March 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] wm8994: playback => capture
by Dan Carpenter 24 Mar '10

24 Mar '10

Sparse caught that initialize "playback" two times instead of initializing "capture". Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771..d10d651 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3401,7 +3401,7 @@ struct snd_soc_dai wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, - .playback = { + .capture = { .stream_name = "AIF3 Capture", .channels_min = 2, .channels_max = 2,

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Re: [alsa-devel] cmipci and sound skipping ?
by Clemens Ladisch 24 Mar '10

24 Mar '10

Matija Nalis wrote: > On Tue, Mar 23, 2010 at 09:36:01AM +0100, Clemens Ladisch wrote: > > Are you okay with your email address (which one?) to be published with > > the patch: > > Reported-and-tested-by: Matija Nalis <...> > > Sure, mnalis-alsadev @ voyager.hr would be ok for e-mail. Applied. Thanks! Clemens

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[alsa-devel] [PATCH] ALSA: aureon - Patch for suspend/resume for Terratec Aureon cards.
by Bernhard Urban 23 Mar '10

23 Mar '10

Add proper suspend/resume code for Terratec Aureon cards. Based on ice1724 suspend/resume work of Igor Chernyshev. Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4944 Tested on linux-2.6.32.9 Signed-off-by: Bernhard Urban <lewurm(a)gmail.com> --- sound/pci/ice1712/aureon.c | 90 +++++++++++++++++++++++++++++++------------ 1 files changed, 65 insertions(+), 25 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 110d16e..1772e1a 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1971,11 +1971,10 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) return 0; } - /* - * initialize the chip + * reset the chip */ -static int __devinit aureon_init(struct snd_ice1712 *ice) +static int aureon_reset(struct snd_ice1712 *ice) { static const unsigned short wm_inits_aureon[] = { /* These come first to reduce init pop noise */ @@ -2062,30 +2061,10 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) 0x0605, /* slave, 24bit, MSB on second OSCLK, SDOUT for right channel when OLRCK is high */ (unsigned short)-1 }; - struct aureon_spec *spec; unsigned int tmp; const unsigned short *p; - int err, i; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - ice->spec = spec; - - if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { - ice->num_total_dacs = 6; - ice->num_total_adcs = 2; - } else { - /* aureon 7.1 and prodigy 7.1 */ - ice->num_total_dacs = 8; - ice->num_total_adcs = 2; - } - - /* to remeber the register values of CS8415 */ - ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (!ice->akm) - return -ENOMEM; - ice->akm_codecs = 1; + int err; + struct aureon_spec *spec = ice->spec; err = aureon_ac97_init(ice); if (err != 0) @@ -2133,6 +2112,62 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* initialize PCA9554 pin directions & set default input */ aureon_pca9554_write(ice, PCA9554_DIR, 0x00); aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */ + return 0; +} + +/* + * suspend/resume + */ +#ifdef CONFIG_PM +static int aureon_resume(struct snd_ice1712 *ice) +{ + struct aureon_spec *spec = ice->spec; + int err, i; + + err = aureon_reset(ice); + if (err != 0) + return err; + + /* workaround for poking volume with alsamixer after resume: + * just set stored volume again */ + for (i = 0; i < ice->num_total_dacs; i++) { + wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); + } + return 0; +} +#endif + +/* + * initialize the chip + */ +static int __devinit aureon_init(struct snd_ice1712 *ice) +{ + struct aureon_spec *spec; + int i, err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { + ice->num_total_dacs = 6; + ice->num_total_adcs = 2; + } else { + /* aureon 7.1 and prodigy 7.1 */ + ice->num_total_dacs = 8; + ice->num_total_adcs = 2; + } + + /* to remeber the register values of CS8415 */ + ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + if (!ice->akm) + return -ENOMEM; + ice->akm_codecs = 1; + + err = aureon_reset(ice); + if (err != 0) + return err; spec->master[0] = WM_VOL_MUTE; spec->master[1] = WM_VOL_MUTE; @@ -2141,6 +2176,11 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); } +#ifdef CONFIG_PM + ice->pm_resume = aureon_resume; + ice->pm_suspend_enabled = 1; +#endif + return 0; } -- 1.7.0.1

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 23 Mar '10

23 Mar '10

The following changes since commit 74511020dde10252f8b8e648690d99dba721de14: Mark Brown (1): Merge branch 'for-2.6.34' into for-2.6.35 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Kuninori Morimoto (3): ASoC: ak4642: Add pll select support ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops ASoC: ak4642: Add enhanced sampling rate Mark Brown (1): ASoC: Implement WM8994 DAI tristate support sound/soc/codecs/ak4642.c | 175 ++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8994.c | 38 ++++++++++ sound/soc/sh/fsi-ak4642.c | 14 ++++ 3 files changed, 192 insertions(+), 35 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: ak4642: sampling rate update
by Kuninori Morimoto 23 Mar '10

23 Mar '10

Dear Mark These patches are sampling rate update for ak4642 Kuninori Morimoto (3): ASoC: ak4642: Add pll select support ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops ASoC: ak4642: Add enhanced sampling rate It are tested on ms7724se/AP4 board. Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: TWL4030: PM fix for output amplifiers
by Peter Ujfalusi 23 Mar '10

23 Mar '10

Gain controls on outputs affect the power consumption when the gain is set to non 0 value. To prevent amps beeing enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 72 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 56 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af..bf59b8a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -135,9 +135,11 @@ struct twl4030_priv { unsigned int sysclk; - /* Headset output state handling */ - unsigned int hsl_enabled; - unsigned int hsr_enabled; + /* Output (with associated amp) states */ + u8 hsl_enabled, hsr_enabled; + u8 earpiece_enabled; + u8 predrivel_enabled, predriver_enabled; + u8 carkitl_enabled, carkitr_enabled; }; /* @@ -173,12 +175,47 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + struct twl4030_priv *twl4030 = codec->private_data; + int write_to_reg = 0; + twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, - reg); - else - return 0; + if (likely(reg < TWL4030_REG_SW_SHADOW)) { + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; + } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + } + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) @@ -525,26 +562,26 @@ static int micpath_event(struct snd_soc_dapm_widget *w, * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the * amplifier associated with the output pin) - * On mute bypass the reg_cache and mute the volume - * On unmute: restore the register content + * On mute bypass the reg_cache and write 0 to the register + * On unmute: restore the register content from the reg_cache * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R */ #define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ struct snd_kcontrol *kcontrol, int event) \ { \ - u8 reg_val; \ + struct twl4030_priv *twl4030 = w->codec->private_data; \ \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ + twl4030->pin_name##_enabled = 1; \ twl4030_write(w->codec, reg, \ twl4030_read_reg_cache(w->codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ - reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ - reg_val & (~mask), \ - reg); \ + twl4030->pin_name##_enabled = 0; \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + 0, reg); \ break; \ } \ return 0; \ @@ -664,7 +701,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + /* Actually write to the register */ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain, + TWL4030_REG_HS_GAIN_SET); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Wait ramp delay time + 1, so the VMID can settle */ -- 1.7.0.2

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[alsa-devel] cmipci and sound skipping ?
by Matija Nalis 23 Mar '10

23 Mar '10

My card is "Asonic C-Media8738, 4 channel" and mostly works with snd_cmipci driver - it is just that (even on completely unloaded system) the sound is (very annoyingly) skipping. The speaker-test application throws out "Write error: -32,Broken pipe" errors on every skip (usually every 5-30 times a wav is played). mpg321 is much worse (every few seconds it clips the sample) making the listening to music unedurable. I'm using Debian Lenny with kernel 2.6.26 (also tried 2.6.32, no change) I've upgraded to the latest snapshot of driver, lib and utils as recommended, the bug is still there. When driver is compiled with debug, on every skip it throws out lines in dmesg with info as: ALSA pcm_lib.c:316: BUG: pcmC0D0p:0, pos = 4294918144, buffer size = 16384, period size = 1024 Note that the pos is *always* the same (4294918144 = 0xffff4000 = -16384) which might be important. I've reported the problem in bug #4899 at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4899 (alsa-info.sh and more is attached there) There I got pointer to contact the alsa-devel list with XRUN_Debug info. So here it is. When I enable it with "echo 101 > /proc/asound/card0/pcm0p/xrun_debug" I get the following output : [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816904, pos=1037/1024/16384, hwptr=230413/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2052/1024/16384, hwptr=230413/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231428/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2061/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2061/1024/16384, hwptr=231437/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816914, pos=3076/1024/16384, hwptr=231437/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816914, pos=3076/1024/16384, hwptr=232452/229376 [33802.573395] ALSA pcm_lib.c:316: BUG: pcmC0D0p:0, pos = 4294918144, buffer size = 16384, period size = 1024 [33802.573395] ALSA pcm_lib.c:415: PCM: hw_ptr skipping! (pos=0, delta=13308, period=1024, jdelta=0/69/0, hw_ptr=232452/232452) [33806.867658] ALSA pcm_lib.c:316: BUG: pcmC0D0p:0, pos = 4294918144, buffer size = 16384, period size = 1024 [33806.867666] ALSA pcm_lib.c:415: PCM: hw_ptr skipping! (pos=0, delta=16380, period=1024, jdelta=0/85/0, hw_ptr=425988/425988) Also it seems that enabling xrun_debug kludges around the bug - the annoying skipping (and "Write error: -32,Broken pipe" errors in speaker-test) disappears or at least becomes almost inaudiable. Would some other xrun_debug setting be more useful to give better insight ? Anything else I can provide or test to help debug the problem ? Please CC: me as I'm not subscribed to the list. -- Opinions above are GNU-copylefted.

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 23 Mar '10

23 Mar '10

The following changes since commit fb91e0beb95e7e253a22fc87b39b5543073eae33: Stephen Rothwell (1): Add linux-next specific files for 20100322 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Pull request against -next due to the merge up of v2.6.34-rc2 (log only, diffstat got confused by the use of -next and tried to include the entire contents of the branch not just the additions). Daniel Mack (1): ASoC: pxa-pcm-lib: initialize DMA channel to -1 Mark Brown (8): Merge commit 'v2.6.34-rc2' into for-2.6.34 ASoC: Remove BROKEN from i.MX audio after dependencies merged ASoC: Add a notifier for jack status changes ASoC: Allow force enabled pins to be disabled ASoC: Move WM8350 microphone detection bias managment out of driver ASoC: Allow disabling of WM835x jack detection ASoC: Allow WM8903 mic detect disable and don't force bias on Merge branch 'for-2.6.34' into for-2.6.35 Peter Ujfalusi (1): ASoC: TWL4030: PM fix for output amplifiers

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[alsa-devel] [PATCH v2] ASoC: TWL4030: PM fix for output amplifiers
by Peter Ujfalusi 22 Mar '10

22 Mar '10

Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/twl4030.c | 72 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 56 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af..bf59b8a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -135,9 +135,11 @@ struct twl4030_priv { unsigned int sysclk; - /* Headset output state handling */ - unsigned int hsl_enabled; - unsigned int hsr_enabled; + /* Output (with associated amp) states */ + u8 hsl_enabled, hsr_enabled; + u8 earpiece_enabled; + u8 predrivel_enabled, predriver_enabled; + u8 carkitl_enabled, carkitr_enabled; }; /* @@ -173,12 +175,47 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + struct twl4030_priv *twl4030 = codec->private_data; + int write_to_reg = 0; + twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, - reg); - else - return 0; + if (likely(reg < TWL4030_REG_SW_SHADOW)) { + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; + } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + } + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) @@ -525,26 +562,26 @@ static int micpath_event(struct snd_soc_dapm_widget *w, * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the * amplifier associated with the output pin) - * On mute bypass the reg_cache and mute the volume - * On unmute: restore the register content + * On mute bypass the reg_cache and write 0 to the register + * On unmute: restore the register content from the reg_cache * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R */ #define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ struct snd_kcontrol *kcontrol, int event) \ { \ - u8 reg_val; \ + struct twl4030_priv *twl4030 = w->codec->private_data; \ \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ + twl4030->pin_name##_enabled = 1; \ twl4030_write(w->codec, reg, \ twl4030_read_reg_cache(w->codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ - reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ - reg_val & (~mask), \ - reg); \ + twl4030->pin_name##_enabled = 0; \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + 0, reg); \ break; \ } \ return 0; \ @@ -664,7 +701,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + /* Actually write to the register */ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain, + TWL4030_REG_HS_GAIN_SET); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Wait ramp delay time + 1, so the VMID can settle */ -- 1.7.0.2

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 22 Mar '10

22 Mar '10

The following changes since commit 220bf991b0366cc50a94feede3d7341fa5710ee4: Linus Torvalds (1): Linux 2.6.34-rc2 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Plus a merge up of 2.6.34-rc2 which means that the ARM build issues are pretty much gone. Only the two new patches for ASoC are listed below. Daniel Mack (1): ASoC: pxa-pcm-lib: initialize DMA channel to -1 Mark Brown (1): ASoC: Remove BROKEN from i.MX audio after dependencies merged sound/arm/pxa2xx-pcm-lib.c | 1 + sound/soc/imx/Kconfig | 2 +-

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