January 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 13:49:22 +0100, Jens Axboe wrote: > > On 2013-01-10 11:21, Takashi Iwai wrote: > > At Thu, 10 Jan 2013 09:23:48 +0100, > > Jens Axboe wrote: > >> > >> Hi, > >> > >> I have a USB DAC that I use for music. Just upgraded my workstation from > >> 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > >> an immediate oops. I have attached the picture. It's crashing here: > >> > >> (gdb) l *snd_usb_pcm_prepare+0x153 > >> 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > >> 465 snd_pcm_format_t pcm_format) > >> 466 { > >> 467 int i; > >> 468 int score = 0; > >> 469 > >> 470 if (fp->channels < 1) { > >> 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > >> 472 return 0; > >> 473 } > >> 474 > >> > >> 'fp' is clearly NULL. > > > > Hmm... it's a function called only from a single place in > > configure_sync_endpoint() and it's in list_for_each_entry(): > > > > list_for_each_entry(fp, &sync_subs->fmt_list, list) { > > int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, > > subs->cur_rate, subs->pcm_format); > > > > so it must be hardly NULL in normal situations. > > The only scenario I can imagine right now is that the whole structure > > is still uninitialized while it's called. If so, it's a race problem, > > and shouldn't be new to 3.8. A patch like below might paper over the > > problem although it's far from perfect. > > > > > >> Let me know if you want a bisect. > > > > Yeah, that'll be really helpful. > > Here it is, it's from the one introducing the audioformat lookup. > Confirmed that 3.8-rc3 with this backed out works fine, too. So should > be fairly confident in that result. Thanks! I'm slowly understanding what's happening there. Could you try the patch below? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: usb-audio: Fix NULL dereference by access to non-existing substream The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-by: Jens Axboe <axboe(a)kernel.dk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..21c0001 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -511,6 +511,17 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) struct snd_usb_substream *sync_subs = &subs->stream->substream[subs->direction ^ 1]; + if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA || + !subs->stream) { + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + } + /* Try to find the best matching audioformat. */ list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, -- 1.8.1

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[alsa-devel] [PATCH] ASoC: dapm: Fix sense of regulator bypass mode
by Mark Brown 10 Jan '13

10 Jan '13

Enable bypass when the regulator is idle, not when it is in use. This is consistent with what the few existing users actually want. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc18d78..17dba18 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, true); + ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to bypass %s: %d\n", @@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, false); + ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to unbypass %s: %d\n", @@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } + + if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + ret = regulator_allow_bypass(w->regulator, true); + if (ret != 0) + dev_warn(w->dapm->dev, + "ASoC: Failed to unbypass %s: %d\n", + w->name, ret); + } break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP -- 1.7.10.4

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[alsa-devel] [GIT PULL] ASoC updates for v3.8
by Mark Brown 10 Jan '13

10 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-3.8-rc2 for you to fetch changes up to 49a170bcf28af5ef372cdbed7750ba8d4162bb0e: Merge remote-tracking branch 'asoc/fix/wm5100' into tmp (2013-01-10 12:22:30 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.8 Nothing terribly exciting here except for the DOUBLE_RANGE fix which just hadn't worked before, nobody noticed due to lack of use. ---------------------------------------------------------------- Axel Lin (5): ASoC: sta529: Fix update register bits in sta529_set_dai_fmt ASoC: wm2200: Fix setting dai format in wm2200_set_fmt ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt() ASoC: arizona: Do proper shift for setting AIF rate ASoC: cs42l52: Catch no-match case in cs42l52_get_clk Chuansheng Liu (2): ASoC: core: fix the memory leak in case of device_add() failure ASoC: core: fix the memory leak in case of remove_aux_dev() Daniel Mack (2): ASoC: cs4271: fix sparse warning ASoC: cs4271: fix property check Fabio Estevam (2): ASoC: soc-core: Remove unused 'ret' variable ASoC: sgtl5000: Fix maximum value for microphone gain MR.Swami.Reddy(a)ti.com (2): ASoC: lm49453: Fix adc, mic and sidetone volume ranges ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B Mark Brown (20): ASoC: core: Fix SOC_DOUBLE_RANGE() macros ASoC: arizona: Correct FLL source definitions ASoC: wm5102: Improve speaker enable performance ASoC: wm2200: Remove DSP B and left justified AIF modes ASoC: arizona: Remove DSP B and left justified AIF modes ASoC: wm5100: Remove DSP B and left justified formats ASoC: wm2000: Fix sense of speech clarity enable ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/cs4271' into tmp Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp Merge remote-tracking branch 'asoc/fix/lm49453' into tmp Merge remote-tracking branch 'asoc/fix/pxa' into tmp Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp Merge remote-tracking branch 'asoc/fix/sta529' into tmp Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Merge remote-tracking branch 'asoc/fix/wm5100' into tmp Mike Dunn (2): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset Patrick Lai (1): ASoC: pcm: allow backend hardware to be freed in pause state arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 + arch/arm/mach-pxa/pxa27x.c | 4 +- include/sound/cs4271.h | 2 +- include/sound/soc.h | 10 ++- sound/arm/pxa2xx-ac97-lib.c | 26 ++++++- sound/soc/codecs/arizona.c | 9 +-- sound/soc/codecs/arizona.h | 18 ++--- sound/soc/codecs/cs4271.c | 6 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/lm49453.c | 106 ++++++++++----------------- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta529.c | 9 ++- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2200.c | 8 +- sound/soc/codecs/wm5100.c | 6 -- sound/soc/codecs/wm5102.c | 48 +++++++++++- sound/soc/codecs/wm_adsp.c | 23 +++++- sound/soc/soc-core.c | 35 ++++++++- sound/soc/soc-pcm.c | 1 + 19 files changed, 201 insertions(+), 125 deletions(-)

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[alsa-devel] [PATCH RFC 0/8] Use generic parser for HD-audio codec drivers
by Takashi Iwai 10 Jan '13

10 Jan '13

Hi, this is the next series of patches to convert various codec drivers to use the new generic parser code. Most of changes are obvious, but some places it gets tricky. As mentioned in another mail, the patches are found in sound-unstable git tree test/hda-migrate branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git Takashi

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[alsa-devel] [PATCH] ASoC: ak4642: add Device Tree support
by Kuninori Morimoto 10 Jan '13

10 Jan '13

Support for loading the ak4642 codec module via devicetree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/ak4642.txt | 17 ++++++++++ sound/soc/codecs/ak4642.c | 33 ++++++++++++++++++-- 2 files changed, 48 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt new file mode 100644 index 0000000..623d4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.txt @@ -0,0 +1,17 @@ +AK4642 I2C transmitter + +This device supports I2C mode only. + +Required properties: + + - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" + - reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4648: ak4648@0x12 { + compatible = "asahi-kasei,ak4642"; + reg = <0x12>; + }; +}; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f0cdab..c78794d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/initval.h> @@ -513,12 +514,31 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct device_node *np = i2c->dev.of_node; + const struct snd_soc_codec_driver *driver; + + driver = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4642_of_match, &i2c->dev); + if (of_id) + driver = of_id->data; + } else { + driver = (struct snd_soc_codec_driver *)id->driver_data; + } + + if (!driver) { + dev_err(&i2c->dev, "no driver\n"); + return -EINVAL; + } + return snd_soc_register_codec(&i2c->dev, - (struct snd_soc_codec_driver *)id->driver_data, - &ak4642_dai, 1); + driver, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -527,6 +547,14 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } +static struct of_device_id ak4642_of_match[] __devinitconst = { + { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4642_of_match); + static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, @@ -539,6 +567,7 @@ static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", .owner = THIS_MODULE, + .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, .remove = ak4642_i2c_remove, -- 1.7.9.5

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Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 09:23:48 +0100, Jens Axboe wrote: > > Hi, > > I have a USB DAC that I use for music. Just upgraded my workstation from > 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > an immediate oops. I have attached the picture. It's crashing here: > > (gdb) l *snd_usb_pcm_prepare+0x153 > 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > 465 snd_pcm_format_t pcm_format) > 466 { > 467 int i; > 468 int score = 0; > 469 > 470 if (fp->channels < 1) { > 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > 472 return 0; > 473 } > 474 > > 'fp' is clearly NULL. Hmm... it's a function called only from a single place in configure_sync_endpoint() and it's in list_for_each_entry(): list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, subs->cur_rate, subs->pcm_format); so it must be hardly NULL in normal situations. The only scenario I can imagine right now is that the whole structure is still uninitialized while it's called. If so, it's a race problem, and shouldn't be new to 3.8. A patch like below might paper over the problem although it's far from perfect. > Let me know if you want a bisect. Yeah, that'll be really helpful. Or, at least, could you try to copy sound/usb/* from 3.7 to 3.8 and see whether it's really a regression there? thanks, Takashi --- diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ad181d5..3bfc564 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -81,8 +81,7 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) */ static void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp) + int stream) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -96,6 +95,13 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->speed = snd_usb_get_speed(subs->dev); snd_usb_set_pcm_ops(as->pcm, stream); +} + +static void snd_usb_add_substream_format(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; list_add_tail(&fp->list, &subs->fmt_list); subs->formats |= fp->formats; @@ -342,7 +348,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp); + snd_usb_add_substream_format(as, stream, fp); return add_chmap(as->pcm, stream, subs); } @@ -370,7 +376,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strcpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_PLAYBACK); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_CAPTURE); + snd_usb_add_substream_format(as, stream, fp); list_add(&as->list, &chip->pcm_list); chip->pcm_devs++;

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[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Make ebox44_table static
by Sachin Kamat 10 Jan '13

10 Jan '13

Fixes the following sparse warning: sound/usb/mixer_quirks.c:1209:23: warning: symbol 'ebox44_table' was not declared. Should it be static? Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/mixer_quirks.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0422b13..15520de 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -struct std_mono_table ebox44_table[] = { +static struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, -- 1.7.4.1

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[alsa-devel] [PATCH] ALSA: hdspm: Fix wordclock status on AES32
by Andre Schramm 09 Jan '13

09 Jan '13

Use correct bitmask for AES32 cards to determine wordclock lock state, add missing bitmask for sync check and make output of the corresponding control and /proc coherent. Signed-off-by: Andre Schramm <andre.schramm(a)iosono-sound.com> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e02e06..36e247f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); */ /* status */ #define HDSPM_AES32_wcLock 0x0200000 +#define HDSPM_AES32_wcSync 0x0100000 #define HDSPM_AES32_wcFreq_bit 22 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ @@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) switch (hdspm->io_type) { case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_wcSync) - return 2; - else if (status & HDSPM_wcLock) - return 1; + if (status & HDSPM_AES32_wcLock) { + if (status & HDSPM_AES32_wcSync) + return 2; + else + return 1; + } return 0; break; @@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, unsigned int status; unsigned int status2; unsigned int timecode; + unsigned int wcLock, wcSync; int pref_syncref; char *autosync_ref; int x; @@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Status:\n"); + wcLock = status & HDSPM_AES32_wcLock; + wcSync = wcLock && (status & HDSPM_AES32_wcSync); + snd_iprintf(buffer, "Word: %s Frequency: %d\n", - (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock", + (wcLock) ? (wcSync ? "Sync " : "Lock") : "No Lock", HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF)); for (x = 0; x < 8; x++) { -- 1.7.1

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[alsa-devel] [PATCH] Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
by David Henningsson 09 Jan '13

09 Jan '13

This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. The original patch was meant to remove clicking, but in fact caused even more clicking instead. Thanks to c4pp4 for doing most of the work with this bug. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- @c4pp4: Is this consistent with what you have done to reduce the clicking? Will you recommend upstream Linux to take the patch below? sound/pci/hda/patch_conexant.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60890bf..dd798c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM -static int conexant_suspend(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); - return 0; -} -#endif - static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; /* -- 1.7.9.5

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Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 09 Jan '13

09 Jan '13

At Tue, 08 Jan 2013 21:25:21 +0100, Vincent Blut wrote: > > Le mardi 08 janvier 2013 à 13:56 +0100, Takashi Iwai a écrit : > > At Tue, 08 Jan 2013 13:28:55 +0100, > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Disable runtime D3 for Intel CPT & co > > > > We've got a few bug reports that the runtime D3 results in the dead > > HD-audio controller. It seems that the problem is in a deeper level > > than the sound driver itself, so as a temporal solution, disable the > > feature for these controllers again. > > > > Reported-by: Vincent Blut <vincent.debian(a)free.fr> > > Reported-by: Maurizio Avogadro <mavoga(a)gmail.com> > > Cc: <stable(a)vger.kernel.org> [v3.7] > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/hda_intel.c | 13 ++++++++----- > > 1 file changed, 8 insertions(+), 5 deletions(-) > > > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index cca8727..0b6aeba 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -573,9 +573,12 @@ enum { > > #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ > > > > /* quirks for Intel PCH */ > > -#define AZX_DCAPS_INTEL_PCH \ > > +#define AZX_DCAPS_INTEL_PCH_NOPM \ > > (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ > > - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) > > + AZX_DCAPS_COUNT_LPIB_DELAY) > > + > > +#define AZX_DCAPS_INTEL_PCH \ > > + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) > > > > /* quirks for ATI SB / AMD Hudson */ > > #define AZX_DCAPS_PRESET_ATI_SB \ > > @@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci) > > static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > > /* CPT */ > > { PCI_DEVICE(0x8086, 0x1c20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* PBG */ > > { PCI_DEVICE(0x8086, 0x1d20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Panther Point */ > > { PCI_DEVICE(0x8086, 0x1e20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Lynx Point */ > > { PCI_DEVICE(0x8086, 0x8c20), > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > Well, my sound controller behaves nicely with the above quirk. > > Tested-by: Vincent Blut <vincent.debian(a)free.fr> OK, I queued the patch now. thanks, Takashi

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