January 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Raymond Yau 11 Jan '15

11 Jan '15

> > > >If Combo audio connector does not support conventional microphone, this mean > >the driver should not create mic jack control for node 0x1a > > > >http://support.lenovo.com/us/en/documents/pd029567 > > > >Connect headphones or a headset that has a 3.5-mm (0.14-inch), 4-pole plug > >to the combo audio connector to listen to the sound from the computer. > > I do not have any 4-pole (3-rings) equipment to test. > A conventional 3-pole (2-ring) microphone certainly does not do its job. > > Then again, I ditched unpowered microphones (most external ones are). > The internal mic next to the internal webcam is powered and provides > better gain. > > The Mic control in alsamixer should stay. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… The debug version of driver message indicate whether the codec can differentiate headphone , headset (ctia or omtp) or not http://voices.canonical.com/david.henningsson/2014/03/07/headset-jacks-on-n… http://bazaar.launchpad.net/~unity-settings-daemon-team/unity-settings-daem… Headphone Mic Jack - indicates headphone and mic-in mode share the same jack, i e, not two separate jacks. Hardware cannot distinguish between a headphone and a mic. Headset Mic Phantom Jack - indicates headset jack where hardware can not distinguish between headphones and headsets Headset Mic Jack - indicates headset jack where hardware can distinguish between headphones and headsets. There is no use popping up a dialog in this case, unless we already need to do this for the mic-in mode. control.14 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } What is the usage of this jack detect control if conventional mic is not supported ?

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Re: [alsa-devel] ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine
by Dan Carpenter 10 Jan '15

10 Jan '15

Hello David Henningsson, The patch 33f4acd3b214: "ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine" from Jan 7, 2015, leads to the following static checker warning: sound/pci/hda/patch_realtek.c:3551 alc280_fixup_hp_gpio2_mic_hotkey() error: don't call input_free_device() after input_unregister_device() sound/pci/hda/patch_realtek.c 3545 switch (action) { 3546 case HDA_FIXUP_ACT_PROBE: 3547 spec->init_amp = ALC_INIT_DEFAULT; 3548 break; 3549 case HDA_FIXUP_ACT_FREE: 3550 input_unregister_device(spec->kb_dev); 3551 input_free_device(spec->kb_dev); I think this is a double free. 3552 spec->kb_dev = NULL; 3553 } 3554 #endif 3555 } regards, dan carpenter

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[alsa-devel] [PATCH 1/8] ASoC: corgi: Automatically disconnect non-connected pins
by Lars-Peter Clausen 09 Jan '15

09 Jan '15

All DAPM input and output pins of the wm8994 are either used in the card's DAPM routing table or are marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b7cd0a7..3580d10 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -259,20 +259,6 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { corgi_set_spk), }; -/* - * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device - */ -static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_nc_pin(dapm, "LLINEIN"); - snd_soc_dapm_nc_pin(dapm, "RLINEIN"); - - return 0; -} - /* corgi digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", @@ -281,7 +267,6 @@ static struct snd_soc_dai_link corgi_dai = { .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", - .init = corgi_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &corgi_ops, @@ -300,6 +285,7 @@ static struct snd_soc_card corgi = { .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), .dapm_routes = corgi_audio_map, .num_dapm_routes = ARRAY_SIZE(corgi_audio_map), + .fully_routed = true, }; static int corgi_probe(struct platform_device *pdev) -- 1.8.0

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[alsa-devel] [PATCH] ASoC: ttc-dkb: Remove unnecessary snd_soc_dapm_disable_pin() calls
by Lars-Peter Clausen 09 Jan '15

09 Jan '15

The "Headset Mic 2" and the "Headset Stereophone" widget are managed by the jack detection logic. Their will be set depending on whether something is connected to the jack or not. There is no need to manually change the state beforehand as it will be overwritten anyway. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/ttc-dkb.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index e3d7257a..5001dbb 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -76,10 +76,6 @@ static const struct snd_soc_dapm_route ttc_audio_map[] = { static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE -- 1.8.0

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[alsa-devel] [PATCH] ASoC: pcm512x: Fix DSP program selection
by Peter Rosin 09 Jan '15

09 Jan '15

From: Peter Rosin <peda(a)axentia.se> The DSP programs are listed out of order. Signed-off-by: Peter Rosin <peda(a)axentia.se> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 0c8aefab404c..640c99198cda 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -188,8 +188,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", - "Fixed process flow", "High attenuation with de-emphasis", + "Fixed process flow", "Ringing-less low latency FIR", }; -- 1.7.10.4

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[alsa-devel] [PATCH 1/3] ASoC: rt5677: Modify the behavior that updates the PLL parameter.
by Oder Chiou 09 Jan '15

09 Jan '15

The patch modified the behavior that updates the PLL parameter. It set the update bit before the PLL power up. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8900221..d27630a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2174,10 +2174,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2192,10 +2196,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2303,9 +2311,11 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, - 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT, - 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5677_ASRC_1, 0, 0, NULL, 0), -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: Intel: Split hsw_pcm_data for playback and capture
by Jie Yang 09 Jan '15

09 Jan '15

There may be 2 pcm streams for a same DAI at most, and these 2 streams should have different hsw_pcm_data, e.g. they have different persistent data, so here we need split hsw_pcm_data for playback and capture, to make sure they won't be mixed and keep cleaned. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 91 ++++++++++++++++++++++++++------------- 1 file changed, 61 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5448f79..ad7f4a5 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -135,7 +135,17 @@ struct hsw_priv_data { struct snd_dma_buffer dmab[HSW_PCM_COUNT][2]; /* DAI data */ - struct hsw_pcm_data pcm[HSW_PCM_COUNT]; + struct hsw_pcm_data pcm[HSW_PCM_COUNT][2]; +}; + + +/* static mappings between PCMs and modules - may be dynamic in future */ +static struct hsw_pcm_module_map mod_map[] = { + {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, + {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, + {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, }; static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data); @@ -168,9 +178,14 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); - struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -212,9 +227,14 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); - struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -309,7 +329,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -353,7 +373,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; struct sst_module *module_data; struct sst_dsp *dsp; @@ -362,7 +382,10 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, enum sst_hsw_stream_path_id path_id; u32 rate, bits, map, pages, module_id; u8 channels; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; /* check if we are being called a subsequent time */ if (pcm_data->allocated) { @@ -552,8 +575,12 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -597,11 +624,16 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; snd_pcm_uframes_t offset; uint64_t ppos; - u32 position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); + u32 position; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; + position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); @@ -618,8 +650,10 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; - pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -648,9 +682,12 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); ret = sst_hsw_stream_reset(hsw, pcm_data->stream); @@ -685,15 +722,6 @@ static struct snd_pcm_ops hsw_pcm_ops = { .page = snd_pcm_sgbuf_ops_page, }; -/* static mappings between PCMs and modules - may be dynamic in future */ -static struct hsw_pcm_module_map mod_map[] = { - {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, - {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, - {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, -}; - static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) { struct sst_hsw *hsw = pdata->hsw; @@ -701,7 +729,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; /* create new runtime module, use same offset if recreated */ pcm_data->runtime = sst_hsw_runtime_module_create(hsw, @@ -716,7 +744,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) err: for (--i; i >= 0; i--) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } @@ -729,7 +757,7 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } @@ -757,7 +785,10 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) return ret; } } - priv_data->pcm[rtd->cpu_dai->id].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; return ret; } @@ -866,10 +897,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { - mutex_init(&priv_data->pcm[i].mutex); - /* playback */ if (hsw_dais[i].playback.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_PLAYBACK].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][0]); if (ret < 0) @@ -878,6 +908,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* capture */ if (hsw_dais[i].capture.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_CAPTURE].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][1]); if (ret < 0) @@ -1076,7 +1107,7 @@ static void hsw_pcm_complete(struct device *dev) } for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; @@ -1110,7 +1141,7 @@ static int hsw_pcm_prepare(struct device *dev) return 0; /* suspend all active streams */ for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; @@ -1129,7 +1160,7 @@ static int hsw_pcm_prepare(struct device *dev) /* preserve persistent memory */ for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; -- 1.9.1

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[alsa-devel] [PATCH 00/33] Simplify PM callbacks of PCI drivers
by Takashi Iwai 09 Jan '15

09 Jan '15

This is a series of patches to get rid of superfluous calls found in suspend/resume callbacks of PCI drivers. Just systematically to each driver code. Takashi

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[alsa-devel] [PATCH v2] ALSA: usb-audio: Add support for Akai MPC Element USB MIDI controller
by Paul Bonser 09 Jan '15

09 Jan '15

From: Paul Bonser <misterpib(a)gmail.com> The Akai MPC Element incorrectly reports its bInterfaceClass as 255, but otherwise implements the USB MIDI spec correctly. This adds a quirks-table.h entry which allows the device to be recognized as a standard USB MIDI device. Signed-off-by: Paul Bonser <misterpib(a)gmail.com> --- diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8590a32..e869b6c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2516,6 +2516,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* Akai MPC Element */ + USB_DEVICE(0x09e8, 0x0021), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012),

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[alsa-devel] unload Audio drivers while playback stream is active case kernel crash
by Wang, Jiada (ESD) 09 Jan '15

09 Jan '15

Hi Community Sorry previous email doesn't have formatted crash log. I am using i.MX6Q sabreSD board, which have imx_wm892 machine driver, wm8962 codec and SSI CPU DAI, I got Kernel crash when unloading audio drivers (playback stream is active) modprobe -r snd_soc_imx_wm8962 modprobe -r snd_soc_fsl_ssi modprobe -r snd_soc_wm8962 ######################### [ 31.363894] wm8962 0-001a: Failed to read DSPCLK: -1 [ 31.386320] wm8962 0-001a: Failed to read DCS status: -1 [ 31.417252] Unable to handle kernel NULL pointer dereference at virtual address 0000001c [ 31.425380] pgd = 80004000 [ 31.428154] [0000001c] *pgd=00000000 [ 31.431796] Internal error: Oops: 17 [#1] SMP ARM [ 31.436527] Modules linked in: snd_soc_imx_audmux snd_soc_wm8962 snd_soc_fsl_ssi imx_pcm_fiq evbug] [ 31.448764] CPU: 3 PID: 380 Comm: kworker/3:3 Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #8 [ 31.457210] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 31.463763] Workqueue: events_power_efficient close_delayed_work [ 31.469792] task: be20ce00 ti: bbe20000 task.ti: bbe20000 [ 31.475202] PC is at soc_dapm_dai_stream_event+0x14/0xac [ 31.480522] LR is at snd_soc_dapm_stream_event+0x64/0x90 [ 31.485842] pc : [<8052b298>] lr : [<8052e5cc>] psr: 60000013 [ 31.485842] sp : bbe21e38 ip : bbe21e50 fp : bbe21e4c [ 31.497324] r10: 00000001 r9 : 00000000 r8 : bd449ccc [ 31.502555] r7 : 00000000 r6 : 00000002 r5 : 00000002 r4 : 00000000 [ 31.509087] r3 : 00000000 r2 : 00000002 r1 : 00000000 r0 : 00000000 [ 31.515622] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 31.522936] Control: 10c5387d Table: 4d42004a DAC: 00000015 [ 31.528689] Process kworker/3:3 (pid: 380, stack limit = 0xbbe20238) [ 31.535048] Stack: (0xbbe21e38 to 0xbbe22000) [ 31.539412] 1e20: 00000000 ba945810 [ 31.547599] 1e40: bbe21e74 bbe21e50 8052e5cc 8052b290 00000000 ba945dd4 ba94581c ba945810 [ 31.555785] 1e60: bbe21eb0 be7c0c00 bbe21e94 bbe21e78 80530038 8052e574 8052ffec bd585680 [ 31.563971] 1e80: ba945dd4 be7bd380 bbe21eec bbe21e98 8003ef74 8052fff8 00000001 00000000 [ 31.572158] 1ea0: 8003ef08 8003f294 00000000 00000000 811d1670 80b33330 00000000 808b61c0 [ 31.580344] 1ec0: 806bac24 bd585680 be7bd3b0 809c7882 bd585698 00000008 be7bd380 be7bd380 [ 31.588529] 1ee0: bbe21f24 bbe21ef0 8003f388 8003edec bdad17c0 bd585680 8003f228 00000000 [ 31.596715] 1f00: bdad17c0 bd585680 8003f228 00000000 00000000 00000000 bbe21fac bbe21f28 [ 31.604902] 1f20: 800446cc 8003f234 bbe21f44 00000000 80062798 bd585680 00000000 00000000 [ 31.613087] 1f40: dead4ead ffffffff ffffffff 809c9f34 00000000 00000000 80838390 bbe21f5c [ 31.621275] 1f60: bbe21f5c 00000000 00000000 dead4ead ffffffff ffffffff 809c9f34 00000000 [ 31.629461] 1f80: 00000000 80838390 bbe21f88 bbe21f88 bdad17c0 800445f0 00000000 00000000 [ 31.637648] 1fa0: 00000000 bbe21fb0 8000ed68 800445fc 00000000 00000000 00000000 00000000 [ 31.645834] 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 31.654019] 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 [ 31.662200] Backtrace: [ 31.664680] [<8052b284>] (soc_dapm_dai_stream_event) from [<8052e5cc>] (snd_soc_dapm_stream_event+) [ 31.674512] r5:ba945810 r4:00000000 [ 31.678134] [<8052e568>] (snd_soc_dapm_stream_event) from [<80530038>] (close_delayed_work+0x4c/0x) [ 31.687358] r8:be7c0c00 r7:bbe21eb0 r6:ba945810 r5:ba94581c r4:ba945dd4 r3:00000000 [ 31.695206] [<8052ffec>] (close_delayed_work) from [<8003ef74>] (process_one_work+0x194/0x40c) [ 31.703821] r6:be7bd380 r5:ba945dd4 r4:bd585680 r3:8052ffec [ 31.709558] [<8003ede0>] (process_one_work) from [<8003f388>] (worker_thread+0x160/0x48c) [ 31.717739] r10:be7bd380 r9:be7bd380 r8:00000008 r7:bd585698 r6:809c7882 r5:be7bd3b0 [ 31.725659] r4:bd585680 [ 31.728220] [<8003f228>] (worker_thread) from [<800446cc>] (kthread+0xdc/0xf8) [ 31.735447] r10:00000000 r9:00000000 r8:00000000 r7:8003f228 r6:bd585680 r5:bdad17c0 [ 31.743365] r4:00000000 [ 31.745931] [<800445f0>] (kthread) from [<8000ed68>] (ret_from_fork+0x14/0x2c) [ 31.753158] r7:00000000 r6:00000000 r5:800445f0 r4:bdad17c0 [ 31.758895] Code: e92dd830 e24cb004 e3510000 e1a05002 (0590401c) [ 31.765088] ---[ end trace c7913ba4763772ee ]--- Seems it's because when playback stream is active soc_cleanup_card_resources() -> snd_card_free() Will trigger soc_pcm_close() which will queue close_delayed_work(), and then when close_delayed_work() is scheduled the necessary resource has already been released. I tried to set rtd->pmdown_time to 0 in soc_cleanup_card_resources() to avoid any new delayed_work be queued. BUT when unload audio drivers, I got another kernel crash ######################### [ 28.234740] Unable to handle kernel paging request at virtual address 6faaaf90 [ 28.242075] pgd = bbde0000 [ 28.244818] [6faaaf90] *pgd=00000000 [ 28.248511] Internal error: Oops: 5 [#1] SMP ARM [ 28.253155] Modules linked in: snd_soc_imx_wm8962(-) snd_soc_imx_audmux snd_soc_wm8962 snd_soc_fsl] [ 28.267055] CPU: 1 PID: 188 Comm: pulseaudio Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #7 [ 28.275435] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 28.281991] task: bc3e1d40 ti: bbc1e000 task.ti: bbc1e000 [ 28.287431] PC is at dapm_seq_insert+0x34/0xf4 [ 28.291902] LR is at dapm_power_widgets+0x234/0x798 [ 28.296809] pc : [<8052ad74>] lr : [<8052d090>] psr: 600f0113 [ 28.296809] sp : bbc1fdd0 ip : 809bdd90 fp : bbc1fdec [ 28.308308] r10: bbc3bb90 r9 : bc0405c4 r8 : bc0405ac [ 28.313554] r7 : bbc1fe24 r6 : 809bdd90 r5 : bbc3b480 r4 : bbc3b134 [ 28.320093] r3 : bbc3b0c0 r2 : 00000000 r1 : bbc1fe1c r0 : bbc3bb40 [ 28.326628] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user [ 28.333771] Control: 10c5387d Table: 4bde004a DAC: 00000015 [ 28.339524] Process pulseaudio (pid: 188, stack limit = 0xbbc1e238) [ 28.345797] Stack: (0xbbc1fdd0 to 0xbbc20000) [ 28.350162] fdc0: 00000000 bbc3bb40 bbc3bba4 bbc1fe24 [ 28.358349] fde0: bbc1fe5c bbc1fdf0 8052d090 8052ad4c 80062798 8006259c bbc1fe5c 00000002 [ 28.366535] fe00: bc04046c bc0405bc bbc1fe1c bbc1fe14 bbc1fe44 bbc1fe14 bbc1fe14 bbc3b134 [ 28.374721] fe20: bbc3b134 bbc1fe24 bbc1fe24 bbc3bb40 00000004 00000001 bd916010 00000002 [ 28.382907] fe40: 00000000 bc0404cc 00000000 bd91601c bbc1fe84 bbc1fe60 8052e5f4 8052ce68 [ 28.391093] fe60: 00000000 bd916010 00000000 bc2c5a00 bbebb800 bdbe720c bbc1feac bbc1fe88 [ 28.399279] fe80: 805317cc 8052e57c bc2c5a00 bc2c5a00 bc3d9b80 bbdf7640 bc2c4ae8 00000008 [ 28.407465] fea0: bbc1fec4 bbc1feb0 8051b610 80531664 bc2c4a00 bc2c5a00 bbc1feec bbc1fec8 [ 28.415651] fec0: 8051b6a0 8051b5c0 8051b664 bbdf75c0 809bd508 bc3d9b80 bdba8f30 bdba8f30 [ 28.423837] fee0: bbc1ff0c bbc1fef0 8050eb2c 8051b670 bc3d9b80 bdba8f30 bd9b9850 bb8f75c0 [ 28.432023] ff00: bbc1ff4c bbc1ff10 800ef5bc 8050ea50 00000000 00000000 80042d68 bc3d9b88 [ 28.440208] ff20: bc3d9b80 bc3e20f0 00000000 809c9f14 bc3e1d40 00000000 bbc1e000 00000000 [ 28.448393] ff40: bbc1ff5c bbc1ff50 800ef760 800ef538 bbc1ff84 bbc1ff60 80042ec0 800ef75c [ 28.456579] ff60: bc3d9b80 8000ee64 bbc1ffb0 00000000 00000006 8000ee64 bbc1ffac bbc1ff88 [ 28.464765] ff80: 80011884 80042e14 8000ecac 00000004 00000000 0116aa50 0116ab58 0118e100 [ 28.472950] ffa0: 00000000 bbc1ffb0 8000ecf8 800117dc 00000000 76fcf084 76fcf4c0 725fcc94 [ 28.481136] ffc0: 0116aa50 0116ab58 0118e100 00000006 01118168 4ce839e4 00000000 00000000 [ 28.489322] ffe0: 00000000 7e957978 4e1e6d24 4e1e6d34 800f0010 00000016 4eff6821 4eff6c21 [ 28.497501] Backtrace: [ 28.499980] [<8052ad40>] (dapm_seq_insert) from [<8052d090>] (dapm_power_widgets+0x234/0x798) [ 28.508509] r7:bbc1fe24 r6:bbc3bba4 r5:bbc3bb40 r4:00000000 [ 28.514251] [<8052ce5c>] (dapm_power_widgets) from [<8052e5f4>] (snd_soc_dapm_stream_event+0x84/0x) [ 28.523473] r10:bd91601c r9:00000000 r8:bc0404cc r7:00000000 r6:00000002 r5:bd916010 [ 28.531393] r4:00000001 [ 28.533959] [<8052e570>] (snd_soc_dapm_stream_event) from [<805317cc>] (soc_pcm_close+0x174/0x28c) [ 28.542920] r8:bdbe720c r7:bbebb800 r6:bc2c5a00 r5:00000000 r4:bd916010 r3:00000000 [ 28.550764] [<80531658>] (soc_pcm_close) from [<8051b610>] (snd_pcm_release_substream+0x5c/0xb0) [ 28.559552] r10:00000008 r8:bc2c4ae8 r7:bbdf7640 r6:bc3d9b80 r5:bc2c5a00 r4:bc2c5a00 [ 28.567479] [<8051b5b4>] (snd_pcm_release_substream) from [<8051b6a0>] (snd_pcm_release+0x3c/0x88) [ 28.576440] r5:bc2c5a00 r4:bc2c4a00 [ 28.580062] [<8051b664>] (snd_pcm_release) from [<8050eb2c>] (snd_disconnect_release+0xe8/0xf8) [ 28.588764] r8:bdba8f30 r7:bdba8f30 r6:bc3d9b80 r5:809bd508 r4:bbdf75c0 r3:8051b664 [ 28.596604] [<8050ea44>] (snd_disconnect_release) from [<800ef5bc>] (__fput+0x90/0x1d4) [ 28.604611] r7:bb8f75c0 r6:bd9b9850 r5:bdba8f30 r4:bc3d9b80 [ 28.610345] [<800ef52c>] (__fput) from [<800ef760>] (____fput+0x10/0x14) [ 28.617050] r10:00000000 r9:bbc1e000 r8:00000000 r7:bc3e1d40 r6:809c9f14 r5:00000000 [ 28.624968] r4:bc3e20f0 [ 28.627535] [<800ef750>] (____fput) from [<80042ec0>] (task_work_run+0xb8/0xe8) [ 28.634865] [<80042e08>] (task_work_run) from [<80011884>] (do_work_pending+0xb4/0xd4) [ 28.642786] r8:8000ee64 r7:00000006 r6:00000000 r5:bbc1ffb0 r4:8000ee64 r3:bc3d9b80 [ 28.650624] [<800117d0>] (do_work_pending) from [<8000ecf8>] (work_pending+0xc/0x20) [ 28.658371] r6:0118e100 r5:0116ab58 r4:0116aa50 [ 28.663047] Code: e5905000 e3520000 e24c6070 01a0600c (e7965105) [ 28.669220] ---[ end trace 69b76cbdebd80b05 ]--- >From the backtrace, seems it's because when soc_pcm_close() is called, it will power widget due to stream event, but at that time, all widgets have already been freed by soc_remove_dai_links() Currently I am out of idea how to fix this issue, As I think this is a generic issue not i.MX platform specific, Someone in community may has already find this issue, Could anyone give some inputs/suggestion to fix this issue? Regards & Thanks, Jiada

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