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

[alsa-devel] CX20585 - selecting wrong digital output
by Andy Robinson 06 Jan '11

06 Jan '11

I opened bug 0005235 in the bug tracking system regarding no output from my laptop's (Asus U50F) mini-TOSLINK jack, even though ALSA was detecting and providing a SPDIF output device. Using the alsa-info output, somebody was able to diagnose the issue, and I was able to confirm the problem. It looks like the CX20585 has two digital outputs: Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Since the CX20585 uses the CXT5066 patch in ALSA, node 0x21 is used for the SPDIF device. However, at least in the case of my laptop, that digital output is connected to a pin complex which is N/A: Pin Default 0x400001f0: [N/A] Line Out at Ext N/A However, node 0x12 IS connected to the optical out jack: Pin Default 0x034511f0: [Jack] SPDIF Out at Ext Left So, in this case, ALSA should use node 0x12 instead of node 0x21 for SPDIF out. I confirmed this with the latest snapshot by changing the digital output from 0x21 to 0x12, and that solved the problem. If there are no volunteers that want to take a look at a "real" solution for this :-), I might take a crack at it myself, but I'm not all that familiar with the driver right now. There seems to be different ways this could be handled: (1) Introduce a model that would use 0x12 instead of 0x21 as the digital output (2) Check the pin complexes associated with outputs 0x12 and 0x21 to determine which one is actually connected to a jack, and use that output as the SPDIF device instead of always choosing 0x21 (3) What if both digital outputs are connected to a valid jack? Should ALSA then create devices for both? As of now, it's not even creating a device for 0x12 even though it's a valid audio output. Any comments or suggestions are welcome.

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[alsa-devel] [PATCH] ASoC: Taint the kernel if debugfs is used to write directly to CODECs
by Mark Brown 06 Jan '11

06 Jan '11

Since direct register writes may confuse the drivers and are supposed to be used only in diagnostic situations discourage their use in production by tainting the kernel when we do a write. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a233607..dbca5023 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -209,6 +209,10 @@ static ssize_t codec_reg_write_file(struct file *file, start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; + + /* Userspace has been fiddling around behind the kernel's back */ + add_taint(TAINT_USER); + snd_soc_write(codec, reg, value); return buf_size; } -- 1.7.2.3

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[alsa-devel] Do I have to use plugin?
by Radivoje Jovanovic 06 Jan '11

06 Jan '11

Hi, I have a system that is slow and cannot support overhead for re-sampling input streams. The codec on the system supports all rates from 8K-48K and it is continuous so pretty much any rate should work no problem. However, I noticed that when I play a song that has weird rate that is not specified in the constraint list aplay would come back saying that it cannot set hardware parameters. If I play the same song using the virtual device through "plug" plugin the thing works no problem. I guess that alsa-lib re-samples the data to rate specified in the virtual device. Is there any way for me to force alsa to forward any rate to the hardware driver? On the other hand is there any way I can specify multiple rates in the plug? For example my plug has: rate 32000 I tried rate 8000, 32000 and rate 8000 32000 but non of these worked. Cheers Ogi

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[alsa-devel] [PATCH] ALSA: ASoC: Fix SmartQ compilation
by Maurus Cuelenaere 05 Jan '11

05 Jan '11

This makes the SmartQ sound driver compile again with recent ASoC changes. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/s3c24xx/smartq_wm8987.c | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c index dd20ca7..aef94c7 100644 --- a/sound/soc/s3c24xx/smartq_wm8987.c +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -39,8 +39,8 @@ static int smartq_hifi_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 snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct s3c_i2sv2_rate_calc div; unsigned int clk = 0; int ret; @@ -156,8 +156,9 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LINPUT2", NULL, "Mic Bias"}, }; -static int smartq_wm8987_init(struct snd_soc_codec *codec) +static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_codec *codec = rtd->codec; int err = 0; /* Add SmartQ specific widgets */ @@ -190,8 +191,8 @@ static int smartq_wm8987_init(struct snd_soc_codec *codec) return err; /* Headphone jack detection */ - err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack); + err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &smartq_jack); if (err) return err; -- 1.7.1

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[alsa-devel] [PATCH] ASoC: omap: rx51: Add stereo output support to audio jack
by Jarkko Nikula 05 Jan '11

05 Jan '11

Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier. This patch adds support for it and stereo output can be active when "Jack Function" == "TV-OUT" || "Headphone". As the TPA6130 can output very high volume levels the output is limited with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This support could have been implemented earlier but I was targeting to have ASoC cross-device support first. But as there's no DAPM widgets anymore in TPA6130, I don't see any other way to implement this than by registering TPA6130 controls to aic34a with tpa6130a2_add_controls and by using tpa6130a2_stereo_enable. This is from top of "ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers" but applies also into lrg's tree. --- sound/soc/omap/Kconfig | 1 + sound/soc/omap/rx51.c | 43 +++++++++++++++++++++++++++++++++++++++---- 2 files changed, 40 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a088db6..b592298 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,6 +24,7 @@ config SND_OMAP_SOC_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 09fb0df..251afbe 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -31,6 +31,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <plat/mcbsp.h> +#include "../codecs/tpa6130a2.h" #include <asm/mach-types.h> @@ -47,7 +48,8 @@ enum { RX51_JACK_DISABLED, - RX51_JACK_TVOUT, /* tv-out */ + RX51_JACK_TVOUT, /* tv-out with stereo output */ + RX51_JACK_HP, /* headphone: stereo output, no mic */ }; static int rx51_spk_func; @@ -57,6 +59,15 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; + int hp = 0, tvout = 0; + + switch (rx51_jack_func) { + case RX51_JACK_TVOUT: + tvout = 1; + case RX51_JACK_HP: + hp = 1; + break; + } if (rx51_spk_func) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -66,9 +77,12 @@ static void rx51_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(dapm, "DMic"); else snd_soc_dapm_disable_pin(dapm, "DMic"); + if (hp) + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - gpio_set_value(RX51_TVOUT_SEL_GPIO, - rx51_jack_func == RX51_JACK_TVOUT); + gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); snd_soc_dapm_sync(dapm); } @@ -153,6 +167,19 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, return 0; } +static int rx51_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->dapm->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + tpa6130a2_stereo_enable(codec, 1); + else + tpa6130a2_stereo_enable(codec, 0); + + return 0; +} + static int rx51_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -212,11 +239,14 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), + SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), }; static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Headphone Jack", NULL, "LLOUT"}, + {"Headphone Jack", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -224,7 +254,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; -static const char *jack_function[] = {"Off", "TV-OUT"}; +static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -265,6 +295,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) /* Set up RX-51 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + err = tpa6130a2_add_controls(codec); + if (err < 0) + return err; + snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- 1.7.2.3

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[alsa-devel] [PATCH 0/5] Adding OMAP DMIC driver to kernel
by David Lambert 05 Jan '11

05 Jan '11

This is a patch series to add the OMAP Digital Microphone driver for OMAP4. It includes the driver, a generic DMIC codec, platform devices, as well as HWMOD entries for OMAP44xx chipsets. David Lambert (5): ASoC: DMIC: Adding the OMAP DMIC driver ASoC: DMIC codec: Adding a generic DMIC codec ASoC: DMIC: Adding OMAP DMIC driver to build OMAP4: hwmod: add entries for DMIC driver MAP4: DMIC: Add DMIC codec platform devices arch/arm/mach-omap2/board-4430sdp.c | 11 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 91 +++++ arch/arm/plat-omap/devices.c | 35 ++ arch/arm/plat-omap/include/plat/dmic.h | 83 ++++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dmic.c | 79 ++++ sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 596 ++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 38 ++ 11 files changed, 945 insertions(+), 0 deletions(-) create mode 100644 arch/arm/plat-omap/include/plat/dmic.h create mode 100644 sound/soc/codecs/dmic.c create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h

3 11

[alsa-devel] [PATCH] sound: usb-audio: Digidesign Mbox 2 Duplex mode support Fixed the tab indentation. Removed unnecessary comments. Put IDs in the right order. Removed support for spdif using a device_setup parameter, now only analogue duplex mode is supported.
by damien.zammit＠gmail.com 05 Jan '11

05 Jan '11

From: Damien Zammit <damien.zammit(a)gmail.com> Clemens wrote last year: >> +mbox2_reboot: >> + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), >> + 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); >> + >> + if (bootresponse == MBOX2_BOOT_LOADING) { >> + snd_printdd("device not ready, resending boot sequence...\n"); >> + goto mbox2_reboot; >> + } >This could be made a loop. And it could run infinitely long; maybe you >should add a timeout. (How long does it usually need?) I need help to write a loop with a timeout that is RT safe? The device responds before 5 seconds from zero power to bootup. --- sound/usb/midi.c | 12 ++++ sound/usb/quirks-table.h | 52 +++++++++++++++++++ sound/usb/quirks.c | 125 ++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 ++ sound/usb/usbaudio.h | 1 + 5 files changed, 195 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..2ea3636 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2477,6 +2477,58 @@ 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..917b991 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -287,6 +287,7 @@ 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 +480,9 @@ 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 +510,10 @@ 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 +535,9 @@ 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 +605,117 @@ 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; + u8 enablemagic[3]; + u8 bootresponse; + u8 temp[12]; + int fwsize; + + if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { + snd_printdd("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() + */ + +mbox2_reboot: + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); + + if (bootresponse == MBOX2_BOOT_LOADING) { + snd_printdd("device not ready, resending boot sequence...\n"); + goto mbox2_reboot; + } + + if (bootresponse == MBOX2_BOOT_READY) { + snd_printdd("device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); + err = usb_reset_configuration(dev); + if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); + snd_printdd("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_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); + return -ENODEV; + } + snd_printk("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[12]; + + /* 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..13fd3ad 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

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[alsa-devel] [PATCH] ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers
by Jarkko Nikula 05 Jan '11

05 Jan '11

After multi-component conversion these machine drivers don't actually need anything from sound/soc/codecs/tlv320aic3x.h so don't include it. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- rx51.c and n810.c compile tested but these other machine drivers seems to not use anything from tlv320aic3x.h either so should be safe to remove. --- sound/soc/davinci/davinci-evm.c | 1 - sound/soc/omap/n810.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/s6000/s6105-ipcam.c | 1 - sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 -- 5 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 24b031e..0c2d6ba 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -26,7 +26,6 @@ #include <mach/edma.h> #include <mach/mux.h> -#include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fe215f3..83d213b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -35,7 +35,6 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" -#include "../codecs/tlv320aic3x.h" #define N810_HEADSET_AMP_GPIO 10 #define N810_SPEAKER_AMP_GPIO 101 diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 150042a..09fb0df 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -36,7 +36,6 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" -#include "../codecs/tlv320aic3x.h" #define RX51_TVOUT_SEL_GPIO 40 #define RX51_JACK_DETECT_GPIO 177 diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index bf3f3f9..5890e43 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -21,7 +21,6 @@ #include <variant/dmac.h> -#include "../codecs/tlv320aic3x.h" #include "s6000-pcm.h" #include "s6000-i2s.h" diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 0083aff..bb4292e 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -21,8 +21,6 @@ #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" -#include "../codecs/tlv320aic3x.h" - static const struct snd_soc_dapm_widget dapm_widgets[] = { SND_SOC_DAPM_LINE("GSM Out", NULL), SND_SOC_DAPM_LINE("GSM In", NULL), -- 1.7.2.3

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[alsa-devel] [PATCH] Fix the device references to codec and platform drivers
by Harsha, Priya 05 Jan '11

05 Jan '11

From: Harsha Priya <priya.harsha(a)intel.com> The soc-core takes the platform and codec driver reference during probe. Few of these references are not released during remove. This cause the platform and codec driver module unload to fail. This patch fixes by the taking only one reference to platform and codec module during probe and releases them correctly during remove. This allows load/unload properly Signed-off-by: Vinod Koul<vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/soc-core.c | 14 +++++++------- 1 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a233607..bac7291 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1259,9 +1259,6 @@ find_codec: if (!strcmp(codec->name, dai_link->codec_name)) { rtd->codec = codec; - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - /* CODEC found, so find CODEC DAI from registered DAIs from this CODEC*/ list_for_each_entry(codec_dai, &dai_list, list) { if (codec->dev == codec_dai->dev && @@ -1287,10 +1284,6 @@ find_platform: /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(platform, &platform_list, list) { if (!strcmp(platform->name, dai_link->platform_name)) { - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - rtd->platform = platform; goto out; } @@ -1425,6 +1418,9 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ + if (!try_module_get(codec->dev->driver->owner)) + return -ENODEV; + codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); list_add(&codec->dapm.list, &card->dapm_list); @@ -1556,6 +1552,10 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } /* mark platform as probed and add to card platform list */ + + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + platform->probed = 1; list_add(&platform->card_list, &card->platform_dev_list); } -- 1.7.2.3

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[alsa-devel] [PATCH] Alsa-info: Fix typo
by David Henningsson 05 Jan '11

05 Jan '11

This simple typo has annoyed me for a while. Please fix it, so I can focus on more important issues :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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