0 * for OFFSET_CURR_COUNT. */ if (c_status & VIA_REG_STAT_EOL) @@ -783,7 +783,7 @@ static inline unsigned int calc_linear_pos(struct viadev *viadev, unsigned int i (int)size, (int)count); res = viadev->lastpos; } else { - if (! count) { + if (!count) { /* Some mobos report count = 0 on the DMA boundary, * i.e. count = size indeed. * Let's check whether this step is above the expected size. @@ -855,7 +855,7 @@ static snd_pcm_uframes_t snd_via8233_pcm_pointer(struct snd_pcm_substream *subst struct viadev *viadev = substream->runtime->private_data; unsigned int idx, count, res; int status; - + if (snd_BUG_ON(!viadev->tbl_entries)) return 0; @@ -865,7 +865,7 @@ static snd_pcm_uframes_t snd_via8233_pcm_pointer(struct snd_pcm_substream *subst if (!status) status = inb(VIADEV_REG(viadev, OFFSET_STATUS)); - /* An apparent bug in the 8251 is worked around by sending a + /* An apparent bug in the 8251 is worked around by sending a * REG_CTRL_START. */ if (chip->revision == VIA_REV_8251 && (status & VIA_REG_STAT_EOL)) snd_via82xx_pcm_trigger(substream, SNDRV_PCM_TRIGGER_START); @@ -896,7 +896,7 @@ static snd_pcm_uframes_t snd_via8233_pcm_pointer(struct snd_pcm_substream *subst res += viadev->fragsize; } } - } + } unlock: viadev->lastpos = res; spin_unlock(&chip->reg_lock); @@ -1166,7 +1166,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, struct via_rate_lock *ratep; runtime->hw = snd_via82xx_hw; - + /* set the hw rate condition */ ratep = &chip->rates[viadev->direction]; spin_lock_irq(&ratep->lock); @@ -1185,7 +1185,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, SNDRV_PCM_RATE_8000_48000); runtime->hw.rate_min = 8000; runtime->hw.rate_max = 48000; - } else if (! ratep->rate) { + } else if (!ratep->rate) { int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC; runtime->hw.rates = chip->ac97->rates[idx]; snd_pcm_limit_hw_rates(runtime); @@ -1276,11 +1276,11 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream) ratep = &chip->rates[viadev->direction]; spin_lock_irq(&ratep->lock); ratep->used--; - if (! ratep->used) + if (!ratep->used) ratep->rate = 0; spin_unlock_irq(&ratep->lock); - if (! ratep->rate) { - if (! viadev->direction) { + if (!ratep->rate) { + if (!viadev->direction) { snd_ac97_update_power(chip->ac97, AC97_PCM_FRONT_DAC_RATE, 0); snd_ac97_update_power(chip->ac97, @@ -1457,7 +1457,7 @@ static int __devinit snd_via8233a_pcm_new(struct via82xx *chip) 64*1024, VIA_MAX_BUFSIZE); /* SPDIF supported? */ - if (! ac97_can_spdif(chip->ac97)) + if (!ac97_can_spdif(chip->ac97)) return 0; /* PCM #1: DXS3 playback (for spdif) */ @@ -1921,7 +1921,7 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) memset(&sid, 0, sizeof(sid)); strcpy(sid.name, "PCM Playback Volume"); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - if (! snd_ctl_find_id(chip->card, &sid)) { + if (!snd_ctl_find_id(chip->card, &sid)) { snd_printd(KERN_INFO "Using DXS as PCM Playback\n"); err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_pcmdxs_volume_control, chip)); if (err < 0) @@ -2026,7 +2026,7 @@ static void snd_via82xx_proc_read(struct snd_info_entry *entry, { struct via82xx *chip = entry->private_data; int i; - + snd_iprintf(buffer, "%s\n\n", chip->card->longname); for (i = 0; i < 0xa0; i += 4) { snd_iprintf(buffer, "%02x: %08x\n", i, inl(chip->port + i)); @@ -2037,7 +2037,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "via82xx", &entry)) + if (!snd_card_proc_new(chip->card, "via82xx", &entry)) snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read); } @@ -2057,7 +2057,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, 0); #endif pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); - if (! (pval & VIA_ACLINK_C00_READY)) { /* codec not ready? */ + if (!(pval & VIA_ACLINK_C00_READY)) { /* codec not ready? */ /* deassert ACLink reset, force SYNC */ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_ENABLE | @@ -2078,7 +2078,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_INIT); udelay(100); } - + /* Make sure VRA is enabled, in case we didn't do a * complete codec reset, above */ pci_read_config_byte(chip->pci, VIA_ACLINK_CTRL, &pval); @@ -2239,7 +2239,7 @@ static int snd_via82xx_free(struct via82xx *chip) if (chip->irq >= 0) free_irq(chip->irq, chip); - __end_hw: +__end_hw: release_and_free_resource(chip->mpu_res); pci_release_regions(chip->pci); @@ -2477,7 +2477,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, err = -EINVAL; goto __error; } - + if ((err = snd_via82xx_create(card, pci, chip_type, pci->revision, ac97_clock, &chip)) < 0) goto __error; @@ -2493,7 +2493,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, if (chip_type == TYPE_VIA8233A) { if ((err = snd_via8233a_pcm_new(chip)) < 0) goto __error; - // chip->dxs_fixed = 1; /* FIXME: use 48k for DXS #3? */ + /* chip->dxs_fixed = 1; FIXME: use 48k for DXS #3? */ } else { if ((err = snd_via8233_pcm_new(chip)) < 0) goto __error; @@ -2527,7 +2527,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); return 0; - __error: +__error: snd_card_free(card); return err; } diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 5bd79d2..833f56e 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -31,7 +31,7 @@ * modems. */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -90,7 +90,7 @@ module_param(enable, bool, 0444); #define VIA_REG_CTRL_TERMINATE 0x40 /* WO */ #define VIA_REG_CTRL_AUTOSTART 0x20 #define VIA_REG_CTRL_PAUSE 0x08 /* RW */ -#define VIA_REG_CTRL_INT_STOP 0x04 +#define VIA_REG_CTRL_INT_STOP 0x04 #define VIA_REG_CTRL_INT_EOL 0x02 #define VIA_REG_CTRL_INT_FLAG 0x01 #define VIA_REG_CTRL_RESET 0x01 /* RW - probably reset? undocumented */ @@ -212,7 +212,7 @@ DEFINE_VIA_REGSET(MI, 0x50); struct snd_via_sg_table { unsigned int offset; unsigned int size; -} ; +}; #define VIA_TABLE_SIZE 255 @@ -291,9 +291,9 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre &dev->table) < 0) return -ENOMEM; } - if (! dev->idx_table) { + if (!dev->idx_table) { dev->idx_table = kmalloc(sizeof(*dev->idx_table) * VIA_TABLE_SIZE, GFP_KERNEL); - if (! dev->idx_table) + if (!dev->idx_table) return -ENOMEM; } @@ -321,14 +321,14 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre if (rest < r) r = rest; rest -= r; - if (! rest) { + if (!rest) { if (i == periods - 1) flag = VIA_TBL_BIT_EOL; /* buffer boundary */ else flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; @@ -363,17 +363,17 @@ static inline unsigned int snd_via82xx_codec_xread(struct via82xx_modem *chip) { return inl(VIAREG(chip, AC97)); } - + static inline void snd_via82xx_codec_xwrite(struct via82xx_modem *chip, unsigned int val) { outl(val, VIAREG(chip, AC97)); } - + static int snd_via82xx_codec_ready(struct via82xx_modem *chip, int secondary) { unsigned int timeout = 1000; /* 1ms */ unsigned int val; - + while (timeout-- > 0) { udelay(1); if (!((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)) @@ -383,14 +383,14 @@ static int snd_via82xx_codec_ready(struct via82xx_modem *chip, int secondary) secondary, snd_via82xx_codec_xread(chip)); return -EIO; } - + static int snd_via82xx_codec_valid(struct via82xx_modem *chip, int secondary) { unsigned int timeout = 1000; /* 1ms */ unsigned int val, val1; unsigned int stat = !secondary ? VIA_REG_AC97_PRIMARY_VALID : VIA_REG_AC97_SECONDARY_VALID; - + while (timeout-- > 0) { val = snd_via82xx_codec_xread(chip); val1 = val & (VIA_REG_AC97_BUSY | stat); @@ -400,7 +400,7 @@ static int snd_via82xx_codec_valid(struct via82xx_modem *chip, int secondary) } return -EIO; } - + static void snd_via82xx_codec_wait(struct snd_ac97 *ac97) { struct via82xx_modem *chip = ac97->private_data; @@ -416,10 +416,10 @@ static void snd_via82xx_codec_write(struct snd_ac97 *ac97, { struct via82xx_modem *chip = ac97->private_data; unsigned int xval; - if(reg == AC97_GPIO_STATUS) { + if (reg == AC97_GPIO_STATUS) { outl(val, VIAREG(chip, GPI_STATUS)); return; - } + } xval = !ac97->num ? VIA_REG_AC97_CODEC_ID_PRIMARY : VIA_REG_AC97_CODEC_ID_SECONDARY; xval <<= VIA_REG_AC97_CODEC_ID_SHIFT; xval |= reg << VIA_REG_AC97_CMD_SHIFT; @@ -466,7 +466,7 @@ static void snd_via82xx_channel_reset(struct via82xx_modem *chip, struct viadev /* clear interrupts */ outb(0x03, VIADEV_REG(viadev, OFFSET_STATUS)); outb(0x00, VIADEV_REG(viadev, OFFSET_TYPE)); /* for via686 */ - // outl(0, VIADEV_REG(viadev, OFFSET_CURR_PTR)); + /* outl(0, VIADEV_REG(viadev, OFFSET_CURR_PTR)); */ viadev->lastpos = 0; } @@ -482,7 +482,7 @@ static irqreturn_t snd_via82xx_interrupt(int irq, void *dev_id) unsigned int i; status = inl(VIAREG(chip, SGD_SHADOW)); - if (! (status & chip->intr_mask)) { + if (!(status & chip->intr_mask)) { return IRQ_NONE; } // _skip_sgd: @@ -493,7 +493,7 @@ static irqreturn_t snd_via82xx_interrupt(int irq, void *dev_id) struct viadev *viadev = &chip->devs[i]; unsigned char c_status = inb(VIADEV_REG(viadev, OFFSET_STATUS)); c_status &= (VIA_REG_STAT_EOL|VIA_REG_STAT_FLAG|VIA_REG_STAT_STOPPED); - if (! c_status) + if (!c_status) continue; if (viadev->substream && viadev->running) { spin_unlock(&chip->reg_lock); @@ -583,7 +583,7 @@ static inline unsigned int calc_linear_pos(struct viadev *viadev, unsigned int i "using last valid pointer\n"); res = viadev->lastpos; } else { - if (! count) + if (!count) /* bogus count 0 on the DMA boundary? */ res = viadev->idx_table[idx].offset; else @@ -744,7 +744,7 @@ static int snd_via82xx_modem_pcm_open(struct via82xx_modem *chip, struct viadev }; runtime->hw = snd_via82xx_hw; - + if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) return err; @@ -918,7 +918,7 @@ static void snd_via82xx_proc_read(struct snd_info_entry *entry, struct snd_info_ { struct via82xx_modem *chip = entry->private_data; int i; - + snd_iprintf(buffer, "%s\n\n", chip->card->longname); for (i = 0; i < 0xa0; i += 4) { snd_iprintf(buffer, "%02x: %08x\n", i, inl(chip->port + i)); @@ -929,7 +929,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx_modem *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "via82xx", &entry)) + if (!snd_card_proc_new(chip->card, "via82xx", &entry)) snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read); } @@ -944,13 +944,13 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) unsigned char pval; pci_read_config_byte(chip->pci, VIA_MC97_CTRL, &pval); - if((pval & VIA_MC97_CTRL_INIT) != VIA_MC97_CTRL_INIT) { + if ((pval & VIA_MC97_CTRL_INIT) != VIA_MC97_CTRL_INIT) { pci_write_config_byte(chip->pci, 0x44, pval|VIA_MC97_CTRL_INIT); udelay(100); } pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); - if (! (pval & VIA_ACLINK_C00_READY)) { /* codec not ready? */ + if (!(pval & VIA_ACLINK_C00_READY)) { /* codec not ready? */ /* deassert ACLink reset, force SYNC */ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_ENABLE | @@ -970,7 +970,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_INIT); udelay(100); } - + pci_read_config_byte(chip->pci, VIA_ACLINK_CTRL, &pval); if ((pval & VIA_ACLINK_CTRL_INIT) != VIA_ACLINK_CTRL_INIT) { /* ACLink on, deassert ACLink reset, VSR, SGD data out */ @@ -1009,7 +1009,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) __ac97_ok2: /* route FM trap to IRQ, disable FM trap */ - // pci_write_config_byte(chip->pci, VIA_FM_NMI_CTRL, 0); + /* pci_write_config_byte(chip->pci, VIA_FM_NMI_CTRL, 0); */ /* disable all GPI interrupts */ outl(0, VIAREG(chip, GPI_INTR)); @@ -1078,7 +1078,7 @@ static int snd_via82xx_free(struct via82xx_modem *chip) for (i = 0; i < chip->num_devs; i++) snd_via82xx_channel_reset(chip, &chip->devs[i]); - __end_hw: +__end_hw: if (chip->irq >= 0) free_irq(chip->irq, chip); pci_release_regions(chip->pci); @@ -1182,7 +1182,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, err = -EINVAL; goto __error; } - + if ((err = snd_via82xx_create(card, pci, chip_type, pci->revision, ac97_clock, &chip)) < 0) goto __error; @@ -1190,7 +1190,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, if ((err = snd_via82xx_mixer_new(chip)) < 0) goto __error; - if ((err = snd_via686_pcm_new(chip)) < 0 ) + if ((err = snd_via686_pcm_new(chip)) < 0) goto __error; /* disable interrupts */ @@ -1209,7 +1209,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); return 0; - __error: +__error: snd_card_free(card); return err; }

1 0

[alsa-devel] [PATCH 21/24] sound/pci: coding style fixes: trident
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/trident/trident.c | 4 +- sound/pci/trident/trident_main.c | 478 ++++++++++++++++++------------------ sound/pci/trident/trident_memory.c | 12 +- 3 files changed, 247 insertions(+), 247 deletions(-) diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index d94b16f..11b49f7 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -63,9 +63,9 @@ module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); static struct pci_device_id snd_trident_ids[] = { - {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), + {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, - {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), + {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), 0, 0, 0}, {PCI_DEVICE(PCI_VENDOR_ID_SI, PCI_DEVICE_ID_SI_7018), 0, 0, 0}, { 0, } diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index c612b43..863f34f 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -43,7 +43,7 @@ #include <sound/trident.h> #include <sound/asoundef.h> -#include <asm/io.h> +#include <linux/io.h> static int snd_trident_pcm_mixer_build(struct snd_trident *trident, struct snd_trident_voice * voice, @@ -87,7 +87,7 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) printk("ESO: 0x%x\n", val >> 16); printk("Delta: 0x%x\n", val & 0xffff); val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); - } else { // TRIDENT_DEVICE_ID_NX + } else { /* TRIDENT_DEVICE_ID_NX */ val = inl(TRID_REG(trident, CH_NX_DELTA_CSO)); tmp = (val >> 24) & 0xff; printk("CSO: 0x%x\n", val & 0x00ffffff); @@ -107,15 +107,15 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) /*--------------------------------------------------------------------------- unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned short reg) - + Description: This routine will do all of the reading from the external CODEC (AC97). - + Parameters: ac97 - ac97 codec structure reg - CODEC register index, from AC97 Hal. - + returns: 16 bit value read from the AC97. - + ---------------------------------------------------------------------------*/ static unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned short reg) { @@ -167,16 +167,16 @@ static unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned sho /*--------------------------------------------------------------------------- void snd_trident_codec_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short wdata) - + Description: This routine will do all of the writing to the external CODEC (AC97). - + Parameters: ac97 - ac97 codec structure reg - CODEC register index, from AC97 Hal. data - Lower 16 bits are the data to write to CODEC. - + returns: TRUE if everything went ok, else FALSE. - + ---------------------------------------------------------------------------*/ static void snd_trident_codec_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short wdata) @@ -236,14 +236,14 @@ static void snd_trident_codec_write(struct snd_ac97 *ac97, unsigned short reg, /*--------------------------------------------------------------------------- void snd_trident_enable_eso(struct snd_trident *trident) - + Description: This routine will enable end of loop interrupts. End of loop interrupts will occur when a running channel reaches ESO. Also enables middle of loop interrupts. - + Parameters: trident - pointer to target device class for 4DWave. - + ---------------------------------------------------------------------------*/ static void snd_trident_enable_eso(struct snd_trident * trident) @@ -260,17 +260,17 @@ static void snd_trident_enable_eso(struct snd_trident * trident) /*--------------------------------------------------------------------------- void snd_trident_disable_eso(struct snd_trident *trident) - + Description: This routine will disable end of loop interrupts. End of loop interrupts will occur when a running channel reaches ESO. Also disables middle of loop interrupts. - - Parameters: + + Parameters: trident - pointer to target device class for 4DWave. - + returns: TRUE if everything went ok, else FALSE. - + ---------------------------------------------------------------------------*/ static void snd_trident_disable_eso(struct snd_trident * trident) @@ -333,13 +333,13 @@ EXPORT_SYMBOL(snd_trident_stop_voice); /*--------------------------------------------------------------------------- int snd_trident_allocate_pcm_channel(struct snd_trident *trident) - + Description: Allocate hardware channel in Bank B (32-63). - + Parameters : trident - pointer to target device class for 4DWave. - + Return Value: hardware channel - 32-63 or -1 when no channel is available - + ---------------------------------------------------------------------------*/ static int snd_trident_allocate_pcm_channel(struct snd_trident * trident) @@ -360,14 +360,14 @@ static int snd_trident_allocate_pcm_channel(struct snd_trident * trident) /*--------------------------------------------------------------------------- void snd_trident_free_pcm_channel(int channel) - + Description: Free hardware channel in Bank B (32-63) - + Parameters : trident - pointer to target device class for 4DWave. channel - hardware channel number 0-63 - + Return Value: none - + ---------------------------------------------------------------------------*/ static void snd_trident_free_pcm_channel(struct snd_trident *trident, int channel) @@ -383,13 +383,13 @@ static void snd_trident_free_pcm_channel(struct snd_trident *trident, int channe /*--------------------------------------------------------------------------- unsigned int snd_trident_allocate_synth_channel(void) - + Description: Allocate hardware channel in Bank A (0-31). - + Parameters : trident - pointer to target device class for 4DWave. - + Return Value: hardware channel - 0-31 or -1 when no channel is available - + ---------------------------------------------------------------------------*/ static int snd_trident_allocate_synth_channel(struct snd_trident * trident) @@ -407,15 +407,15 @@ static int snd_trident_allocate_synth_channel(struct snd_trident * trident) } /*--------------------------------------------------------------------------- - void snd_trident_free_synth_channel( int channel ) - + void snd_trident_free_synth_channel( int channel) + Description: Free hardware channel in Bank B (0-31). - + Parameters : trident - pointer to target device class for 4DWave. channel - hardware channel number 0-63 - + Return Value: none - + ---------------------------------------------------------------------------*/ static void snd_trident_free_synth_channel(struct snd_trident *trident, int channel) @@ -431,14 +431,14 @@ static void snd_trident_free_synth_channel(struct snd_trident *trident, int chan /*--------------------------------------------------------------------------- snd_trident_write_voice_regs - + Description: This routine will complete and write the 5 hardware channel registers to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure Each register field. - + ---------------------------------------------------------------------------*/ void snd_trident_write_voice_regs(struct snd_trident * trident, @@ -509,14 +509,14 @@ EXPORT_SYMBOL(snd_trident_write_voice_regs); /*--------------------------------------------------------------------------- snd_trident_write_cso_reg - + Description: This routine will write the new CSO offset register to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure CSO - new CSO value - + ---------------------------------------------------------------------------*/ static void snd_trident_write_cso_reg(struct snd_trident * trident, @@ -535,14 +535,14 @@ static void snd_trident_write_cso_reg(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_write_eso_reg - + Description: This routine will write the new ESO offset register to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure ESO - new ESO value - + ---------------------------------------------------------------------------*/ static void snd_trident_write_eso_reg(struct snd_trident * trident, @@ -561,14 +561,14 @@ static void snd_trident_write_eso_reg(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_write_vol_reg - + Description: This routine will write the new voice volume register to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure Vol - new voice volume - + ---------------------------------------------------------------------------*/ static void snd_trident_write_vol_reg(struct snd_trident * trident, @@ -583,7 +583,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident, outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2)); break; case TRIDENT_DEVICE_ID_SI7018: - // printk("voice->Vol = 0x%x\n", voice->Vol); + /* printk("voice->Vol = 0x%x\n", voice->Vol); */ outw((voice->CTRL << 12) | voice->Vol, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); break; @@ -592,14 +592,14 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_write_pan_reg - + Description: This routine will write the new voice pan register to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure Pan - new pan value - + ---------------------------------------------------------------------------*/ static void snd_trident_write_pan_reg(struct snd_trident * trident, @@ -614,14 +614,14 @@ static void snd_trident_write_pan_reg(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_write_rvol_reg - + Description: This routine will write the new reverb volume register to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure RVol - new reverb volume - + ---------------------------------------------------------------------------*/ static void snd_trident_write_rvol_reg(struct snd_trident * trident, @@ -638,14 +638,14 @@ static void snd_trident_write_rvol_reg(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_write_cvol_reg - + Description: This routine will write the new chorus volume register to hardware. - + Parameters: trident - pointer to target device class for 4DWave. voice - synthesizer voice structure CVol - new chorus volume - + ---------------------------------------------------------------------------*/ static void snd_trident_write_cvol_reg(struct snd_trident * trident, @@ -664,21 +664,21 @@ static void snd_trident_write_cvol_reg(struct snd_trident * trident, snd_trident_convert_rate Description: This routine converts rate in HZ to hardware delta value. - + Parameters: trident - pointer to target device class for 4DWave. rate - Real or Virtual channel number. - + Returns: Delta value. - + ---------------------------------------------------------------------------*/ static unsigned int snd_trident_convert_rate(unsigned int rate) { unsigned int delta; - // We special case 44100 and 8000 since rounding with the equation - // does not give us an accurate enough value. For 11025 and 22050 - // the equation gives us the best answer. All other frequencies will - // also use the equation. JDW + /* We special case 44100 and 8000 since rounding with the equation */ + /* does not give us an accurate enough value. For 11025 and 22050 */ + /* the equation gives us the best answer. All other frequencies will */ + /* also use the equation. JDW */ if (rate == 44100) delta = 0xeb3; else if (rate == 8000) @@ -694,21 +694,21 @@ static unsigned int snd_trident_convert_rate(unsigned int rate) snd_trident_convert_adc_rate Description: This routine converts rate in HZ to hardware delta value. - + Parameters: trident - pointer to target device class for 4DWave. rate - Real or Virtual channel number. - + Returns: Delta value. - + ---------------------------------------------------------------------------*/ static unsigned int snd_trident_convert_adc_rate(unsigned int rate) { unsigned int delta; - // We special case 44100 and 8000 since rounding with the equation - // does not give us an accurate enough value. For 11025 and 22050 - // the equation gives us the best answer. All other frequencies will - // also use the equation. JDW + /* We special case 44100 and 8000 since rounding with the equation */ + /* does not give us an accurate enough value. For 11025 and 22050 */ + /* the equation gives us the best answer. All other frequencies will */ + /* also use the equation. JDW */ if (rate == 44100) delta = 0x116a; else if (rate == 8000) @@ -724,12 +724,12 @@ static unsigned int snd_trident_convert_adc_rate(unsigned int rate) snd_trident_spurious_threshold Description: This routine converts rate in HZ to spurious threshold. - + Parameters: trident - pointer to target device class for 4DWave. rate - Real or Virtual channel number. - + Returns: Delta value. - + ---------------------------------------------------------------------------*/ static unsigned int snd_trident_spurious_threshold(unsigned int rate, unsigned int period_size) @@ -746,12 +746,12 @@ static unsigned int snd_trident_spurious_threshold(unsigned int rate, snd_trident_control_mode Description: This routine returns a control mode for a PCM channel. - + Parameters: trident - pointer to target device class for 4DWave. substream - PCM substream - + Returns: Control value. - + ---------------------------------------------------------------------------*/ static unsigned int snd_trident_control_mode(struct snd_pcm_substream *substream) { @@ -763,11 +763,11 @@ static unsigned int snd_trident_control_mode(struct snd_pcm_substream *substream */ CTRL = 0x00000001; if (snd_pcm_format_width(runtime->format) == 16) - CTRL |= 0x00000008; // 16-bit data + CTRL |= 0x00000008; /* 16-bit data */ if (snd_pcm_format_signed(runtime->format)) - CTRL |= 0x00000002; // signed data + CTRL |= 0x00000002; /* signed data */ if (runtime->channels > 1) - CTRL |= 0x00000004; // stereo data + CTRL |= 0x00000004; /* stereo data */ return CTRL; } @@ -777,15 +777,15 @@ static unsigned int snd_trident_control_mode(struct snd_pcm_substream *substream /*--------------------------------------------------------------------------- snd_trident_ioctl - + Description: Device I/O control handler for playback/capture parameters. - + Parameters: substream - PCM substream class cmd - what ioctl message to process - arg - additional message infoarg - + arg - additional message infoarg + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_ioctl(struct snd_pcm_substream *substream, @@ -800,14 +800,14 @@ static int snd_trident_ioctl(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_allocate_pcm_mem - + Description: Allocate PCM ring buffer for given substream - + Parameters: substream - PCM substream class hw_params - hardware parameters - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_allocate_pcm_mem(struct snd_pcm_substream *substream, @@ -834,14 +834,14 @@ static int snd_trident_allocate_pcm_mem(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_allocate_evoice - + Description: Allocate extra voice as interrupt generator - + Parameters: substream - PCM substream class hw_params - hardware parameters - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_allocate_evoice(struct snd_pcm_substream *substream, @@ -874,14 +874,14 @@ static int snd_trident_allocate_evoice(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_hw_params - + Description: Set the hardware parameters for the playback device. - + Parameters: substream - PCM substream class hw_params - hardware parameters - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_hw_params(struct snd_pcm_substream *substream, @@ -897,13 +897,13 @@ static int snd_trident_hw_params(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_playback_hw_free - + Description: Release the hardware resources for the playback device. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_hw_free(struct snd_pcm_substream *substream) @@ -929,13 +929,13 @@ static int snd_trident_hw_free(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_playback_prepare - + Description: Prepare playback device for playback. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_playback_prepare(struct snd_pcm_substream *substream) @@ -946,7 +946,7 @@ static int snd_trident_playback_prepare(struct snd_pcm_substream *substream) struct snd_trident_voice *evoice = voice->extra; struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[substream->number]; - spin_lock_irq(&trident->reg_lock); + spin_lock_irq(&trident->reg_lock); /* set delta (rate) value */ voice->Delta = snd_trident_convert_rate(runtime->rate); @@ -957,7 +957,7 @@ static int snd_trident_playback_prepare(struct snd_pcm_substream *substream) voice->LBA = voice->memblk->offset; else voice->LBA = runtime->dma_addr; - + voice->CSO = 0; voice->ESO = runtime->buffer_size - 1; /* in samples */ voice->CTRL = snd_trident_control_mode(substream); @@ -1014,14 +1014,14 @@ static int snd_trident_playback_prepare(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_capture_hw_params - + Description: Set the hardware parameters for the capture device. - + Parameters: substream - PCM substream class hw_params - hardware parameters - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_capture_hw_params(struct snd_pcm_substream *substream, @@ -1032,13 +1032,13 @@ static int snd_trident_capture_hw_params(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_capture_prepare - + Description: Prepare capture device for playback. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) @@ -1050,29 +1050,29 @@ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&trident->reg_lock); - // Initilize the channel and set channel Mode + /* Initilize the channel and set channel Mode */ outb(0, TRID_REG(trident, LEGACY_DMAR15)); - // Set DMA channel operation mode register + /* Set DMA channel operation mode register */ outb(0x54, TRID_REG(trident, LEGACY_DMAR11)); - // Set channel buffer Address, DMAR0 expects contiguous PCI memory area + /* Set channel buffer Address, DMAR0 expects contiguous PCI memory area */ voice->LBA = runtime->dma_addr; outl(voice->LBA, TRID_REG(trident, LEGACY_DMAR0)); if (voice->memblk) voice->LBA = voice->memblk->offset; - // set ESO + /* set ESO */ ESO_bytes = snd_pcm_lib_buffer_bytes(substream) - 1; outb((ESO_bytes & 0x00ff0000) >> 16, TRID_REG(trident, LEGACY_DMAR6)); outw((ESO_bytes & 0x0000ffff), TRID_REG(trident, LEGACY_DMAR4)); ESO_bytes++; - // Set channel sample rate, 4.12 format + /* Set channel sample rate, 4.12 format */ val = (((unsigned int) 48000L << 12) + (runtime->rate/2)) / runtime->rate; outw(val, TRID_REG(trident, T4D_SBDELTA_DELTA_R)); - // Set channel interrupt blk length + /* Set channel interrupt blk length */ if (snd_pcm_format_width(runtime->format) == 16) { val = (unsigned short) ((ESO_bytes >> 1) - 1); } else { @@ -1081,9 +1081,9 @@ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) outl((val << 16) | val, TRID_REG(trident, T4D_SBBL_SBCL)); - // Right now, set format and start to run captureing, - // continuous run loop enable. - trident->bDMAStart = 0x19; // 0001 1001b + /* Right now, set format and start to run captureing, */ + /* continuous run loop enable. */ + trident->bDMAStart = 0x19; /* 0001 1001b */ if (snd_pcm_format_width(runtime->format) == 16) trident->bDMAStart |= 0x80; @@ -1092,7 +1092,7 @@ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) if (runtime->channels > 1) trident->bDMAStart |= 0x40; - // Prepare capture intr channel + /* Prepare capture intr channel */ voice->Delta = snd_trident_convert_rate(runtime->rate); voice->spurious_threshold = snd_trident_spurious_threshold(runtime->rate, runtime->period_size); @@ -1100,7 +1100,7 @@ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) voice->isync_mark = runtime->period_size; voice->isync_max = runtime->buffer_size; - // Set voice parameters + /* Set voice parameters */ voice->CSO = 0; voice->ESO = voice->isync_ESO = (runtime->period_size * 2) + 6 - 1; voice->CTRL = snd_trident_control_mode(substream); @@ -1123,14 +1123,14 @@ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_si7018_capture_hw_params - + Description: Set the hardware parameters for the capture device. - + Parameters: substream - PCM substream class hw_params - hardware parameters - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_si7018_capture_hw_params(struct snd_pcm_substream *substream, @@ -1146,13 +1146,13 @@ static int snd_trident_si7018_capture_hw_params(struct snd_pcm_substream *substr /*--------------------------------------------------------------------------- snd_trident_si7018_capture_hw_free - + Description: Release the hardware resources for the capture device. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_si7018_capture_hw_free(struct snd_pcm_substream *substream) @@ -1172,13 +1172,13 @@ static int snd_trident_si7018_capture_hw_free(struct snd_pcm_substream *substrea /*--------------------------------------------------------------------------- snd_trident_si7018_capture_prepare - + Description: Prepare capture device for playback. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_si7018_capture_prepare(struct snd_pcm_substream *substream) @@ -1194,7 +1194,7 @@ static int snd_trident_si7018_capture_prepare(struct snd_pcm_substream *substrea voice->Delta = snd_trident_convert_adc_rate(runtime->rate); voice->spurious_threshold = snd_trident_spurious_threshold(runtime->rate, runtime->period_size); - // Set voice parameters + /* Set voice parameters */ voice->CSO = 0; voice->ESO = runtime->buffer_size - 1; /* in samples */ voice->CTRL = snd_trident_control_mode(substream); @@ -1236,20 +1236,20 @@ static int snd_trident_si7018_capture_prepare(struct snd_pcm_substream *substrea evoice->isync_mark = runtime->period_size; evoice->ESO = (runtime->period_size * 2) - 1; } - + spin_unlock_irq(&trident->reg_lock); return 0; } /*--------------------------------------------------------------------------- snd_trident_foldback_prepare - + Description: Prepare foldback capture device for playback. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_foldback_prepare(struct snd_pcm_substream *substream) @@ -1320,14 +1320,14 @@ static int snd_trident_foldback_prepare(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_spdif_hw_params - + Description: Set the hardware parameters for the spdif device. - + Parameters: substream - PCM substream class hw_params - hardware parameters - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_spdif_hw_params(struct snd_pcm_substream *substream, @@ -1355,21 +1355,21 @@ static int snd_trident_spdif_hw_params(struct snd_pcm_substream *substream, else trident->spdif_pcm_bits &= ~(IEC958_AES3_CON_FS << 24); if (params_rate(hw_params) >= 48000) { - trident->spdif_pcm_ctrl = 0x3c; // 48000 Hz + trident->spdif_pcm_ctrl = 0x3c; /* 48000 Hz */ trident->spdif_pcm_bits |= trident->spdif_bits & IEC958_AES0_PROFESSIONAL ? IEC958_AES0_PRO_FS_48000 : (IEC958_AES3_CON_FS_48000 << 24); } else if (params_rate(hw_params) >= 44100) { - trident->spdif_pcm_ctrl = 0x3e; // 44100 Hz + trident->spdif_pcm_ctrl = 0x3e; /* 44100 Hz */ trident->spdif_pcm_bits |= trident->spdif_bits & IEC958_AES0_PROFESSIONAL ? IEC958_AES0_PRO_FS_44100 : (IEC958_AES3_CON_FS_44100 << 24); } else { - trident->spdif_pcm_ctrl = 0x3d; // 32000 Hz + trident->spdif_pcm_ctrl = 0x3d; /* 32000 Hz */ trident->spdif_pcm_bits |= trident->spdif_bits & IEC958_AES0_PROFESSIONAL ? IEC958_AES0_PRO_FS_32000 : @@ -1386,13 +1386,13 @@ static int snd_trident_spdif_hw_params(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_spdif_prepare - + Description: Prepare SPDIF device for playback. - + Parameters: substream - PCM substream class - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_spdif_prepare(struct snd_pcm_substream *substream) @@ -1458,7 +1458,7 @@ static int snd_trident_spdif_prepare(struct snd_pcm_substream *substream) outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS)); } else { /* SiS */ - + /* set delta (rate) value */ voice->Delta = 0x800; voice->spurious_threshold = snd_trident_spurious_threshold(48000, runtime->period_size); @@ -1524,19 +1524,19 @@ static int snd_trident_spdif_prepare(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_trigger - + Description: Start/stop devices - + Parameters: substream - PCM substream class cmd - trigger command (STOP, GO) - + Returns: Error status - + ---------------------------------------------------------------------------*/ static int snd_trident_trigger(struct snd_pcm_substream *substream, int cmd) - + { struct snd_trident *trident = snd_pcm_substream_chip(substream); struct snd_pcm_substream *s; @@ -1624,13 +1624,13 @@ static int snd_trident_trigger(struct snd_pcm_substream *substream, /*--------------------------------------------------------------------------- snd_trident_playback_pointer - + Description: This routine return the playback position - + Parameters: substream - PCM substream class Returns: position of buffer - + ---------------------------------------------------------------------------*/ static snd_pcm_uframes_t snd_trident_playback_pointer(struct snd_pcm_substream *substream) @@ -1649,7 +1649,7 @@ static snd_pcm_uframes_t snd_trident_playback_pointer(struct snd_pcm_substream * if (trident->device != TRIDENT_DEVICE_ID_NX) { cso = inw(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS + 2)); - } else { // ID_4DWAVE_NX + } else { /* ID_4DWAVE_NX */ cso = (unsigned int) inl(TRID_REG(trident, CH_NX_DELTA_CSO)) & 0x00ffffff; } @@ -1663,13 +1663,13 @@ static snd_pcm_uframes_t snd_trident_playback_pointer(struct snd_pcm_substream * /*--------------------------------------------------------------------------- snd_trident_capture_pointer - + Description: This routine return the capture position - + Parameters: pcm1 - PCM device class Returns: position of buffer - + ---------------------------------------------------------------------------*/ static snd_pcm_uframes_t snd_trident_capture_pointer(struct snd_pcm_substream *substream) @@ -1693,13 +1693,13 @@ static snd_pcm_uframes_t snd_trident_capture_pointer(struct snd_pcm_substream *s /*--------------------------------------------------------------------------- snd_trident_spdif_pointer - + Description: This routine return the SPDIF playback position - + Parameters: substream - PCM substream class Returns: position of buffer - + ---------------------------------------------------------------------------*/ static snd_pcm_uframes_t snd_trident_spdif_pointer(struct snd_pcm_substream *substream) @@ -1868,10 +1868,10 @@ static int snd_trident_playback_open(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_playback_close - - Description: This routine will close the 4DWave playback device. For now + + Description: This routine will close the 4DWave playback device. For now we will simply free the dma transfer buffer. - + Parameters: substream - PCM substream class ---------------------------------------------------------------------------*/ @@ -1887,13 +1887,13 @@ static int snd_trident_playback_close(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_spdif_open - + Description: This routine will open the 4DWave SPDIF device. Parameters: substream - PCM substream class Returns: status - success or failure flag - + ---------------------------------------------------------------------------*/ static int snd_trident_spdif_open(struct snd_pcm_substream *substream) @@ -1901,7 +1901,7 @@ static int snd_trident_spdif_open(struct snd_pcm_substream *substream) struct snd_trident *trident = snd_pcm_substream_chip(substream); struct snd_trident_voice *voice; struct snd_pcm_runtime *runtime = substream->runtime; - + voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_PCM, 0, 0); if (voice == NULL) return -EAGAIN; @@ -1930,9 +1930,9 @@ static int snd_trident_spdif_open(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_spdif_close - + Description: This routine will close the 4DWave SPDIF device. - + Parameters: substream - PCM substream class ---------------------------------------------------------------------------*/ @@ -1943,7 +1943,7 @@ static int snd_trident_spdif_close(struct snd_pcm_substream *substream) unsigned int temp; spin_lock_irq(&trident->reg_lock); - // restore default SPDIF setting + /* restore default SPDIF setting */ if (trident->device != TRIDENT_DEVICE_ID_SI7018) { outb(trident->spdif_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); outl(trident->spdif_bits, TRID_REG(trident, NX_SPCSTATUS)); @@ -1966,7 +1966,7 @@ static int snd_trident_spdif_close(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_capture_open - + Description: This routine will open the 4DWave capture device. Parameters: substream - PCM substream class @@ -1996,10 +1996,10 @@ static int snd_trident_capture_open(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_capture_close - - Description: This routine will close the 4DWave capture device. For now + + Description: This routine will close the 4DWave capture device. For now we will simply free the dma transfer buffer. - + Parameters: substream - PCM substream class ---------------------------------------------------------------------------*/ @@ -2010,7 +2010,7 @@ static int snd_trident_capture_close(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_foldback_open - + Description: This routine will open the 4DWave foldback capture device. Parameters: substream - PCM substream class @@ -2039,10 +2039,10 @@ static int snd_trident_foldback_open(struct snd_pcm_substream *substream) /*--------------------------------------------------------------------------- snd_trident_foldback_close - - Description: This routine will close the 4DWave foldback capture device. + + Description: This routine will close the 4DWave foldback capture device. For now we will simply free the dma transfer buffer. - + Parameters: substream - PCM substream class ---------------------------------------------------------------------------*/ @@ -2052,7 +2052,7 @@ static int snd_trident_foldback_close(struct snd_pcm_substream *substream) struct snd_trident_voice *voice; struct snd_pcm_runtime *runtime = substream->runtime; voice = runtime->private_data; - + /* stop capture channel */ spin_lock_irq(&trident->reg_lock); outb(0x00, TRID_REG(trident, T4D_RCI + voice->foldback_chan)); @@ -2156,13 +2156,13 @@ static struct snd_pcm_ops snd_trident_spdif_7018_ops = { /*--------------------------------------------------------------------------- snd_trident_pcm - + Description: This routine registers the 4DWave device for PCM support. - + Parameters: trident - pointer to target device class for 4DWave. Returns: None - + ---------------------------------------------------------------------------*/ int __devinit snd_trident_pcm(struct snd_trident * trident, @@ -2214,13 +2214,13 @@ int __devinit snd_trident_pcm(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_foldback_pcm - + Description: This routine registers the 4DWave device for foldback PCM support. - + Parameters: trident - pointer to target device class for 4DWave. Returns: None - + ---------------------------------------------------------------------------*/ int __devinit snd_trident_foldback_pcm(struct snd_trident * trident, @@ -2271,13 +2271,13 @@ int __devinit snd_trident_foldback_pcm(struct snd_trident * trident, /*--------------------------------------------------------------------------- snd_trident_spdif - + Description: This routine registers the 4DWave-NX device for SPDIF support. - + Parameters: trident - pointer to target device class for 4DWave-NX. Returns: None - + ---------------------------------------------------------------------------*/ int __devinit snd_trident_spdif_pcm(struct snd_trident * trident, @@ -2904,7 +2904,7 @@ static void snd_trident_notify_pcm_change1(struct snd_card *card, { struct snd_ctl_elem_id id; - if (! kctl) + if (!kctl) return; if (activate) kctl->vd[num].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -2957,13 +2957,13 @@ static int snd_trident_pcm_mixer_free(struct snd_trident *trident, struct snd_tr /*--------------------------------------------------------------------------- snd_trident_mixer - + Description: This routine registers the 4DWave device for mixer support. - + Parameters: trident - pointer to target device class for 4DWave. Returns: None - + ---------------------------------------------------------------------------*/ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spdif_device) @@ -2989,7 +2989,7 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd _ac97.private_data = trident; trident->ac97_detect = 1; - __again: +__again: if ((err = snd_ac97_mixer(trident->ac97_bus, &_ac97, &trident->ac97)) < 0) { if (trident->device == TRIDENT_DEVICE_ID_SI7018) { if ((err = snd_trident_sis_reset(trident)) < 0) @@ -3000,7 +3000,7 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd } goto __out; } - + /* secondary codec? */ if (trident->device == TRIDENT_DEVICE_ID_SI7018 && (inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) & SI_AC97_PRIMARY_READY) != 0) { @@ -3008,7 +3008,7 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd err = snd_ac97_mixer(trident->ac97_bus, &_ac97, &trident->ac97_sec); if (err < 0) snd_printk(KERN_ERR "SI7018: the secondary codec - invalid access\n"); -#if 0 // only for my testing purpose --jk +#if 0 /* only for my testing purpose --jk */ { struct snd_ac97 *mc97; err = snd_ac97_modem(trident->card, &_ac97, &mc97); @@ -3017,7 +3017,7 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd } #endif } - + trident->ac97_detect = 0; if (trident->device != TRIDENT_DEVICE_ID_SI7018) { @@ -3034,7 +3034,7 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd for (idx = 0; idx < 32; idx++) { struct snd_trident_pcm_mixer *tmix; - + tmix = &trident->pcm_mixer[idx]; tmix->voice = NULL; } @@ -3042,7 +3042,7 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd goto __nomem; if ((err = snd_ctl_add(card, trident->ctl_vol))) goto __out; - + if ((trident->ctl_pan = snd_ctl_new1(&snd_trident_pcm_pan_control, trident)) == NULL) goto __nomem; if ((err = snd_ctl_add(card, trident->ctl_pan))) @@ -3114,10 +3114,10 @@ static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spd err = 0; goto __out; - __nomem: +__nomem: err = -ENOMEM; - __out: +__out: kfree(uctl); return err; @@ -3161,7 +3161,7 @@ static int snd_trident_gameport_cooked_read(struct gameport *gameport, int *axes axes[i] = inw(TRID_REG(chip, GAMEPORT_AXES + i * 2)); if (axes[i] == 0xffff) axes[i] = -1; } - + return 0; } @@ -3284,11 +3284,11 @@ static int snd_trident_sis_reset(struct snd_trident *trident) return 0; } -/* +/* * /proc interface */ -static void snd_trident_proc_read(struct snd_info_entry *entry, +static void snd_trident_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_trident *trident = entry->private_data; @@ -3327,10 +3327,10 @@ static void __devinit snd_trident_proc_init(struct snd_trident * trident) { struct snd_info_entry *entry; const char *s = "trident"; - + if (trident->device == TRIDENT_DEVICE_ID_SI7018) s = "sis7018"; - if (! snd_card_proc_new(trident->card, s, &entry)) + if (!snd_card_proc_new(trident->card, s, &entry)) snd_info_set_text_ops(entry, trident, snd_trident_proc_read); } @@ -3342,14 +3342,14 @@ static int snd_trident_dev_free(struct snd_device *device) /*--------------------------------------------------------------------------- snd_trident_tlb_alloc - + Description: Allocate and set up the TLB page table on 4D NX. Each entry has 4 bytes (physical PCI address). - + Parameters: trident - pointer to target device class for 4DWave. Returns: 0 or negative error code - + ---------------------------------------------------------------------------*/ static int __devinit snd_trident_tlb_alloc(struct snd_trident *trident) @@ -3418,7 +3418,7 @@ static int snd_trident_4d_dx_init(struct snd_trident *trident) udelay(100); pci_write_config_byte(pci, 0x46, 0); /* release reset */ udelay(100); - + /* warm reset of the AC'97 codec */ outl(0x00000001, TRID_REG(trident, DX_ACR2_AC97_COM_STAT)); udelay(100); @@ -3436,7 +3436,7 @@ static int snd_trident_4d_dx_init(struct snd_trident *trident) snd_printk(KERN_ERR "AC'97 codec ready error\n"); return -EIO; - __dx_ok: +__dx_ok: snd_trident_stop_all_voices(trident); return 0; @@ -3474,7 +3474,7 @@ static int snd_trident_4d_nx_init(struct snd_trident *trident) snd_printk(KERN_ERR "AC'97 codec ready error [0x%x]\n", inl(TRID_REG(trident, NX_ACR0_AC97_COM_STAT))); return -EIO; - __nx_ok: +__nx_ok: /* DAC on */ trident->ac97_ctrl = 0x00000002; outl(trident->ac97_ctrl, TRID_REG(trident, NX_ACR0_AC97_COM_STAT)); @@ -3519,10 +3519,10 @@ static int snd_trident_sis_init(struct snd_trident *trident) /*--------------------------------------------------------------------------- snd_trident_create - + Description: This routine will create the device specific class for the 4DWave card. It will also perform basic initialization. - + Parameters: card - which card to create pci - interface to PCI bus resource info dma1ptr - playback dma buffer @@ -3530,7 +3530,7 @@ static int snd_trident_sis_init(struct snd_trident *trident) irqptr - interrupt resource info Returns: 4DWave device class private data - + ---------------------------------------------------------------------------*/ int __devinit snd_trident_create(struct snd_card *card, @@ -3560,7 +3560,7 @@ int __devinit snd_trident_create(struct snd_card *card, pci_disable_device(pci); return -ENXIO; } - + trident = kzalloc(sizeof(*trident), GFP_KERNEL); if (trident == NULL) { pci_disable_device(pci); @@ -3577,7 +3577,7 @@ int __devinit snd_trident_create(struct snd_card *card, if (pcm_streams > 32) pcm_streams = 32; trident->ChanPCM = pcm_streams; - if (max_wavetable_size < 0 ) + if (max_wavetable_size < 0) max_wavetable_size = 0; trident->synth.max_size = max_wavetable_size * 1024; trident->irq = -1; @@ -3639,7 +3639,7 @@ int __devinit snd_trident_create(struct snd_card *card, if ((err = snd_trident_mixer(trident, pcm_spdif_device)) < 0) return err; - + /* initialise synth voices */ for (i = 0; i < 64; i++) { voice = &trident->synth.voices[i]; @@ -3665,21 +3665,21 @@ int __devinit snd_trident_create(struct snd_card *card, /*--------------------------------------------------------------------------- snd_trident_free - + Description: This routine will free the device specific class for - the 4DWave card. - + the 4DWave card. + Parameters: trident - device specific private data for 4DWave card Returns: None. - + ---------------------------------------------------------------------------*/ static int snd_trident_free(struct snd_trident *trident) { snd_trident_free_gameport(trident); snd_trident_disable_eso(trident); - // Disable S/PDIF out + /* Disable S/PDIF out */ if (trident->device == TRIDENT_DEVICE_ID_NX) outb(0x00, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); else if (trident->device == TRIDENT_DEVICE_ID_SI7018) { @@ -3704,9 +3704,9 @@ static int snd_trident_free(struct snd_trident *trident) /*--------------------------------------------------------------------------- snd_trident_interrupt - + Description: ISR for Trident 4DWave device - + Parameters: trident - device specific private data for 4DWave card Problems: It seems that Trident chips generates interrupts more than @@ -3717,7 +3717,7 @@ static int snd_trident_free(struct snd_trident *trident) work on all computers. [jaroslav] Returns: None. - + ---------------------------------------------------------------------------*/ static irqreturn_t snd_trident_interrupt(int irq, void *dev_id) @@ -3731,7 +3731,7 @@ static irqreturn_t snd_trident_interrupt(int irq, void *dev_id) if ((audio_int & (ADDRESS_IRQ|MPU401_IRQ)) == 0) return IRQ_NONE; if (audio_int & ADDRESS_IRQ) { - // get interrupt status for all channels + /* get interrupt status for all channels */ spin_lock(&trident->reg_lock); stimer = inl(TRID_REG(trident, T4D_STIMER)) & 0x00ffffff; chn_int = inl(TRID_REG(trident, T4D_AINT_A)); @@ -3813,7 +3813,7 @@ static irqreturn_t snd_trident_interrupt(int irq, void *dev_id) inb(TRID_REG(trident, T4D_MPUR0)); } } - // outl((ST_TARGET_REACHED | MIXER_OVERFLOW | MIXER_UNDERFLOW), TRID_REG(trident, T4D_MISCINT)); + /* outl((ST_TARGET_REACHED | MIXER_OVERFLOW | MIXER_UNDERFLOW), TRID_REG(trident, T4D_MISCINT)); */ return IRQ_HANDLED; } @@ -3826,7 +3826,7 @@ struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, spin_lock_irqsave(&trident->voice_alloc, flags); if (type == SNDRV_TRIDENT_VOICE_TYPE_PCM) { idx = snd_trident_allocate_pcm_channel(trident); - if(idx < 0) { + if (idx < 0) { spin_unlock_irqrestore(&trident->voice_alloc, flags); return NULL; } @@ -3842,7 +3842,7 @@ struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, } if (type == SNDRV_TRIDENT_VOICE_TYPE_SYNTH) { idx = snd_trident_allocate_synth_channel(trident); - if(idx < 0) { + if (idx < 0) { spin_unlock_irqrestore(&trident->voice_alloc, flags); return NULL; } diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index f9779e2..4bf4bbb 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -23,7 +23,7 @@ * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/pci.h> #include <linux/time.h> #include <linux/mutex.h> @@ -203,7 +203,7 @@ snd_trident_alloc_sg_pages(struct snd_trident *trident, if (snd_BUG_ON(!hdr)) return NULL; - + mutex_lock(&hdr->block_mutex); blk = search_empty(hdr, runtime->dma_bytes); @@ -211,7 +211,7 @@ snd_trident_alloc_sg_pages(struct snd_trident *trident, mutex_unlock(&hdr->block_mutex); return NULL; } - + /* set TLB entries */ idx = 0; for (page = firstpg(blk); page <= lastpg(blk); page++, idx++) { @@ -219,7 +219,7 @@ snd_trident_alloc_sg_pages(struct snd_trident *trident, dma_addr_t addr = snd_pcm_sgbuf_get_addr(substream, ofs); unsigned long ptr = (unsigned long) snd_pcm_sgbuf_get_ptr(substream, ofs); - if (! is_valid_page(addr)) { + if (!is_valid_page(addr)) { __snd_util_mem_free(hdr, blk); mutex_unlock(&hdr->block_mutex); return NULL; @@ -258,13 +258,13 @@ snd_trident_alloc_cont_pages(struct snd_trident *trident, mutex_unlock(&hdr->block_mutex); return NULL; } - + /* set TLB entries */ addr = runtime->dma_addr; ptr = (unsigned long)runtime->dma_area; for (page = firstpg(blk); page <= lastpg(blk); page++, ptr += SNDRV_TRIDENT_PAGE_SIZE, addr += SNDRV_TRIDENT_PAGE_SIZE) { - if (! is_valid_page(addr)) { + if (!is_valid_page(addr)) { __snd_util_mem_free(hdr, blk); mutex_unlock(&hdr->block_mutex); return NULL;

1 0

[alsa-devel] [PATCH 20/24] sound/pci: coding style fixes: sonicvibes
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/sonicvibes.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index cd408b8..e6ed3a2 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -39,7 +39,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_DESCRIPTION("S3 SonicVibes PCI"); @@ -523,7 +523,7 @@ static int snd_sonicvibes_hw_constraint_dac_rate(struct snd_pcm_hw_params *param { unsigned int rate, div, r, m, n; - if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min == + if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min == hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max) { rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min; div = 48000 / rate; @@ -895,7 +895,7 @@ static int snd_sonicvibes_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl static int snd_sonicvibes_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&sonic->reg_lock); ucontrol->value.enumerated.item[0] = ((snd_sonicvibes_in1(sonic, SV_IREG_LEFT_ADC) & SV_RECSRC_OUT) >> 5) - 1; ucontrol->value.enumerated.item[1] = ((snd_sonicvibes_in1(sonic, SV_IREG_RIGHT_ADC) & SV_RECSRC_OUT) >> 5) - 1; @@ -908,7 +908,7 @@ static int snd_sonicvibes_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol); unsigned short left, right, oval1, oval2; int change; - + if (ucontrol->value.enumerated.item[0] >= 7 || ucontrol->value.enumerated.item[1] >= 7) return -EINVAL; @@ -950,7 +950,7 @@ static int snd_sonicvibes_get_single(struct snd_kcontrol *kcontrol, struct snd_c int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; int invert = (kcontrol->private_value >> 24) & 0xff; - + spin_lock_irq(&sonic->reg_lock); ucontrol->value.integer.value[0] = (snd_sonicvibes_in1(sonic, reg)>> shift) & mask; spin_unlock_irq(&sonic->reg_lock); @@ -968,7 +968,7 @@ static int snd_sonicvibes_put_single(struct snd_kcontrol *kcontrol, struct snd_c int invert = (kcontrol->private_value >> 24) & 0xff; int change; unsigned short val, oval; - + val = (ucontrol->value.integer.value[0] & mask); if (invert) val = mask - val; @@ -1008,7 +1008,7 @@ static int snd_sonicvibes_get_double(struct snd_kcontrol *kcontrol, struct snd_c int shift_right = (kcontrol->private_value >> 19) & 0x07; int mask = (kcontrol->private_value >> 24) & 0xff; int invert = (kcontrol->private_value >> 22) & 1; - + spin_lock_irq(&sonic->reg_lock); ucontrol->value.integer.value[0] = (snd_sonicvibes_in1(sonic, left_reg) >> shift_left) & mask; ucontrol->value.integer.value[1] = (snd_sonicvibes_in1(sonic, right_reg) >> shift_right) & mask; @@ -1031,7 +1031,7 @@ static int snd_sonicvibes_put_double(struct snd_kcontrol *kcontrol, struct snd_c int invert = (kcontrol->private_value >> 22) & 1; int change; unsigned short val1, val2, oval1, oval2; - + val1 = ucontrol->value.integer.value[0] & mask; val2 = ucontrol->value.integer.value[1] & mask; if (invert) { @@ -1110,7 +1110,7 @@ static int __devinit snd_sonicvibes_mixer(struct sonicvibes * sonic) */ -static void snd_sonicvibes_proc_read(struct snd_info_entry *entry, +static void snd_sonicvibes_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct sonicvibes *sonic = entry->private_data; @@ -1144,7 +1144,7 @@ static void __devinit snd_sonicvibes_proc_init(struct sonicvibes * sonic) { struct snd_info_entry *entry; - if (! snd_card_proc_new(sonic->card, "sonicvibes", &entry)) + if (!snd_card_proc_new(sonic->card, "sonicvibes", &entry)) snd_info_set_text_ops(entry, sonic, snd_sonicvibes_proc_read); } @@ -1422,7 +1422,7 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, dev++; return -ENOENT; } - + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; @@ -1482,7 +1482,7 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, snd_card_free(card); return err; } - + pci_set_drvdata(pci, card); dev++; return 0;

1 0

[alsa-devel] [PATCH 19/24] sound/pci: coding style fixes: sound/pci/r*
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/riptide/riptide.c | 22 +++--- sound/pci/rme32.c | 128 +++++++++++++------------- sound/pci/rme96.c | 210 +++++++++++++++++++++--------------------- 3 files changed, 180 insertions(+), 180 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e9f0706..04b21de 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -21,29 +21,29 @@ /* History: - 02/15/2004 first release - + This Driver is based on the OSS Driver version from Linuxant (riptide-0.6lnxtbeta03111100) credits from the original files: - - MODULE NAME: cnxt_rt.h + + MODULE NAME: cnxt_rt.h AUTHOR: K. Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 02/1/2000 KNL - MODULE NAME: int_mdl.c + MODULE NAME: int_mdl.c AUTHOR: Konstantin Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/01/99 KNL - - MODULE NAME: riptide.h + + MODULE NAME: riptide.h AUTHOR: O. Druzhinin (Transcribed by OLD) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/16/97 OLD - MODULE NAME: Rp_Cmdif.cpp + MODULE NAME: Rp_Cmdif.cpp AUTHOR: O. Druzhinin (Transcribed by OLD) K. Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By @@ -51,7 +51,7 @@ Adopted from NT4 driver 6/22/99 OLD Ported to Linux 9/01/99 KNL - MODULE NAME: rt_hw.c + MODULE NAME: rt_hw.c AUTHOR: O. Druzhinin (Transcribed by OLD) C. Lazarev (Transcribed by CNL) HISTORY: Major Revision Date By @@ -73,13 +73,13 @@ ----------------------------- -------- ----- Created 11/18/97 CNL - MODULE NAME: rt_mdl.c + MODULE NAME: rt_mdl.c AUTHOR: Konstantin Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/01/99 KNL - MODULE NAME: mixer.h + MODULE NAME: mixer.h AUTHOR: K. Kenney HISTORY: Major Revision Date By ----------------------------- -------- ----- @@ -97,7 +97,7 @@ #include <linux/gameport.h> #include <linux/device.h> #include <linux/firmware.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/info.h> #include <sound/control.h> diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index e7ef3a1..21ec1e2 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -6,7 +6,7 @@ * * Thanks to : Anders Torger <torger(a)ludd.luth.se>, * Henk Hesselink <henk(a)anda.nl> - * for writing the digi96-driver + * for writing the digi96-driver * and RME for all informations. * * This program is free software; you can redistribute it and/or modify @@ -22,36 +22,36 @@ * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - * + * + * * **************************************************************************** - * + * * Note #1 "Sek'd models" ................................... martin 2002-12-07 - * + * * Identical soundcards by Sek'd were labeled: * RME Digi 32 = Sek'd Prodif 32 * RME Digi 32 Pro = Sek'd Prodif 96 * RME Digi 32/8 = Sek'd Prodif Gold - * + * * **************************************************************************** - * + * * Note #2 "full duplex mode" ............................... martin 2002-12-07 - * + * * Full duplex doesn't work. All cards (32, 32/8, 32Pro) are working identical * in this mode. Rec data and play data are using the same buffer therefore. At * first you have got the playing bits in the buffer and then (after playing - * them) they were overwitten by the captured sound of the CS8412/14. Both + * them) they were overwitten by the captured sound of the CS8412/14. Both * modes (play/record) are running harmonically hand in hand in the same buffer - * and you have only one start bit plus one interrupt bit to control this + * and you have only one start bit plus one interrupt bit to control this * paired action. * This is opposite to the latter rme96 where playing and capturing is totally - * separated and so their full duplex mode is supported by alsa (using two - * start bits and two interrupts for two different buffers). + * separated and so their full duplex mode is supported by alsa (using two + * start bits and two interrupts for two different buffers). * But due to the wrong sequence of playing and capturing ALSA shows no solved * full duplex support for the rme32 at the moment. That's bad, but I'm not * able to solve it. Are you motivated enough to solve this problem now? Your * patch would be welcome! - * + * * **************************************************************************** * * "The story after the long seeking" -- tiwai @@ -85,12 +85,12 @@ #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static int fullduplex[SNDRV_CARDS]; /* = {[0 ... (SNDRV_CARDS - 1)] = 1}; */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Digi32 soundcard."); @@ -126,10 +126,10 @@ MODULE_SUPPORTED_DEVICE("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}"); #define RME32_WCR_MONO (1 << 1) /* 0=stereo, 1=mono Setting the whole card to mono doesn't seem to be very useful. - A software-solution can handle + A software-solution can handle full-duplex with one direction in - stereo and the other way in mono. - So, the hardware should work all + stereo and the other way in mono. + So, the hardware should work all the time in stereo! */ #define RME32_WCR_MODE24 (1 << 2) /* 0=16bit, 1=32bit */ #define RME32_WCR_SEL (1 << 3) /* 0=input on output, 1=normal playback/capture */ @@ -304,13 +304,13 @@ static int snd_rme32_capture_copy(struct snd_pcm_substream *substream, int chann static struct snd_pcm_hardware snd_rme32_spdif_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_S16_LE | + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), .rate_min = 32000, .rate_max = 48000, @@ -335,7 +335,7 @@ static struct snd_pcm_hardware snd_rme32_adat_info = SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START), .formats= SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_44100 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), .rate_min = 44100, .rate_max = 48000, @@ -355,13 +355,13 @@ static struct snd_pcm_hardware snd_rme32_adat_info = static struct snd_pcm_hardware snd_rme32_spdif_fd_info = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_S16_LE | + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), .rate_min = 32000, .rate_max = 48000, @@ -386,7 +386,7 @@ static struct snd_pcm_hardware snd_rme32_adat_fd_info = SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START), .formats= SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_44100 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), .rate_min = 44100, .rate_max = 48000, @@ -434,7 +434,7 @@ static int snd_rme32_capture_getrate(struct rme32 * rme32, int *is_adat) int n; *is_adat = 0; - if (rme32->rcreg & RME32_RCR_LOCK) { + if (rme32->rcreg & RME32_RCR_LOCK) { /* ADAT rate */ *is_adat = 1; } @@ -466,7 +466,7 @@ static int snd_rme32_capture_getrate(struct rme32 * rme32, int *is_adat) default: return -1; break; - } + } else switch (n) { /* supporting the CS8412 */ case 0: @@ -499,38 +499,38 @@ static int snd_rme32_playback_setrate(struct rme32 * rme32, int rate) switch (rate) { case 32000: rme32->wcreg &= ~RME32_WCR_DS_BM; - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) & + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) & ~RME32_WCR_FREQ_1; break; case 44100: rme32->wcreg &= ~RME32_WCR_DS_BM; - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_1) & + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_1) & ~RME32_WCR_FREQ_0; break; case 48000: rme32->wcreg &= ~RME32_WCR_DS_BM; - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) | + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) | RME32_WCR_FREQ_1; break; case 64000: if (rme32->pci->device != PCI_DEVICE_ID_RME_DIGI32_PRO) return -EINVAL; rme32->wcreg |= RME32_WCR_DS_BM; - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) & + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) & ~RME32_WCR_FREQ_1; break; case 88200: if (rme32->pci->device != PCI_DEVICE_ID_RME_DIGI32_PRO) return -EINVAL; rme32->wcreg |= RME32_WCR_DS_BM; - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_1) & + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_1) & ~RME32_WCR_FREQ_0; break; case 96000: if (rme32->pci->device != PCI_DEVICE_ID_RME_DIGI32_PRO) return -EINVAL; rme32->wcreg |= RME32_WCR_DS_BM; - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) | + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) | RME32_WCR_FREQ_1; break; default: @@ -552,22 +552,22 @@ static int snd_rme32_setclockmode(struct rme32 * rme32, int mode) switch (mode) { case RME32_CLOCKMODE_SLAVE: /* AutoSync */ - rme32->wcreg = (rme32->wcreg & ~RME32_WCR_FREQ_0) & + rme32->wcreg = (rme32->wcreg & ~RME32_WCR_FREQ_0) & ~RME32_WCR_FREQ_1; break; case RME32_CLOCKMODE_MASTER_32: /* Internal 32.0kHz */ - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) & + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) & ~RME32_WCR_FREQ_1; break; case RME32_CLOCKMODE_MASTER_44: /* Internal 44.1kHz */ - rme32->wcreg = (rme32->wcreg & ~RME32_WCR_FREQ_0) | + rme32->wcreg = (rme32->wcreg & ~RME32_WCR_FREQ_0) | RME32_WCR_FREQ_1; break; case RME32_CLOCKMODE_MASTER_48: /* Internal 48.0kHz */ - rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) | + rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) | RME32_WCR_FREQ_1; break; default: @@ -587,19 +587,19 @@ static int snd_rme32_setinputtype(struct rme32 * rme32, int type) { switch (type) { case RME32_INPUT_OPTICAL: - rme32->wcreg = (rme32->wcreg & ~RME32_WCR_INP_0) & + rme32->wcreg = (rme32->wcreg & ~RME32_WCR_INP_0) & ~RME32_WCR_INP_1; break; case RME32_INPUT_COAXIAL: - rme32->wcreg = (rme32->wcreg | RME32_WCR_INP_0) & + rme32->wcreg = (rme32->wcreg | RME32_WCR_INP_0) & ~RME32_WCR_INP_1; break; case RME32_INPUT_INTERNAL: - rme32->wcreg = (rme32->wcreg & ~RME32_WCR_INP_0) | + rme32->wcreg = (rme32->wcreg & ~RME32_WCR_INP_0) | RME32_WCR_INP_1; break; case RME32_INPUT_XLR: - rme32->wcreg = (rme32->wcreg | RME32_WCR_INP_0) | + rme32->wcreg = (rme32->wcreg | RME32_WCR_INP_0) | RME32_WCR_INP_1; break; default: @@ -741,7 +741,7 @@ snd_rme32_capture_hw_params(struct snd_pcm_substream *substream, if ((rate = snd_rme32_capture_getrate(rme32, &isadat)) > 0) { if ((int)params_rate(params) != rate) { spin_unlock_irq(&rme32->lock); - return -EIO; + return -EIO; } if ((isadat && runtime->hw.channels_min == 2) || (!isadat && runtime->hw.channels_min == 8)) { @@ -771,7 +771,7 @@ snd_rme32_capture_hw_params(struct snd_pcm_substream *substream, static int snd_rme32_pcm_hw_free(struct snd_pcm_substream *substream) { struct rme32 *rme32 = snd_pcm_substream_chip(substream); - if (! rme32->fullduplex_mode) + if (!rme32->fullduplex_mode) return 0; return snd_pcm_lib_free_pages(substream); } @@ -800,7 +800,7 @@ static void snd_rme32_pcm_stop(struct rme32 * rme32, int to_pause) if (rme32->wcreg & RME32_WCR_SEL) rme32->wcreg |= RME32_WCR_MUTE; writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER); - if (! to_pause) + if (!to_pause) writel(0, rme32->iobase + RME32_IO_RESET_POS); } @@ -834,7 +834,7 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = { static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime) { - if (! rme32->fullduplex_mode) { + if (!rme32->fullduplex_mode) { snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME32_BUFFER_SIZE, RME32_BUFFER_SIZE); @@ -876,7 +876,7 @@ static int snd_rme32_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate); runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; - } + } snd_rme32_set_buffer_constraint(rme32, runtime); @@ -931,10 +931,10 @@ snd_rme32_playback_adat_open(struct snd_pcm_substream *substream) int rate, dummy; struct rme32 *rme32 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - + snd_pcm_set_sync(substream); - spin_lock_irq(&rme32->lock); + spin_lock_irq(&rme32->lock); if (rme32->playback_substream != NULL) { spin_unlock_irq(&rme32->lock); return -EBUSY; @@ -943,7 +943,7 @@ snd_rme32_playback_adat_open(struct snd_pcm_substream *substream) writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER); rme32->playback_substream = substream; spin_unlock_irq(&rme32->lock); - + if (rme32->fullduplex_mode) runtime->hw = snd_rme32_adat_fd_info; else @@ -954,7 +954,7 @@ snd_rme32_playback_adat_open(struct snd_pcm_substream *substream) runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate); runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; - } + } snd_rme32_set_buffer_constraint(rme32, runtime); return 0; @@ -981,8 +981,8 @@ snd_rme32_capture_adat_open(struct snd_pcm_substream *substream) } snd_pcm_set_sync(substream); - - spin_lock_irq(&rme32->lock); + + spin_lock_irq(&rme32->lock); if (rme32->capture_substream != NULL) { spin_unlock_irq(&rme32->lock); return -EBUSY; @@ -1091,7 +1091,7 @@ snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } snd_pcm_trigger_done(s, substream); } - + /* prefill playback buffer */ if (cmd == SNDRV_PCM_TRIGGER_START && rme32->fullduplex_mode) { snd_pcm_group_for_each_entry(s, substream) { @@ -1108,7 +1108,7 @@ snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) snd_rme32_pcm_start(rme32, 0); break; case SNDRV_PCM_TRIGGER_STOP: - if (! rme32->running && RME32_ISWORKING(rme32)) + if (!rme32->running && RME32_ISWORKING(rme32)) snd_rme32_pcm_stop(rme32, 0); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -1402,23 +1402,23 @@ static int __devinit snd_rme32_create(struct rme32 * rme32) 1, 1, &rme32->adat_pcm)) < 0) { return err; - } + } rme32->adat_pcm->private_data = rme32; rme32->adat_pcm->private_free = snd_rme32_free_adat_pcm; strcpy(rme32->adat_pcm->name, "Digi32 ADAT"); if (rme32->fullduplex_mode) { - snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_rme32_playback_adat_fd_ops); - snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme32_capture_adat_fd_ops); snd_pcm_lib_preallocate_pages_for_all(rme32->adat_pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), 0, RME32_MID_BUFFER_SIZE); rme32->adat_pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; } else { - snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_rme32_playback_adat_ops); - snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme32_capture_adat_ops); rme32->adat_pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; } @@ -1561,7 +1561,7 @@ static void __devinit snd_rme32_proc_init(struct rme32 * rme32) { struct snd_info_entry *entry; - if (! snd_card_proc_new(rme32->card, "rme32", &entry)) + if (!snd_card_proc_new(rme32->card, "rme32", &entry)) snd_info_set_text_ops(entry, rme32, snd_rme32_proc_read); } @@ -1697,9 +1697,9 @@ static int snd_rme32_info_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { "AutoSync", - "Internal 32.0kHz", - "Internal 44.1kHz", + static char *texts[4] = { "AutoSync", + "Internal 32.0kHz", + "Internal 44.1kHz", "Internal 48.0kHz" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 3fdd488..fb8767e 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1,9 +1,9 @@ /* * ALSA driver for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST audio - * interfaces + * interfaces * * Copyright (c) 2000, 2001 Anders Torger <torger(a)ludd.luth.se> - * + * * Thanks to Henk Hesselink <henk(a)anda.nl> for the analog volume control * code. * @@ -21,7 +21,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #include <linux/delay.h> #include <linux/init.h> @@ -38,7 +38,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> /* note, two last pcis should be equal, it is not a bug */ @@ -205,7 +205,7 @@ struct rme96 { int irq; unsigned long port; void __iomem *iobase; - + u32 wcreg; /* cached write control register value */ u32 wcreg_spdif; /* S/PDIF setup */ u32 wcreg_spdif_stream; /* S/PDIF setup (temporary) */ @@ -220,13 +220,13 @@ struct rme96 { int playback_frlog; /* log2 of framesize */ int capture_frlog; - + size_t playback_periodsize; /* in bytes, zero if not used */ size_t capture_periodsize; /* in bytes, zero if not used */ struct snd_card *card; struct snd_pcm *spdif_pcm; - struct snd_pcm *adat_pcm; + struct snd_pcm *adat_pcm; struct pci_dev *pci; struct snd_kcontrol *spdif_ctl; }; @@ -239,7 +239,7 @@ static struct pci_device_id snd_rme96_ids[] = { { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, { 0, } }; @@ -262,11 +262,11 @@ static int snd_rme96_capture_prepare(struct snd_pcm_substream *substream); static int -snd_rme96_playback_trigger(struct snd_pcm_substream *substream, +snd_rme96_playback_trigger(struct snd_pcm_substream *substream, int cmd); static int -snd_rme96_capture_trigger(struct snd_pcm_substream *substream, +snd_rme96_capture_trigger(struct snd_pcm_substream *substream, int cmd); static snd_pcm_uframes_t @@ -275,7 +275,7 @@ snd_rme96_playback_pointer(struct snd_pcm_substream *substream); static snd_pcm_uframes_t snd_rme96_capture_pointer(struct snd_pcm_substream *substream); -static void __devinit +static void __devinit snd_rme96_proc_init(struct rme96 *rme96); static int @@ -355,10 +355,10 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info = .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -384,10 +384,10 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info = .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -412,7 +412,7 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info = SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), - .rates = (SNDRV_PCM_RATE_44100 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), .rate_min = 44100, .rate_max = 48000, @@ -437,7 +437,7 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info = SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), - .rates = (SNDRV_PCM_RATE_44100 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), .rate_min = 44100, .rate_max = 48000, @@ -573,7 +573,7 @@ snd_rme96_setattenuation(struct rme96 *rme96, static int snd_rme96_capture_getrate(struct rme96 *rme96, int *is_adat) -{ +{ int n, rate; *is_adat = 0; @@ -610,14 +610,14 @@ snd_rme96_capture_getrate(struct rme96 *rme96, if (rme96->rcreg & RME96_RCR_VERF) { return -1; } - + /* S/PDIF rate */ n = ((rme96->rcreg >> RME96_RCR_BITPOS_F0) & 1) + (((rme96->rcreg >> RME96_RCR_BITPOS_F1) & 1) << 1) + (((rme96->rcreg >> RME96_RCR_BITPOS_F2) & 1) << 2); - + switch (n) { - case 0: + case 0: if (rme96->rcreg & RME96_RCR_T_OUT) { return 64000; } @@ -763,7 +763,7 @@ snd_rme96_setclockmode(struct rme96 *rme96, { switch (mode) { case RME96_CLOCKMODE_SLAVE: - /* AutoSync */ + /* AutoSync */ rme96->wcreg &= ~RME96_WCR_MASTER; rme96->areg &= ~RME96_AR_WSEL; break; @@ -774,7 +774,7 @@ snd_rme96_setclockmode(struct rme96 *rme96, break; case RME96_CLOCKMODE_WORDCLOCK: /* Word clock is a master mode */ - rme96->wcreg |= RME96_WCR_MASTER; + rme96->wcreg |= RME96_WCR_MASTER; rme96->areg |= RME96_AR_WSEL; break; default: @@ -872,7 +872,7 @@ snd_rme96_setframelog(struct rme96 *rme96, int is_playback) { int frlog; - + if (n_channels == 2) { frlog = 1; } else { @@ -964,7 +964,7 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, /* slave clock */ if ((int)params_rate(params) != rate) { spin_unlock_irq(&rme96->lock); - return -EIO; + return -EIO; } } else if ((err = snd_rme96_playback_setrate(rme96, params_rate(params))) < 0) { spin_unlock_irq(&rme96->lock); @@ -992,7 +992,7 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER); } spin_unlock_irq(&rme96->lock); - + return 0; } @@ -1003,7 +1003,7 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream, struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err, isadat, rate; - + runtime->dma_area = (void __force *)(rme96->iobase + RME96_IO_REC_BUFFER); runtime->dma_addr = rme96->port + RME96_IO_REC_BUFFER; @@ -1024,7 +1024,7 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream, } else if ((rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0) { if ((int)params_rate(params) != rate) { spin_unlock_irq(&rme96->lock); - return -EIO; + return -EIO; } if ((isadat && runtime->hw.channels_min == 2) || (!isadat && runtime->hw.channels_min == 8)) @@ -1084,7 +1084,7 @@ snd_rme96_playback_stop(struct rme96 *rme96) rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); if (rme96->rcreg & RME96_RCR_IRQ) { writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ); - } + } rme96->wcreg &= ~RME96_WCR_START; writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); } @@ -1095,7 +1095,7 @@ snd_rme96_capture_stop(struct rme96 *rme96) rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); if (rme96->rcreg & RME96_RCR_IRQ_2) { writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ); - } + } rme96->wcreg &= ~RME96_WCR_START_2; writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); } @@ -1113,7 +1113,7 @@ snd_rme96_interrupt(int irq, { return IRQ_NONE; } - + if (rme96->rcreg & RME96_RCR_IRQ) { /* playback */ snd_pcm_period_elapsed(rme96->playback_substream); @@ -1121,7 +1121,7 @@ snd_rme96_interrupt(int irq, } if (rme96->rcreg & RME96_RCR_IRQ_2) { /* capture */ - snd_pcm_period_elapsed(rme96->capture_substream); + snd_pcm_period_elapsed(rme96->capture_substream); writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ); } return IRQ_HANDLED; @@ -1161,7 +1161,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - spin_lock_irq(&rme96->lock); + spin_lock_irq(&rme96->lock); if (rme96->playback_substream != NULL) { spin_unlock_irq(&rme96->lock); return -EBUSY; @@ -1180,7 +1180,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate); runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; - } + } rme96_set_buffer_size_constraint(rme96, runtime); rme96->wcreg_spdif_stream = rme96->wcreg_spdif; @@ -1208,7 +1208,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - + spin_lock_irq(&rme96->lock); if (rme96->capture_substream != NULL) { spin_unlock_irq(&rme96->lock); @@ -1216,7 +1216,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) } rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock); - + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1226,9 +1226,9 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) { int rate, dummy; struct rme96 *rme96 = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - spin_lock_irq(&rme96->lock); + struct snd_pcm_runtime *runtime = substream->runtime; + + spin_lock_irq(&rme96->lock); if (rme96->playback_substream != NULL) { spin_unlock_irq(&rme96->lock); return -EBUSY; @@ -1237,7 +1237,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); rme96->playback_substream = substream; spin_unlock_irq(&rme96->lock); - + runtime->hw = snd_rme96_playback_adat_info; if (!(rme96->wcreg & RME96_WCR_MASTER) && snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG && @@ -1247,7 +1247,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate); runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; - } + } rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1273,8 +1273,8 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - - spin_lock_irq(&rme96->lock); + + spin_lock_irq(&rme96->lock); if (rme96->capture_substream != NULL) { spin_unlock_irq(&rme96->lock); return -EBUSY; @@ -1292,7 +1292,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); int spdif = 0; - spin_lock_irq(&rme96->lock); + spin_lock_irq(&rme96->lock); if (RME96_ISPLAYING(rme96)) { snd_rme96_playback_stop(rme96); } @@ -1312,8 +1312,8 @@ static int snd_rme96_capture_close(struct snd_pcm_substream *substream) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); - - spin_lock_irq(&rme96->lock); + + spin_lock_irq(&rme96->lock); if (RME96_ISRECORDING(rme96)) { snd_rme96_capture_stop(rme96); } @@ -1327,8 +1327,8 @@ static int snd_rme96_playback_prepare(struct snd_pcm_substream *substream) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); - - spin_lock_irq(&rme96->lock); + + spin_lock_irq(&rme96->lock); if (RME96_ISPLAYING(rme96)) { snd_rme96_playback_stop(rme96); } @@ -1341,8 +1341,8 @@ static int snd_rme96_capture_prepare(struct snd_pcm_substream *substream) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); - - spin_lock_irq(&rme96->lock); + + spin_lock_irq(&rme96->lock); if (RME96_ISRECORDING(rme96)) { snd_rme96_capture_stop(rme96); } @@ -1352,7 +1352,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream) } static int -snd_rme96_playback_trigger(struct snd_pcm_substream *substream, +snd_rme96_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); @@ -1387,7 +1387,7 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream, snd_rme96_playback_start(rme96, 1); } break; - + default: return -EINVAL; } @@ -1395,7 +1395,7 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream, } static int -snd_rme96_capture_trigger(struct snd_pcm_substream *substream, +snd_rme96_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); @@ -1430,7 +1430,7 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream, snd_rme96_capture_start(rme96, 1); } break; - + default: return -EINVAL; } @@ -1573,8 +1573,8 @@ snd_rme96_create(struct rme96 *rme96) rme96->irq = pci->irq; /* read the card's revision number */ - pci_read_config_byte(pci, 8, &rme96->rev); - + pci_read_config_byte(pci, 8, &rme96->rev); + /* set up ALSA pcm device for S/PDIF */ if ((err = snd_pcm_new(rme96->card, "Digi96 IEC958", 0, 1, 1, &rme96->spdif_pcm)) < 0) @@ -1598,23 +1598,23 @@ snd_rme96_create(struct rme96 *rme96) 1, 1, &rme96->adat_pcm)) < 0) { return err; - } + } rme96->adat_pcm->private_data = rme96; rme96->adat_pcm->private_free = snd_rme96_free_adat_pcm; strcpy(rme96->adat_pcm->name, "Digi96 ADAT"); snd_pcm_set_ops(rme96->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_rme96_playback_adat_ops); snd_pcm_set_ops(rme96->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme96_capture_adat_ops); - + rme96->adat_pcm->info_flags = 0; } rme96->playback_periodsize = 0; rme96->capture_periodsize = 0; - + /* make sure playback/capture is stopped, if by some reason active */ snd_rme96_playback_stop(rme96); snd_rme96_capture_stop(rme96); - + /* set default values in registers */ rme96->wcreg = RME96_WCR_FREQ_1 | /* set 44.1 kHz playback */ @@ -1626,11 +1626,11 @@ snd_rme96_create(struct rme96 *rme96) writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); - + /* reset the ADC */ writel(rme96->areg | RME96_AR_PD2, rme96->iobase + RME96_IO_ADDITIONAL_REG); - writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); /* reset and enable the DAC (order is important). */ snd_rme96_reset_dac(rme96); @@ -1646,7 +1646,7 @@ snd_rme96_create(struct rme96 *rme96) if (RME96_HAS_ANALOG_OUT(rme96)) { snd_rme96_apply_dac_volume(rme96); } - + /* init switch interface */ if ((err = snd_rme96_create_switches(rme96->card, rme96)) < 0) { return err; @@ -1654,7 +1654,7 @@ snd_rme96_create(struct rme96 *rme96) /* init proc interface */ snd_rme96_proc_init(rme96); - + return 0; } @@ -1662,12 +1662,12 @@ snd_rme96_create(struct rme96 *rme96) * proc interface */ -static void +static void snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { int n; struct rme96 *rme96 = (struct rme96 *)entry->private_data; - + rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); snd_iprintf(buffer, rme96->card->longname); @@ -1681,7 +1681,7 @@ snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer snd_iprintf(buffer, " period size: 2048 bytes\n"); } else { snd_iprintf(buffer, " period size: 8192 bytes\n"); - } + } snd_iprintf(buffer, "\nInput settings\n"); switch (snd_rme96_getinputtype(rme96)) { case RME96_INPUT_OPTICAL: @@ -1716,7 +1716,7 @@ snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer } else { snd_iprintf(buffer, " sample format: 16 bit\n"); } - + snd_iprintf(buffer, "\nOutput settings\n"); if (rme96->wcreg & RME96_WCR_SEL) { snd_iprintf(buffer, " output signal: normal playback\n"); @@ -1791,12 +1791,12 @@ snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer } } -static void __devinit +static void __devinit snd_rme96_proc_init(struct rme96 *rme96) { struct snd_info_entry *entry; - if (! snd_card_proc_new(rme96->card, "rme96", &entry)) + if (!snd_card_proc_new(rme96->card, "rme96", &entry)) snd_info_set_text_ops(entry, rme96, snd_rme96_proc_read); } @@ -1810,7 +1810,7 @@ static int snd_rme96_get_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme96->lock); ucontrol->value.integer.value[0] = rme96->wcreg & RME96_WCR_SEL ? 0 : 1; spin_unlock_irq(&rme96->lock); @@ -1822,7 +1822,7 @@ snd_rme96_put_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_ele struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); unsigned int val; int change; - + val = ucontrol->value.integer.value[0] ? 0 : RME96_WCR_SEL; spin_lock_irq(&rme96->lock); val = (rme96->wcreg & ~RME96_WCR_SEL) | val; @@ -1839,7 +1839,7 @@ snd_rme96_info_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_e static char *_texts[5] = { "Optical", "Coaxial", "Internal", "XLR", "Analog" }; struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); char *texts[5] = { _texts[0], _texts[1], _texts[2], _texts[3], _texts[4] }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; switch (rme96->pci->device) { @@ -1875,10 +1875,10 @@ snd_rme96_get_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); unsigned int items = 3; - + spin_lock_irq(&rme96->lock); ucontrol->value.enumerated.item[0] = snd_rme96_getinputtype(rme96); - + switch (rme96->pci->device) { case PCI_DEVICE_ID_RME_DIGI96: case PCI_DEVICE_ID_RME_DIGI96_8: @@ -1905,7 +1905,7 @@ snd_rme96_get_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el if (ucontrol->value.enumerated.item[0] >= items) { ucontrol->value.enumerated.item[0] = items - 1; } - + spin_unlock_irq(&rme96->lock); return 0; } @@ -1915,7 +1915,7 @@ snd_rme96_put_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); unsigned int val; int change, items = 3; - + switch (rme96->pci->device) { case PCI_DEVICE_ID_RME_DIGI96: case PCI_DEVICE_ID_RME_DIGI96_8: @@ -1936,14 +1936,14 @@ snd_rme96_put_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el break; } val = ucontrol->value.enumerated.item[0] % items; - + /* special case for PST */ if (rme96->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST && rme96->rev > 4) { if (val == RME96_INPUT_XLR) { val = RME96_INPUT_ANALOG; } } - + spin_lock_irq(&rme96->lock); change = (int)val != snd_rme96_getinputtype(rme96); snd_rme96_setinputtype(rme96, val); @@ -1955,7 +1955,7 @@ static int snd_rme96_info_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[3] = { "AutoSync", "Internal", "Word" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -1969,7 +1969,7 @@ static int snd_rme96_get_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme96->lock); ucontrol->value.enumerated.item[0] = snd_rme96_getclockmode(rme96); spin_unlock_irq(&rme96->lock); @@ -1981,7 +1981,7 @@ snd_rme96_put_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); unsigned int val; int change; - + val = ucontrol->value.enumerated.item[0] % 3; spin_lock_irq(&rme96->lock); change = (int)val != snd_rme96_getclockmode(rme96); @@ -1994,7 +1994,7 @@ static int snd_rme96_info_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[4] = { "0 dB", "-6 dB", "-12 dB", "-18 dB" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 4; @@ -2008,7 +2008,7 @@ static int snd_rme96_get_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme96->lock); ucontrol->value.enumerated.item[0] = snd_rme96_getattenuation(rme96); spin_unlock_irq(&rme96->lock); @@ -2020,7 +2020,7 @@ snd_rme96_put_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); unsigned int val; int change; - + val = ucontrol->value.enumerated.item[0] % 4; spin_lock_irq(&rme96->lock); @@ -2034,7 +2034,7 @@ static int snd_rme96_info_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[4] = { "1+2", "3+4", "5+6", "7+8" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 4; @@ -2048,7 +2048,7 @@ static int snd_rme96_get_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme96->lock); ucontrol->value.enumerated.item[0] = snd_rme96_getmontracks(rme96); spin_unlock_irq(&rme96->lock); @@ -2060,7 +2060,7 @@ snd_rme96_put_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); unsigned int val; int change; - + val = ucontrol->value.enumerated.item[0] % 4; spin_lock_irq(&rme96->lock); change = (int)val != snd_rme96_getmontracks(rme96); @@ -2101,7 +2101,7 @@ static int snd_rme96_control_spdif_info(struct snd_kcontrol *kcontrol, struct sn static int snd_rme96_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + snd_rme96_convert_to_aes(&ucontrol->value.iec958, rme96->wcreg_spdif); return 0; } @@ -2111,7 +2111,7 @@ static int snd_rme96_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_rme96_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&rme96->lock); change = val != rme96->wcreg_spdif; @@ -2130,7 +2130,7 @@ static int snd_rme96_control_spdif_stream_info(struct snd_kcontrol *kcontrol, st static int snd_rme96_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + snd_rme96_convert_to_aes(&ucontrol->value.iec958, rme96->wcreg_spdif_stream); return 0; } @@ -2140,7 +2140,7 @@ static int snd_rme96_control_spdif_stream_put(struct snd_kcontrol *kcontrol, str struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_rme96_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&rme96->lock); change = val != rme96->wcreg_spdif_stream; @@ -2169,7 +2169,7 @@ static int snd_rme96_dac_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; @@ -2258,9 +2258,9 @@ static struct snd_kcontrol_new snd_rme96_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Connector", - .info = snd_rme96_info_inputtype_control, + .info = snd_rme96_info_inputtype_control, .get = snd_rme96_get_inputtype_control, - .put = snd_rme96_put_inputtype_control + .put = snd_rme96_put_inputtype_control }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2272,21 +2272,21 @@ static struct snd_kcontrol_new snd_rme96_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Sample Clock Source", - .info = snd_rme96_info_clockmode_control, + .info = snd_rme96_info_clockmode_control, .get = snd_rme96_get_clockmode_control, .put = snd_rme96_put_clockmode_control }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitor Tracks", - .info = snd_rme96_info_montracks_control, + .info = snd_rme96_info_montracks_control, .get = snd_rme96_get_montracks_control, .put = snd_rme96_put_montracks_control }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Attenuation", - .info = snd_rme96_info_attenuation_control, + .info = snd_rme96_info_attenuation_control, .get = snd_rme96_get_attenuation_control, .put = snd_rme96_put_attenuation_control }, @@ -2318,7 +2318,7 @@ snd_rme96_create_switches(struct snd_card *card, if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_rme96_controls[idx], rme96))) < 0) return err; } - + return 0; } @@ -2352,7 +2352,7 @@ snd_rme96_probe(struct pci_dev *pci, sizeof(struct rme96))) == NULL) return -ENOMEM; card->private_free = snd_rme96_card_free; - rme96 = (struct rme96 *)card->private_data; + rme96 = (struct rme96 *)card->private_data; rme96->card = card; rme96->pci = pci; snd_card_set_dev(card, &pci->dev); @@ -2360,7 +2360,7 @@ snd_rme96_probe(struct pci_dev *pci, snd_card_free(card); return err; } - + strcpy(card->driver, "Digi96"); switch (rme96->pci->device) { case PCI_DEVICE_ID_RME_DIGI96: @@ -2383,10 +2383,10 @@ snd_rme96_probe(struct pci_dev *pci, } sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, rme96->port, rme96->irq); - + if ((err = snd_card_register(card)) < 0) { snd_card_free(card); - return err; + return err; } pci_set_drvdata(pci, card); dev++;

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