March 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek - Enable the headset of ASUS B9450FA with ALC294
by Jian-Hong Pan 05 Mar '20

05 Mar '20

A headset on the laptop like ASUS B9450FA does not work, until quirk ALC294_FIXUP_ASUS_HPE is applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Singed-off-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 477589e7ec1d..a47f6404aea9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5920,7 +5920,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - + ALC294_FIXUP_ASUS_HPE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7040,7 +7040,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - + [ALC294_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7204,6 +7214,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.25.1

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Re: [PATCH 1/1] ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master
by Takashi Iwai 05 Mar '20

05 Mar '20

On Sun, 23 Feb 2020 10:24:16 +0100, Christian Lachner wrote: > > The Gigabyte X570 Aorus Master motherboard with ALC1220 codec > requires a similar workaround for Clevo laptops to enforce the > DAC/mixer connection path. Set up a quirk entry for that. > > BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 > Signed-off-by: Christian Lachner <gladiac(a)gmail.com> Applied now. Thanks. Takashi

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Enable Headset Button for ThinkPad X1
by Kailang 05 Mar '20

05 Mar '20

Hi Takashi, Lenovo ThinkPad X1 want to support Headset Button. Attach patch will enable it. BR, Kailang

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RE: Enable Headset Mic for Dell desktop
by Kailang 05 Mar '20

05 Mar '20

Sorry! Forgot attach file. > -----Original Message----- > From: Kailang > Sent: Thursday, February 20, 2020 4:30 PM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: Enable Headset Mic for Dell desktop > > Hi Takashi, > > Attach patch was supported headset Mic. > > BR, > Kailang

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snd_hda_intel.power_save=1 causes pops/clicks on Lenovo T470s headphone jack?
by Hans de Goede 05 Mar '20

05 Mar '20

Hi, Because of a bug-report about power-saving related plops/clocks on a Lenovo T470s, I've asked inside Red Hat if people with a T470s and running a recent kernel were also experiencing this. Most people are happy with the audio, but I did get a few bug reports about plops on the headphones-jack. Is this a know issue? and/or is there anything we can do about this? Regards, Hans

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[PATCH for-next] ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification
by Dan Murphy 05 Mar '20

05 Mar '20

Fix the range verification check for the mic_bias and vref device tree entries. Fixes 37bde5acf040 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 93a0cb8e662c..38897568ee96 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -748,9 +748,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) bias_source = ADCX140_MIC_BIAS_VAL_VREF; - if (bias_source != ADCX140_MIC_BIAS_VAL_VREF && - bias_source != ADCX140_MIC_BIAS_VAL_VREF_1096 && - bias_source != ADCX140_MIC_BIAS_VAL_AVDD) { + if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || + bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } @@ -760,9 +759,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) vref_source = ADCX140_MIC_BIAS_VREF_275V; - if (vref_source != ADCX140_MIC_BIAS_VREF_275V && - vref_source != ADCX140_MIC_BIAS_VREF_25V && - vref_source != ADCX140_MIC_BIAS_VREF_1375V) { + if (vref_source < ADCX140_MIC_BIAS_VREF_275V || + vref_source > ADCX140_MIC_BIAS_VREF_1375V) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } -- 2.25.1

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[PATCH] ASoC: sprd: Allow the MCDT driver to build into modules
by Baolin Wang 05 Mar '20

05 Mar '20

Change the config to 'tristate' for MCDT driver to allow it to build into modules, as well as changing to use IS_ENABLED() to validate if need supply dummy functions when building the MCDT driver as a module. Signed-off-by: Baolin Wang <baolin.wang7(a)gmail.com> --- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 5474fd3..5e0ac82 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -8,7 +8,7 @@ config SND_SOC_SPRD the Spreadtrum SoCs' Audio interfaces. config SND_SOC_SPRD_MCDT - bool "Spreadtrum multi-channel data transfer support" + tristate "Spreadtrum multi-channel data transfer support" depends on SND_SOC_SPRD help Say y here to enable multi-channel data transfer support. It diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h index 9cc7e207..679e3af 100644 --- a/sound/soc/sprd/sprd-mcdt.h +++ b/sound/soc/sprd/sprd-mcdt.h @@ -48,7 +48,7 @@ struct sprd_mcdt_chan { struct list_head list; }; -#ifdef CONFIG_SND_SOC_SPRD_MCDT +#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT) struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type); void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan); -- 1.9.1

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Applied "ASoC: soc-pcm: use defined stream" to the asoc tree
by Mark Brown 05 Mar '20

05 Mar '20

The patch ASoC: soc-pcm: use defined stream has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 25c2f5156dd57f03aee2de079248c23a56222c92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 27 Feb 2020 10:54:38 +0900 Subject: [PATCH] ASoC: soc-pcm: use defined stream Many functions defines "stream = substream->stream", but some of them is using "substream->stream" instead of "stream". It is pointless. This patch uses defined stream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87mu947q1t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 90857138c823..8c27eb4d5e4c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -644,8 +644,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * bailed out on a higher level, since there would be no * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(codec_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream); @@ -2149,7 +2148,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - ret = dpcm_be_dai_startup(fe, fe_substream->stream); + ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret); goto be_err; @@ -2180,7 +2179,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) return 0; unwind: - dpcm_be_dai_startup_unwind(fe, fe_substream->stream); + dpcm_be_dai_startup_unwind(fe, stream); be_err: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; @@ -2234,7 +2233,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ - dpcm_be_dai_shutdown(fe, substream->stream); + dpcm_be_dai_shutdown(fe, stream); dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); @@ -2412,9 +2411,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - memcpy(&fe->dpcm[substream->stream].hw_params, params, + memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); - ret = dpcm_be_dai_hw_params(fe, substream->stream); + ret = dpcm_be_dai_hw_params(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; @@ -2736,7 +2735,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; } - ret = dpcm_be_dai_prepare(fe, substream->stream); + ret = dpcm_be_dai_prepare(fe, stream); if (ret < 0) goto out; -- 2.20.1

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Applied "ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths()" to the asoc tree
by Mark Brown 05 Mar '20

05 Mar '20

The patch ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6e02feb0d2663c1b7caa5e271c2a60e219f0ca07 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 27 Feb 2020 10:54:48 +0900 Subject: [PATCH] ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() dpcm_add_paths() checks returned be from dpcm_get_be() static int dpcm_add_paths(...) { ... for_each_dapm_widgets(list, i, widget) { ... be = dpcm_get_be(...); ... /* make sure BE is a real BE */ => if (!be->dai_link->no_pcm) continue; ... } ... } But, dpcm_get_be() itself is checking it already. dpcm_get_be(...) { ... for_each_card_rtds(card, be) { => if (!be->dai_link->no_pcm) continue; ... if (...) => return be; } return NULL } This patch removes duplicate check Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87lfoo7q1j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8c27eb4d5e4c..e3a2c4f7757b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1690,10 +1690,6 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* make sure BE is a real BE */ - if (!be->dai_link->no_pcm) - continue; - /* don't connect if FE is not running */ if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; -- 2.20.1

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sof-hda-dsp: Support systems without integrated graphics audio
by GitHub pull_request - opened 05 Mar '20

05 Mar '20

alsa-project/alsa-ucm-conf pull request #15 was opened from kv2019i: On systems where integrated graphics audio is not present or is disabled, the HDMI PCM nodes are disabled. Add rules to detect these systems by checking presence of HDMI jack controls with UCM2 rules. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/15 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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