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

[alsa-devel] async between dmaengine_pcm_dma_complete and snd_pcm_release
by Qiao Zhou 05 Nov '13

05 Nov '13

Hi Mark, Liam, Jaroslav, Takashi I met an issue in which kernel panic appears in dmaengine_pcm_dma_complete function on a quad-core system. The dmaengine_pcm_dma_complete is running core0, while snd_pcm_release has already been executed on core1, due to in low memory stress oom killer kills the audio thread to release some memory. snd_pcm_release frees the runtime parameters, and runtime is used in dmaengine_pcm_dma_complete, which is a callback from tasklet in dmaengine. In current audio driver, we can't promise that dmaengine_pcm_dma_complete is not executed after snd_pcm_release on multi cores. Maybe we should add some protection. Do you have any suggestion? I have tried to apply below workaround, which can fix the panic, but I'm not confident it's proper. Need your comment and better suggestion. From d568a88e8f66ee21d44324bdfb48d2a3106cf0d1 Mon Sep 17 00:00:00 2001 From: Qiao Zhou <zhouqiao(a)marvell.com> Date: Wed, 9 Oct 2013 15:24:29 +0800 Subject: [PATCH] ASoC: soc-dmaengine: add mutex to protect runtime param under SMP arch, the dmaengine_pcm_dma_complete callback has the chance to run on one cpu while at the same time, the substream is released on another cpu. thus it may access param which is already freed. we need to add mutes to protect such access, and check PCM availability before using it. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-dmaengine-pcm.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d9..5917029 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -125,13 +125,22 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); static void dmaengine_pcm_dma_complete(void *arg) { struct snd_pcm_substream *substream = arg; - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dmaengine_pcm_runtime_data *prtd; + + mutex_lock(&substream->pcm->open_mutex); + if (!substream || !substream->runtime) { + mutex_unlock(&substream->pcm->open_mutex); + return; + } + + prtd = substream_to_prtd(substream); prtd->pos += snd_pcm_lib_period_bytes(substream); if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) prtd->pos = 0; snd_pcm_period_elapsed(substream); + mutex_unlock(&substream->pcm->open_mutex); } static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) -- 1.7.0.4 -- Best Regards Qiao

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Re: [alsa-devel] [Alsa-user] hdmi sound rerouted?
by Dagg Stompler 05 Nov '13

05 Nov '13

> ----- Original Message ----- > From: Takashi Iwai > Sent: 11/04/13 04:06 PM > To: Deucher, Alexander > Subject: Re: [alsa-devel] [Alsa-user] hdmi sound rerouted? > > At Mon, 4 Nov 2013 13:47:54 +0000, > Deucher, Alexander wrote: > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Monday, November 04, 2013 8:05 AM > > > To: daggs(a)gmx.com > > > Cc: Deucher, Alexander; Clemens Ladisch; alsa-user(a)lists.sourceforge.net; > > > alsa-devel(a)alsa-project.org > > > Subject: Re: [alsa-devel] [Alsa-user] hdmi sound rerouted? > > > > > > At Mon, 04 Nov 2013 13:21:25 +0100, > > > daggs(a)gmx.com wrote: > > > > > > > > the original issue is that wen using an ATI gpu with hdmi in it, rerouting the > > > sound via the ati hdmi card doesn't work, only via the generic. > > > > > > > > alsa devs pointed out that the ati cards listed in intel_hda.c aren't updated > > > with latest cards due to insufficient info on about the cards. > > > > > > > > Alex Deucher from AMD's dev answered that he will query the right team > > > in his workplace but no response was received fro him yet. > > > > > > OK, then could you alsa-info.sh output? > > > > > > The situation in the upstream has changed since we have a lot more > > > AMD-specific HDMI codes in the latest driver for 3.13 kernel. So, > > > it's not only a matter of snoop flag. > > > > I was specifically trying to get clarification on which parts needed the snoop flag. Is there some other larger issue? From the feedback I've seen lately, audio has been working pretty well on AMD GPUs with Anssi's patches. > > That's why I started asking :) The recent code should cover known > issues of AMD HDMI codecs, at least. But if there is a new chip, we'd > need to test and add the corresponding entry. > > > Takashi alsa info can be found at http://www.alsa-project.org/db/?f=590cb353dd93e9ea5f9da3e072cdcd5d464e8609 hope it helps

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[alsa-devel] [PATCH 0/3] ALSA: usb channel map fixes
by David Henningsson 05 Nov '13

05 Nov '13

Ok, so I actually tested my patch, and it turned out things were slightly more complex than expected - the original patch is still needed, but in addition, we also need to build the channel map after the format parse, because the format of the descriptor is not always the same as the format of the input terminal. These two fixes made the mono chmap show up on my logitech USB headset, on both output and input side. I also tested my patches with another USB device. This device is UAC2, and when looking at lsusb I found that in UAC2 the streaming descriptor has a channel config too. Given what I found on the headset, it seems reasonable to use this channel config instead of the input terminal's channel config, although in my case it made no practical difference. David Henningsson (3): ALSA: usb: supply channel maps even when wChannelConfig is unspecified ALSA: usb - For class 2 devices, use channel map from altsettings ALSA: usb - Don't trust the channel config if the channel count changed sound/usb/stream.c | 29 +++++++++++++++++++++-------- 1 file changed, 21 insertions(+), 8 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] hda-emu: Add AMD HDMI verbs
by David Henningsson 05 Nov '13

05 Nov '13

Source of information: http://www.x.org/docs/AMD/AMD_HDA_verbs_v2.pdf Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-spec.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/hda-spec.c b/hda-spec.c index e4bb547..d70e54b 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -213,6 +213,49 @@ static struct xhda_verb_table vt2002p_verbs[] = { { } /* terminator */ }; +static struct xhda_verb_table amdhdmi_verbs[] = { + /* Refer to AMD HDA Verbs specification */ + { 0x776, set_cached_verb, "amdhdmi_set_descriptor" }, + { 0xf76, get_cached_verb, "amdhdmi_get_descriptor" }, + { 0x770, set_cached_verb, "amdhdmi_set_ddm_cvt_sync" }, + { 0xf70, get_cached_verb, "amdhdmi_get_speaker_alloc/cvt_sync" }, + { 0xf93, get_cached_verb, "amdhdmi_get_ddm_audio_assoc" }, + + { 0x777, set_cached_verb, "amdhdmi_set_multich_en_01" }, + { 0xf77, get_cached_verb, "amdhdmi_get_multich_en_01" }, + { 0x778, set_cached_verb, "amdhdmi_set_multich_en_23" }, + { 0xf78, get_cached_verb, "amdhdmi_get_multich_en_23" }, + { 0x779, set_cached_verb, "amdhdmi_set_multich_en_45" }, + { 0xf79, get_cached_verb, "amdhdmi_get_multich_en_45" }, + { 0x77a, set_cached_verb, "amdhdmi_set_multich_en_67" }, + { 0xf7a, get_cached_verb, "amdhdmi_get_multich_en_67" }, + + { 0x771, set_cached_verb, "amdhdmi_set_channel_alloc" }, + { 0xf71, get_cached_verb, "amdhdmi_get_channel_alloc" }, + { 0x772, set_cached_verb, "amdhdmi_set_downmix_info" }, + { 0xf72, get_cached_verb, "amdhdmi_get_downmix_info" }, + { 0x789, set_cached_verb, "amdhdmi_set_multich_mode" }, + { 0xf89, get_cached_verb, "amdhdmi_get_multich_mode" }, + + { 0x785, set_cached_verb, "amdhdmi_set_multich_en_1" }, + { 0xf85, get_cached_verb, "amdhdmi_get_multich_en_1" }, + { 0x786, set_cached_verb, "amdhdmi_set_multich_en_3" }, + { 0xf86, get_cached_verb, "amdhdmi_get_multich_en_3" }, + { 0x787, set_cached_verb, "amdhdmi_set_multich_en_5" }, + { 0xf87, get_cached_verb, "amdhdmi_get_multich_en_5" }, + { 0x788, set_cached_verb, "amdhdmi_set_multich_en_7" }, + { 0xf88, get_cached_verb, "amdhdmi_get_multich_en_7" }, + + { 0x77c, set_cached_verb, "amdhdmi_set_hbr_control" }, + { 0xf7c, get_cached_verb, "amdhdmi_get_hbr_control" }, + { 0xf7b, get_cached_verb, "amdhdmi_get_lipsync" }, + { 0x780, set_cached_verb, "amdhdmi_set_sinkinfo_index" }, + { 0xf80, get_cached_verb, "amdhdmi_get_sinkinfo_index" }, + { 0xf81, get_cached_verb, "amdhdmi_get_sinkinfo_data" }, + + { } /* terminator */ +}; + /* */ @@ -351,6 +394,8 @@ static struct verb_ext_list extensions[] = { { .id = 0x11060446, .verbs = vt2002p_verbs }, { .id = 0x11068446, .verbs = vt2002p_verbs }, + { .id = 0x1002aa01, .verbs = amdhdmi_verbs }, + { } }; -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Enable SPDIF for Acer TravelMate 6293
by Takashi Iwai 04 Nov '13

04 Nov '13

BIOS on Acer TravelMate 6293 doesn't set up the SPDIF output pin correctly as default, so enable it via a fixup entry. Reported-and-tested-by: Hagen Heiduck <heiduck.suse(a)fmail.postpro.net> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d7c86d81196..01bf812455a2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2386,6 +2386,7 @@ static const struct hda_verb alc268_beep_init_verbs[] = { enum { ALC268_FIXUP_INV_DMIC, ALC268_FIXUP_HP_EAPD, + ALC268_FIXUP_SPDIF, }; static const struct hda_fixup alc268_fixups[] = { @@ -2400,6 +2401,13 @@ static const struct hda_fixup alc268_fixups[] = { {} } }, + [ALC268_FIXUP_SPDIF] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1e, 0x014b1180 }, /* enable SPDIF out */ + {} + } + }, }; static const struct hda_model_fixup alc268_fixup_models[] = { @@ -2409,6 +2417,7 @@ static const struct hda_model_fixup alc268_fixup_models[] = { }; static const struct snd_pci_quirk alc268_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0139, "Acer TravelMate 6293", ALC268_FIXUP_SPDIF), SND_PCI_QUIRK(0x1025, 0x015b, "Acer AOA 150 (ZG5)", ALC268_FIXUP_INV_DMIC), /* below is codec SSID since multiple Toshiba laptops have the * same PCI SSID 1179:ff00 -- 1.8.4.2

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[alsa-devel] [PATCH v4] ASoC: Add pinctrl PM to components of active DAIs
by Nicolin Chen 04 Nov '13

04 Nov '13

It's quite popular that more drivers are using pinctrl PM, for example: (Documentation/devicetree/bindings/arm/primecell.txt). Just like what runtime PM does, it would deactivate and activate pin group depending on whether it's being used or not. And this pinctrl PM might be also beneficial to cpu dai drivers because they might have actual pinctrl so as to sleep their pins and wake them up as needed. To achieve this goal, this patch sets pins to the default state during resume or startup; While during suspend and shutdown, it would set pins to the sleep state. As pinctrl PM would return zero if there is no such pinctrl sleep state settings, this patch would not break current ASoC subsystem directly. [ However, there is still an exception that the patch can not handle, that is, when cpu dai driver does not have pinctrl property but another device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example. SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio Multiplexer, has it). In this case, this kind of cpu dai driver needs to find a way to obtain the pinctrl property as its own, by moving property from AUDMUX to SSI, or creating a pins link/dependency between these two devices, or using a more decent way after we figure it out. ] Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog: v4: * Fix build break happened to x86 (missing header file) v3: * Add pinctrl_pm() for codec_dai/codec * Add more proper active check for each pinctrl_pm_select_xx() v2: * Add cpu_dai->active check sound/soc/soc-core.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 11 +++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bdc1d74..4e53d87 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -662,6 +662,8 @@ int snd_soc_suspend(struct device *dev) codec->cache_sync = 1; if (codec->using_regmap) regcache_mark_dirty(codec->control_data); + /* deactivate pins to sleep state */ + pinctrl_pm_select_sleep_state(codec->dev); break; default: dev_dbg(codec->dev, @@ -679,6 +681,9 @@ int snd_soc_suspend(struct device *dev) if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); + + /* deactivate pins to sleep state */ + pinctrl_pm_select_sleep_state(cpu_dai->dev); } if (card->suspend_post) @@ -807,6 +812,16 @@ int snd_soc_resume(struct device *dev) if (list_empty(&card->codec_dev_list)) return 0; + /* activate pins from sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + if (cpu_dai->active) + pinctrl_pm_select_default_state(cpu_dai->dev); + if (codec_dai->active) + pinctrl_pm_select_default_state(codec_dai->dev); + } + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume @@ -1929,6 +1944,14 @@ int snd_soc_poweroff(struct device *dev) snd_soc_dapm_shutdown(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + pinctrl_pm_select_sleep_state(codec_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_poweroff); @@ -3767,6 +3790,16 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret != 0) soc_cleanup_card_debugfs(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d81b792..3a7e78f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -19,6 +19,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/pinctrl/consumer.h> #include <linux/pm_runtime.h> #include <linux/slab.h> #include <linux/workqueue.h> @@ -183,6 +184,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); + pinctrl_pm_select_default_state(codec_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); @@ -317,6 +320,10 @@ out: pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); return ret; } @@ -426,6 +433,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); return 0; } -- 1.8.4

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[alsa-devel] [PATCH] ASoC: fsl: fsl_spdif: No need to check the return value of platform_get_resource()
by Fabio Estevam 04 Nov '13

04 Nov '13

When using devm_ioremap_resource(), we do not need to check the return value of platform_get_resource(), so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 76c742a..55193a5 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1107,11 +1107,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "could not determine device resources\n"); - return -ENXIO; - } - regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(regs)) return PTR_ERR(regs); -- 1.8.1.2

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[alsa-devel] [PATCH v2] ASoC: wm_adsp: Interpret ADSP memory region lengths as 32 bit words
by Dimitris Papastamos 04 Nov '13

04 Nov '13

From: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Pad the ADSP word (3 bytes) to 4 bytes in the kernel and calculate lengths based on padded ADSP words instead of treating them as bytes Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 53b6033..df95e5a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1066,6 +1066,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", @@ -1083,6 +1084,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", @@ -1112,6 +1114,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", @@ -1129,6 +1132,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", @@ -1146,6 +1150,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", -- 1.8.4.2

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[alsa-devel] What about symmetric_channels and symmetric_samplebits?
by Nicolin Chen 04 Nov '13

04 Nov '13

Hi all, As far as I know, some SoCs can only work in mono or stereo mode at one time. So if we let them capture a mono stream while playing a stereo stream, there might be a problem occur to one of these two streams: double-paced or slowed down. I just want to ask: Is there any existing solution for the case? In soc-pcm.c, we have soc_pcm_apply_symmetry() to solve unmatched sample rates situation for simultaneous playback and capture. But we don't have one for channels. Is that reasonable to add a similar one for it? Likewise, we can treat symmetric_rate as a solution for those SoCs or CODECs which can not handle asymmetrical LRCLK. But it's also impossible for them to handle asymmetrical BCLK. And accodring to BCLK = LRCLK * channel number * slot size(fixed or sample bits), sample bits might also be a problem if they are not using a fixed slot size. Thank you, Nicolin Chen

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[alsa-devel] [PATCH] ALSA: hda - Introduce the bitmask for excluding output volume
by Takashi Iwai 04 Nov '13

04 Nov '13

Add a bitmask to hda_gen_spec indicating NIDs to exclude from the possible volume controls. That is, when the bit is set, the NID corresponding to the bit won't be picked as an output volume control any longer. Basically this is just a band-aid for working around the issue found with CS4208 codec, where only the headphone pin has a volume AMP with different dB steps. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60811 Cc: <stable(a)vger.kernel.org> [v3.12+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 ++++++-- sound/pci/hda/hda_generic.h | 3 +++ sound/pci/hda/patch_cirrus.c | 2 ++ 3 files changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b7c89dff7066..276f6e759bac 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -549,11 +549,15 @@ static hda_nid_t look_for_out_mute_nid(struct hda_codec *codec, static hda_nid_t look_for_out_vol_nid(struct hda_codec *codec, struct nid_path *path) { + struct hda_gen_spec *spec = codec->spec; int i; for (i = path->depth - 1; i >= 0; i--) { - if (nid_has_volume(codec, path->path[i], HDA_OUTPUT)) - return path->path[i]; + hda_nid_t nid = path->path[i]; + if ((spec->out_vol_mask >> nid) & 1) + continue; + if (nid_has_volume(codec, nid, HDA_OUTPUT)) + return nid; } return 0; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 48d44026705b..7e45cb44d151 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -242,6 +242,9 @@ struct hda_gen_spec { /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; + /* bitmask for skipping volume controls */ + u64 out_vol_mask; + /* badness tables for output path evaluations */ const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 18d972501585..38d073ed4bcc 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -660,6 +660,8 @@ static int patch_cs4208(struct hda_codec *codec) return -ENOMEM; spec->gen.automute_hook = cs_automute; + /* exclude NID 0x10 (HP) from output volumes due to different steps */ + spec->gen.out_vol_mask = 1ULL << 0x10; snd_hda_pick_fixup(codec, cs4208_models, cs4208_fixup_tbl, cs4208_fixups); -- 1.8.4.2

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