January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] HDA - hdmi_show_short_audio_desc puts wrong rates in eld#N
by Anssi Hannula 11 Jan '11

11 Jan '11

Hi! Just noticed that the sample rates reported in eld#X.Y are wrong. They are printed by hdmi_show_short_audio_desc() in hda_eld.c. That function uses the snd_print_pcm_rates() function to convert the rates bitmap to a string. However, snd_print_pcm_rates() takes a HDA rates bitmap, not a generic SNDRV_ one. (BTW, an interesting function name for an internal HDA thing...) This results in several errors. For example, my 32000 44100 48000 88200 96000 176400 192000 becomes 44100 48000 88200 176400 192000 384000. -- Anssi Hannula

1 1

[alsa-devel] UCM updates and fixes
by Liam Girdwood 11 Jan '11

11 Jan '11

Jarolsav, This fixes the remaining issues I can currently see in the alsa-lib ucm branch. It also includes a patch to execute commands, although I dont know if this is what you had in mind for exec. I've also done quite a bit of testing now. What is your merge plan for UCM ? We really need UCM merged soon as there are many users who depend on it. Thanks Liam

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Re: [alsa-devel] Bug in setting channels, related to recent ELD/EDID changes?
by Anssi Hannula 11 Jan '11

11 Jan '11

(restored CC I removed, sorry about that) On 10.01.2011 22:36, Takashi Iwai wrote: > At Mon, 10 Jan 2011 21:58:26 +0200, > Anssi Hannula wrote: >> >> >> Takashi Iwai kirjoitti: >>> At Sun, 02 Jan 2011 12:16:44 +0200, >>> Anssi Hannula wrote: >>>> >>>> Hi! >>>> >>>> Takashi, it seems the "allow all when not plugged in" behaviour you >>>> implemented in bbbe3390 is not enough to allow applications to open the >>>> device before the output device is ready. Switched-off A/V receivers may >>>> provide the information of the plugged-in television which usually >>>> restricts the max_channels to 2, causing applications to fail opening >>>> the device with more channels when a the A/V receiver is switched off. >>>> >>>> Also, even if the application can gracefully fallback to stereo, AFAIK >>>> there is no method for an application to get informed when the >>>> limitation is lifted, so it can't automatically resume multichannel >>>> output when the A/V receiver is switched on again. >>>> >>>> Based on this info, I guess the restrictions based on ELD should be >>>> removed :/ >>>> Unless you have some other ideas to fix the issue, of course. >>> >>> What about the patch below? If it's unplugged, the valid flag should >>> be cleared, so the next open fall backs to the default state (i.e. >>> allows all). >> >> But it is not unplugged. The multichannel-capable receiver forwards EDID >> of TV which doesn't allow more than 2 channels, when the receiver is >> swtiched off. > > Well, in that case, this is the correct behavior, IMO. At this > moment, setting channels to 2 is the right thing. Well, I don't like this at all for two reasons: 1) This common use case worked with earlier kernel revisions: You could start playback of a video / TV channel, then go sit on a couch and start up the A/V receiver to get surround sound. Now it is broken. 2) There is no way this can be fixed in the libalsa-using application either, as it doesn't receive a notification when more channels become available after it has already fallbacked to stereo. So we got something reasonable that worked before but not with 2.6.36+. > Then, how can you > know that more channels will be available in future at all? We don't. However, we don't know that about any analog line-outs either, yet we always allow all channels no matter what amount of speakers are plugged in (AFAIK) (assuming jack detection support). > What you want to have a static configuration way by ignoring the ELD > information. But, the interface isn't pretty easy if it's implemented > in control API, I suppose. > > > Takashi > >> >>> Takashi >>> >>> --- >>> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c >>> index d1b1b57..27e8597 100644 >>> --- a/sound/pci/hda/patch_hdmi.c >>> +++ b/sound/pci/hda/patch_hdmi.c >>> @@ -827,7 +827,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, >>> *codec_pars = *hinfo; >>> >>> eld = &spec->sink_eld[idx]; >>> - if (eld->sad_count > 0) { >>> + if (eld->eld_valid && eld->sad_count > 0) { >>> hdmi_eld_update_pcm_info(eld, hinfo, codec_pars); >>> if (hinfo->channels_min > hinfo->channels_max || >>> !hinfo->rates || !hinfo->formats) >>> >> >> >> -- >> Anssi Hannula >> -- Anssi Hannula

2 2

[alsa-devel] [PATCH 0/7] Tegra ALSA/ASoC driver
by Stephen Warren 11 Jan '11

11 Jan '11

(My apologies; I sent this to both linux-tegra and alsa-devel mailing lists before, but posted from the wrong address, so it got dropped from the ALSA list. Here's a copy, hopefully threaded back into the same conversation as the original copy which made it to linux-tegra) This patch series implements an ALSA/ASoC driver for Tegra, and implements minimal machine support for the Harmony reference board. I believe I've addressed all the feedback from Mark re: the previous patchset. Stephen Warren (7): ASoC: Export debugfs root dentry ASoC: tegra: Add tegra-das driver ASoC: tegra: Add tegra-pcm driver ASoC: tegra: Add tegra-i2s driver ASoC: tegra: Machine utility code ASoC: tegra: Harmony machine support ASoC: tegra: Kconfig and Makefile include/sound/soc.h | 4 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/soc-core.c | 19 +- sound/soc/tegra/Kconfig | 27 ++ sound/soc/tegra/Makefile | 14 + sound/soc/tegra/harmony.c | 179 +++++++++++++ sound/soc/tegra/tegra_asoc_utils.c | 154 +++++++++++ sound/soc/tegra/tegra_asoc_utils.h | 31 +++ sound/soc/tegra/tegra_das.c | 263 +++++++++++++++++++ sound/soc/tegra/tegra_das.h | 135 ++++++++++ sound/soc/tegra/tegra_i2s.c | 502 ++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_i2s.h | 165 ++++++++++++ sound/soc/tegra/tegra_pcm.c | 401 ++++++++++++++++++++++++++++ sound/soc/tegra/tegra_pcm.h | 55 ++++ 15 files changed, 1942 insertions(+), 9 deletions(-) create mode 100644 sound/soc/tegra/Kconfig create mode 100644 sound/soc/tegra/Makefile create mode 100644 sound/soc/tegra/harmony.c create mode 100644 sound/soc/tegra/tegra_asoc_utils.c create mode 100644 sound/soc/tegra/tegra_asoc_utils.h create mode 100644 sound/soc/tegra/tegra_das.c create mode 100644 sound/soc/tegra/tegra_das.h create mode 100644 sound/soc/tegra/tegra_i2s.c create mode 100644 sound/soc/tegra/tegra_i2s.h create mode 100644 sound/soc/tegra/tegra_pcm.c create mode 100644 sound/soc/tegra/tegra_pcm.h

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[alsa-devel] [PATCH] Support for Digidesign Mbox 2 duplex analogue mode.
by damien.zammit＠gmail.com 11 Jan '11

11 Jan '11

From: Damien Zammit <damien.zammit(a)gmail.com> Added suggestions by Takashi Iwai (see previous post on Mbox 2). Some lines are still longer than 80 characters, but all other warnings have been fixed. Signed-off-by: Damien Zammit <damien.zammit(a)gmail.com> --- sound/usb/midi.c | 12 ++++ sound/usb/quirks-table.h | 51 +++++++++++++++++ sound/usb/quirks.c | 137 ++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 ++ sound/usb/usbaudio.h | 1 + 5 files changed, 206 insertions(+), 0 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 25bce7e..5dace7e 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2122,6 +2122,18 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_MBOX2: + /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ + umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; + /* + * We have to make sure that the USB core looks + * again at interface 6 by calling usb_set_interface() on it. + */ + usb_set_interface(umidi->dev, 6, 0); + memcpy(&endpoints[0], quirk->data, + sizeof(struct snd_usb_midi_endpoint_info)); + err = 0; + break; case QUIRK_MIDI_RAW_BYTES: umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; /* diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index ad7079d..ba61dfe 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2477,6 +2477,57 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* DIGIDESIGN MBOX 2 */ +{ + USB_DEVICE(0x0dba, 0x3000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 2", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 5, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 6, + .type = QUIRK_MIDI_MBOX2, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_ep = 0x02, + .out_cables = 0x0001, + .in_ep = 0x81, + .in_interval = 0x01, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index cf8bf08..483a959 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -18,6 +18,7 @@ #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> +#include <linux/wait.h> #include <sound/core.h> #include <sound/info.h> @@ -287,6 +288,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, + [QUIRK_MIDI_MBOX2] = create_any_midi_quirk, [QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, @@ -479,6 +481,9 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x0763, 0x2003)) return audiophile_skip_setting_quirk(chip, iface, altno); + if (chip->usb_id == USB_ID(0x0dba, 0x3000)) + return mbox2_skip_setting_quirk(chip, iface, altno); + return 0; } @@ -506,6 +511,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, if (id == USB_ID(0x0d8c, 0x0102)) return snd_usb_cm6206_boot_quirk(dev); + /* Digidesign Mbox 2 */ + if (id == USB_ID(0x0dba, 0x3000)) + return snd_usb_mbox2_boot_quirk(dev); + /* Access Music VirusTI Desktop */ if (id == USB_ID(0x133e, 0x0815)) return snd_usb_accessmusic_boot_quirk(dev); @@ -527,6 +536,9 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat if (chip->setup == 0x00 || fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3) return 1; + break; + case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ + return 1; } return 0; } @@ -594,3 +606,128 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } +#define MBOX2_FIRMWARE_SIZE 646 +#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ +#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ + +static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + wait_queue_head_t mbox2_wait_queue; + u8 enablemagic[3]; + u8 bootresponse; + u8 temp[3]; + int fwsize; + int count; + + init_waitqueue_head(&mbox2_wait_queue); + fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (fwsize == MBOX2_FIRMWARE_SIZE) { + snd_printk(KERN_INFO "Sending Digidesign Mbox 2 boot sequence...\n"); + /* From USB Snoop, + * + * SetupPacket = RT RQ VHVL INDX SIZE + * RT = 0xRT Request Type + * RQ = 0xRQ Request + * VHVL = 0xVLVH Value in reverse byte order + * INDX = 0xDXIN Index in reverse byte order + * SIZE = 0xZESI Size in reverse byte order + * + * Magic boot code setup packet: c0 85 01 00 00 00 12 00 + * RQ RT VLVH DXIN ZESI + * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT + * for snd_usb_ctl_msg() + */ + + count = 0; + bootresponse = MBOX2_BOOT_LOADING; + while ((bootresponse == MBOX2_BOOT_LOADING) && (count < 5)) { + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x85, 0xc0, 0x0001, 0x0000, + &bootresponse, 0x0012, 1000); + if (bootresponse == MBOX2_BOOT_READY) + break; + + snd_printk(KERN_INFO "device not ready, resending boot sequence...\n"); + count++; + /* 1 second delay */ + wait_event_timeout(mbox2_wait_queue, + bootresponse == MBOX2_BOOT_READY, + msecs_to_jiffies(1000)); + } + + if (bootresponse == MBOX2_BOOT_READY) { + snd_printk(KERN_INFO "device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + snd_printk(KERN_ERR "error usb_get_descriptor: %d\n", err); + err = usb_reset_configuration(dev); + if (err < 0) + snd_printk(KERN_ERR "error usb_reset_configuration: %d\n", err); + snd_printk("mbox2_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + /* We want 48 kHz mode for now */ + enablemagic[0] = 0x80; + enablemagic[1] = 0xbb; + enablemagic[2] = 0x00; + + /* Send the magic! */ + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x01, 0x22, 0x0100, 0x0085, + &temp, 0x0003, 1000); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0085, + &enablemagic, 0x0003, 1000); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0086, + &enablemagic, 0x0003, 1000); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0003, + &enablemagic, 0x0003, 1000); + snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48kHz Analogue"); + + return 0; /* Succesful boot */ + } + + snd_printk(KERN_ERR "Unknown bootresponse, or timed out, ignoring device: %d\n", bootresponse); + return -ENODEV; + } + snd_printk(KERN_ERR "Invalid firmware size: %d\n", fwsize); + return -ENODEV; +} + +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + u8 srate[3]; + u8 temp[3]; + + /* Reset ifaces 2,4,5 to 0 altsetting. */ + usb_set_interface(chip->dev, iface, 0); + + /* setup 48k */ + srate[0] = 0x80; + srate[1] = 0xbb; + srate[2] = 0x00; + + /* Send the magic! */ + snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), + 0x01, 0x22, 0x0100, 0x0085, &temp, 0x0003, 1000); + snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), + 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); + snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), + 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); + snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), + 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); + + if (altno != 3) + return -1; + + return 0; +} diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 03e5e94..1643f3f 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,4 +20,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp); +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno); + +static int snd_usb_mbox2_boot_quirk(struct usb_device *dev); + #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index db3eb21..128c3f4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -70,6 +70,7 @@ enum quirk_type { QUIRK_MIDI_YAMAHA, QUIRK_MIDI_MIDIMAN, QUIRK_MIDI_NOVATION, + QUIRK_MIDI_MBOX2, QUIRK_MIDI_RAW_BYTES, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, -- 1.6.3.3

1 0

[alsa-devel] [PATCH] ASoC: Fix indentation in soc_remove()
by Mark Brown 11 Jan '11

11 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96bf381..ec942a2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1905,7 +1905,7 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int i; - if (card->instantiated) { + if (card->instantiated) { /* make sure any delayed work runs */ for (i = 0; i < card->num_rtd; i++) { -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: soc-core: Simplify compress_type overriding functionality
by Dimitris Papastamos 11 Jan '11

11 Jan '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 9 +-------- 1 files changed, 1 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2ff708a..13c4092 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1761,15 +1761,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (i == card->num_configs) { /* no need to override the compress_type so * go ahead and do the standard thing */ - ret = snd_soc_init_codec_cache(codec, 0); - if (ret < 0) { - mutex_unlock(&card->mutex); - return; - } - continue; + compress_type = 0; } - /* override the compress_type with the one supplied in - * the machine driver */ ret = snd_soc_init_codec_cache(codec, compress_type); if (ret < 0) { mutex_unlock(&card->mutex); -- 1.7.3.5

2 1

[alsa-devel] [PATCH 0/4] ASoC: SAMSUNG: Code clean-up suitable for current ASoC
by Seungwhan Youn 11 Jan '11

11 Jan '11

Hi, This patch remove(or move) unnecessary or un-suitable features on current ASoC for all Samsung audio drivers and its supports. After multi-component model patch, ASoC core part controls dai_links instead of _hard_wired_ dai links on machine drivers. Due to apply this model, many machine drivers don't need codecs and platforms includes to make _hard_wired_ dai_links. And also, ASoC core header file 'sound/soc.h' has basic includes for device driver, so many of includes in each machine/platform driver are duplicated and not essentially needed. So, this patch-set check each platform/machine drivers' includes and definitions are suitable or not, remove unnecessary features with compile tests. These patches are modify below features:- o Remove unnecessary 'ac97.h' header file. o Move each specific definitions, not to be shared by header files. o Clean-up unnecessary header includes. This patch-set is based on two different branches that :- o ASoC patches are based on Mark Brown's git branch 'for-next' (commit id - f6c2ed5dd6ab43447dacc136585fc894e3f3a82d) This patch-set contains followings :- o [PATCH 1/4] ASoC: SAMSUNG: Remove AC97 header file o [PATCH 2/4] ASoC: SAMSUNG: Clean-up DMA header file o [PATCH 3/4] ASoC: SAMSUNG: Move PCM specific definitions into pcm.c o [PATCH 4/4] ASoC: SAMSUNG: Clean-up header includes Thanks, Claude(Seungwhan Youn)

5 17

[alsa-devel] [PATCH] ASoC: WM8995: Remember to flush the cache on resume
by Dimitris Papastamos 11 Jan '11

11 Jan '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8995.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 3d2110c..96e0dc0 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1498,6 +1498,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, 0); + codec->cache_sync = 1; break; } -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: sst_platform created helper functions
by Harsha, Priya 11 Jan '11

11 Jan '11

From: Harsha Priya <priya.harsha(a)intel.com> Few funtions can be modularized in this driver to make them look cleaner like managing the stream status with locks and filling pcm parameters. This patch adds helper functions to do the same. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/sst_platform.c | 123 ++++++++++++++++--------------------- 1 files changed, 53 insertions(+), 70 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 0e994c1..a4e3fa3 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -105,7 +105,45 @@ struct snd_soc_dai_driver sst_platform_dai[] = { }, }, }; + /* helper functions */ +static inline void sst_set_stream_status(struct sst_runtime_stream *stream, + int state) +{ + spin_lock(&stream->status_lock); + stream->stream_status = state; + spin_unlock(&stream->status_lock); +} + +static inline int sst_get_stream_status(struct sst_runtime_stream *stream) +{ + int state; + + spin_lock(&stream->status_lock); + state = stream->stream_status; + spin_unlock(&stream->status_lock); + return state; +} + +static void sst_fill_pcm_params(struct snd_pcm_substream *substream, + struct snd_sst_stream_params *param) +{ + + param->uc.pcm_params.codec = SST_CODEC_TYPE_PCM; + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.reserved = 0; + param->uc.pcm_params.sfreq = substream->runtime->rate; + param->uc.pcm_params.ring_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + param->uc.pcm_params.period_count = substream->runtime->period_size; + param->uc.pcm_params.ring_buffer_addr = + virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->uc.pcm_params.period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", + param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz); +} + static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = @@ -115,26 +153,10 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) int ret_val; /* set codec params and inform SST driver the same */ - - param.uc.pcm_params.codec = SST_CODEC_TYPE_PCM; - param.uc.pcm_params.num_chan = (u8) substream->runtime->channels; - param.uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; - param.uc.pcm_params.reserved = 0; - param.uc.pcm_params.sfreq = substream->runtime->rate; - param.uc.pcm_params.ring_buffer_size = - snd_pcm_lib_buffer_bytes(substream); - param.uc.pcm_params.period_count = substream->runtime->period_size; - param.uc.pcm_params.ring_buffer_addr = - virt_to_phys(substream->dma_buffer.area); + sst_fill_pcm_params(substream, &param); substream->runtime->dma_area = substream->dma_buffer.area; - - pr_debug("period_cnt = %d\n", param.uc.pcm_params.period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", - param.uc.pcm_params.sfreq, param.uc.pcm_params.pcm_wd_sz); - str_params.sparams = param; - str_params.codec = SST_CODEC_TYPE_PCM; - + str_params.codec = param.uc.pcm_params.codec; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { str_params.ops = STREAM_OPS_PLAYBACK; str_params.device_type = substream->pcm->device + 1; @@ -153,28 +175,23 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) stream->stream_info.str_id = ret_val; pr_debug("str id : %d\n", stream->stream_info.str_id); - return ret_val; } - static void sst_period_elapsed(void *mad_substream) { struct snd_pcm_substream *substream = mad_substream; struct sst_runtime_stream *stream; + int status; if (!substream || !substream->runtime) return; stream = substream->runtime->private_data; if (!stream) return; - - spin_lock(&stream->status_lock); - if (stream->stream_status != SST_PLATFORM_RUNNING) { - spin_unlock(&stream->status_lock); + status = sst_get_stream_status(stream); + if (status != SST_PLATFORM_RUNNING) return; - } - spin_unlock(&stream->status_lock); snd_pcm_period_elapsed(substream); } @@ -185,9 +202,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) int ret_val; pr_debug("setting buffer ptr param\n"); - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_INIT; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; stream->stream_info.mad_substream = substream; stream->stream_info.buffer_ptr = 0; @@ -208,21 +223,14 @@ static int sst_platform_open(struct snd_pcm_substream *substream) int ret_val = 0; pr_debug("sst_platform_open called\n"); - runtime = substream->runtime; runtime->hw = sst_platform_pcm_hw; - stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; - spin_lock_init(&stream->status_lock); stream->stream_info.str_id = 0; - - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_INIT; - spin_unlock(&stream->status_lock); - + sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.mad_substream = substream; /* allocate memory for SST API set */ stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), @@ -233,7 +241,6 @@ static int sst_platform_open(struct snd_pcm_substream *substream) return -ENOMEM; } stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; - /* registering with SST driver to get access to SST APIs to use */ ret_val = register_sst_card(stream->sstdrv_ops); if (ret_val) { @@ -251,14 +258,11 @@ static int sst_platform_close(struct snd_pcm_substream *substream) int ret_val = 0, str_id; pr_debug("sst_platform_close called\n"); - stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; - if (str_id) ret_val = stream->sstdrv_ops->control_set( SST_SND_FREE, &str_id); - kfree(stream->sstdrv_ops); kfree(stream); return ret_val; @@ -270,7 +274,6 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) int ret_val = 0, str_id; pr_debug("sst_platform_pcm_prepare called\n"); - stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -299,11 +302,8 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, struct sst_runtime_stream *stream; pr_debug("sst_platform_pcm_trigger called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: pr_debug("sst: Trigger Start\n"); @@ -311,9 +311,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_START, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_RUNNING; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_RUNNING); stream->stream_info.mad_substream = substream; break; case SNDRV_PCM_TRIGGER_STOP: @@ -322,9 +320,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_DROP, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_DROPPED; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_DROPPED); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); @@ -332,9 +328,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_PAUSE, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_PAUSED; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_PAUSED); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); @@ -342,9 +336,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_RESUME, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_RUNNING; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_RUNNING); break; default: ret_val = -EINVAL; @@ -357,18 +349,13 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream; - int ret_val; + int ret_val, status; struct pcm_stream_info *str_info; - stream = substream->runtime->private_data; - spin_lock(&stream->status_lock); - if (stream->stream_status == SST_PLATFORM_INIT) { - spin_unlock(&stream->status_lock); + status = sst_get_stream_status(stream); + if (status == SST_PLATFORM_INIT) return 0; - } - spin_unlock(&stream->status_lock); - str_info = &stream->stream_info; ret_val = stream->sstdrv_ops->control_set( SST_SND_BUFFER_POINTER, str_info); @@ -376,7 +363,6 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; } @@ -402,7 +388,6 @@ int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, int retval = 0; pr_debug("sst_pcm_new called\n"); - if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, @@ -414,7 +399,6 @@ int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, return retval; } } - return retval; } struct snd_soc_platform_driver sst_soc_platform_drv = { @@ -433,6 +417,7 @@ static int sst_platform_probe(struct platform_device *pdev) pr_err("registering soc platform failed\n"); return ret; } + ret = snd_soc_register_dais(&pdev->dev, sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); if (ret) { @@ -446,10 +431,8 @@ static int sst_platform_remove(struct platform_device *pdev) { snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai)); - snd_soc_unregister_platform(&pdev->dev); pr_debug("sst_platform_remove sucess\n"); - return 0; } -- 1.7.3.4

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