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

[alsa-devel] Capture Source/Input Souce of Alsa sound cards and pulseaudio source
by Raymond Yau 31 Mar '11

31 Mar '11

There are 4 Capture Volume/Switch (Front, Surround, Synth and Wave) and Capture source of ac97 codec contain 8 input sources (Mic, CD, Video , Aux, Line, Stereo Mix, Mono Mix and Phone) on SB Live! Platinum How can ALSA application know whether all 12 sources can be selected for device 0 ? card 1: Live [SB Live! Platinum [CT4760P]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Live [SB Live! Platinum [CT4760P]], device 1: emu10k1 mic [Mic Capture] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Live [SB Live! Platinum [CT4760P]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 1/1 Subdevice #0: subdevice #0

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[PATCH] ALSA: HDA: Add dock mic quirk for Lenovo Thinkpad X220
by David Henningsson 31 Mar '11

31 Mar '11

This quirk is needed for the docking station mic of Lenovo Thinkpad X220 to function correctly. BugLink: http://bugs.launchpad.net/bugs/746259 Cc: stable(a)kernel.org Tested-by: James Ferguson <james.ferguson(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d08cf31..69e3386 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3034,6 +3034,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} -- 1.7.1 --------------090408050708050906070307--

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Re: [alsa-devel] 2.6.38+ hard lockup in pcm_lib.c
by Clemens Ladisch 31 Mar '11

31 Mar '11

Kelly Anderson wrote: > I had a consistent hard-lockup on one of my machines, an Amd X6, just > after Kdm loaded the desktop (most certainly when it decides to start > talking to the sound card). Everything was dead, even ssh access to the > computer. I git bisected it until I found the commit that was causing > it. After reverting the patch everything is fine. > > commit 59ff878ffb26bc0be812ca8295799164f413ae88 > Author: Clemens Ladisch > Date: Thu Nov 18 09:43:52 2010 +0100 > > ALSA: pcm: detect xruns in no-period-wakeup mode This looks as if you get an infinite loop, probably because hw_ptr_buffer_jiffies is zero. Please try this patch: --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -425,6 +425,8 @@ static int snd_pcm_hw_params(struct snd_ runtime->no_period_wakeup = (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + runtime->hw_ptr_buffer_jiffies = + max(1uL, (runtime->buffer_size * HZ) / runtime->rate); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -866,8 +868,6 @@ static void snd_pcm_post_start(struct sn struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / - runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0)

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[alsa-devel] [Fwd: Re: Problems with snd_hda_intel in Linux kernel 2.6.38]
by Svante Signell 31 Mar '11

31 Mar '11

I am subscribed. What's the problem? (now without attachment). How can I send the attachment to the developers? > Your mail to 'Alsa-devel' with the subject > > Re: [alsa-devel] Problems with snd_hda_intel in Linux kernel > 2.6.38 > > Is being held until the list moderator can review it for approval. > > The reason it is being held: > > Message body is too big: 488063 bytes with a limit of 60 KB > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. If you would like to cancel > this posting, please visit the following URL: > > > http://mailman.alsa-project.org/mailman/confirm/alsa-devel/9dc4fe44b477fdc3… > -------- Forwarded Message -------- From: Svante Signell <srs(a)kth.se> To: Clemens Ladisch <clemens(a)ladisch.de> Cc: Takashi Iwai <tiwai(a)suse.de>, 619034(a)bugs.debian.org, alsa-devel(a)alsa-project.org, 613979(a)bugs.debian.org Subject: Re: [alsa-devel] Problems with snd_hda_intel in Linux kernel 2.6.38 Date: Thu, 31 Mar 2011 00:42:46 +0200 On Wed, 2011-03-30 at 15:13 +0200, Clemens Ladisch wrote: > Svante Signell wrote: > > Code: f4 01 00 00 ef 31 f6 48 89 df e8 15 dd ff ff 85 c0 > > 0f 88 2b 03 00 00 48 89 ef e8 ee 11 b9 e0 8b 7b 40 e8 9f 25 a7 e0 48 8b > > 43 38 <66> 8b 10 66 89 14 24 8b 43 14 83 e8 03 83 f8 01 77 32 31 d2 be > > 5: 31 f6 xor %esi,%esi > 7: 48 89 df mov %rbx,%rdi > a: e8 15 dd ff ff callq 0xffffffffffffdd24 > f: 85 c0 test %eax,%eax > 11: 0f 88 2b 03 00 00 js 0x342 > 17: 48 89 ef mov %rbp,%rdi > 1a: e8 ee 11 b9 e0 callq 0xffffffffe0b9120d > 1f: 8b 7b 40 mov 0x40(%rbx),%edi > 22: e8 9f 25 a7 e0 callq 0xffffffffe0a725c6 > 27: 48 8b 43 38 mov 0x38(%rbx),%rax > 2b: 66 8b 10 mov (%rax),%dx <-- crash here > 2e: 66 89 14 24 mov %dx,(%rsp) > 32: 8b 43 14 mov 0x14(%rbx),%eax > 35: 83 e8 03 sub $0x3,%eax > 38: 83 f8 01 cmp $0x1,%eax > 3b: 77 32 ja 0x6f > 3d: 31 d2 xor %edx,%edx > > This is the azx_readw(chip, GCAP) in azx_create(); chip->remap_addr is > 0xffffc90011c08000 which does look like a valid pointer, but isn't. Thank you Clemens! Maybe your input is sufficient to solve this problem. I have now installed the debug version of the kernel, the objdump output is attached (please let me know if you are missing something).sorry, I don't know where to find the relevant information in this file, but that is all I have (still very large). (Does not include the error messages on stderr, maybe something is still missing.) Thanks!

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[alsa-devel] [PATCH 1/2] ASoC: soc-cache: Fix indentation issues
by Dimitris Papastamos 30 Mar '11

30 Mar '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1210a6f..22c0b95 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -93,13 +93,13 @@ static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) } static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[2]; @@ -111,7 +111,7 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_4_12_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[2]; @@ -209,7 +209,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, #if defined(CONFIG_SPI_MASTER) static int snd_soc_8_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -257,7 +257,7 @@ static unsigned int do_i2c_read(struct snd_soc_codec *codec, #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) + unsigned int r) { u8 reg = r; u8 data; @@ -307,13 +307,13 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #endif static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[3]; @@ -327,7 +327,7 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_8_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -379,7 +379,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[4]; -- 1.7.4.1

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[alsa-devel] [PATCH 01/11] ASoC: Use data based init for ak4535 DAPM
by Lu Guanqun 30 Mar '11

30 Mar '11

CC: Liam Girdwood <lrg(a)slimlogic.co.uk> CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: Jarkko Nikula <jhnikula(a)gmail.com> CC: Timur Tabi <timur(a)freescale.com> CC: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/soc/codecs/ak4535.c | 19 +++++-------------- 1 files changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 8b38739..e1a214e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -230,7 +230,7 @@ static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AIN"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route ak4535_audio_map[] = { /*stereo mixer */ {"Stereo Mixer", "Playback Switch", "DAC"}, {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, @@ -287,17 +287,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Input Mixer", "Aux Capture Switch", "Aux In"}, }; -static int ak4535_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, ak4535_dapm_widgets, - ARRAY_SIZE(ak4535_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -457,8 +446,6 @@ static int ak4535_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); - ak4535_add_widgets(codec); - return 0; } @@ -480,6 +467,10 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4535_reg, + .dapm_widgets = ak4535_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), + .dapm_routes = ak4535_audio_map, + .num_dapm_routes = ARRAY_SIZE(ak4535_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.2.3

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Re: [alsa-devel] [PATCH 1/2] ASoC: soc-cache: Fix indentation issues
by Liam Girdwood 30 Mar '11

30 Mar '11

On 28/03/11 11:39, dp(a)opensource.wolfsonmicro.com wrote: > From: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> > > Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> All Acked-by: Liam Girdwood <lrg(a)ti.com>

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[alsa-devel] ASoC update for 2.6.39
by Mark Brown 30 Mar '11

30 Mar '11

The following changes since commit d454f39f3ff3ee5a818c271026081a48a537bd41: Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-03-28 13:02:29 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Mark Brown (1): ASoC: Fix CODEC device name for Corgi sound/soc/pxa/corgi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] sound/soc/pxa/zylonite.c: set .codec_dai_name in initializer
by Antonio Ospite 30 Mar '11

30 Mar '11

Fix the initialization of .codec_dai_name in zylonite_dai initializer, do not mix it with the initialization of .codec_name which is set already a few lines above. Signed-off-by: Antonio Ospite <ospite(a)studenti.unina.it> --- Hi, I didn't test that because I don't have the hardware, I just noticed it by reading the code from 2.6.38 and thought this may be the right fix. Is there any mechanism to spot out when the same field in a struct initializer is set more than once? Regards, Antonio Ospite http://ao2.it sound/soc/pxa/zylonite.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 25bba10..5e96f7e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", - .codec_name = "wm9713-hifi", + .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, }, { @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_name = "wm9713-aux", + .codec_dai_name = "wm9713-aux", }, { .name = "WM9713 Voice", @@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", - .codec_name = "wm9713-voice", + .codec_dai_name = "wm9713-voice", .ops = &zylonite_voice_ops, }, }; -- 1.7.4.1

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[alsa-devel] [PATCH 1/2] ASoC: SSM2602: drop rate and sample bits constraints
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 53 -------------------------------------------- 1 files changed, 0 insertions(+), 53 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2727bef..70a33414 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -49,8 +49,6 @@ struct ssm2602_priv { unsigned int sysclk; enum snd_soc_control_type control_type; void *control_data; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; }; /* @@ -278,15 +276,9 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); - if (substream == ssm2602->slave_substream) { - dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -316,44 +308,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; - - return 0; -} - static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -370,16 +324,10 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) @@ -505,7 +453,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops ssm2602_dai_ops = { - .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, -- 1.7.4.1

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