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

Get cross compilation to work
by GitHub pull_request - opened 06 Mar '20

06 Mar '20

alsa-project/alsa-firmware pull request #1 was opened from matthewbauer: Rebased version of original patch by Ding Xiang Fei found in https://github.com/NixOS/nixpkgs/pull/63866. This version applies in alsa-firmware-1.2.1. Co-authored-by: Ding Xiang Fei <dingxiangfei2009(a)users.noreply.github.com> Request URL : https://github.com/alsa-project/alsa-firmware/pull/1 Patch URL : https://github.com/alsa-project/alsa-firmware/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-firmware

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[PATCH] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_ssm4567 machine
by Radoslaw Biernacki 05 Mar '20

05 Mar '20

From: Ben Zhang <benzh(a)chromium.org> This single fix address two issues on machine with nau88125 + ssm4567: 1) Audio distortion, due to lack of required clock rate on MCLK line 2) Loud audible "pops" on headphones if there is no sysclk during nau8825 playback power up sequence Explanation for: 1) Due to Skylake HW limitation, MCLK pin can only output 24MHz clk rate (it can be only connected to XTAL parent clk). The BCLK pin can be driven by dividers and therefore FW is able to set it to rate required by chosen audio format. According to nau8825 datasheet, 256*FS sysclk gives the best audio quality and the only way to achieve this (taking into account the above limitations) its to regenerate the MCLK from BCLK on nau8825 side by FFL. Without required clk rate, audio is distorted by added harmonics. 2) Currently Skylake does not output MCLK/FS when the back-end DAI op hw_param is called, so we cannot switch to MCLK/FS in hw_param. This patch reduces pop by letting nau8825 keep using its internal VCO clock during widget power up sequence, until SNDRV_PCM_TRIGGER_START when MCLK/FS is available. Once device resumes, the system will only enable power sequence for playback without doing hardware parameter, audio format, and PLL configure. In the mean time, the jack detecion sequence has changed PLL parameters and switched to internal clock. Thus, the playback signal distorted without correct PLL parameters. That is why we need to configure the PLL again in SNDRV_PCM_TRIGGER_RESUME case. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 71 ++++++++++++++----- 1 file changed, 52 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index c99c8b23e509..c5354da49670 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,7 +58,7 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,40 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + int ret = 0; - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + msleep(20); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { @@ -671,10 +682,32 @@ static int skylake_card_late_probe(struct snd_soc_card *card) return hdac_hdmi_jack_port_init(component, &card->dapm); } +static int __maybe_unused skylake_nau8825_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + dev_dbg(codec_dai->dev, "playback_active:%d playback_widget->active:%d codec_dai->rate:%d\n", + codec_dai->playback_active, codec_dai->playback_widget->active, + codec_dai->rate); + + if (codec_dai->playback_active && codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", .owner = THIS_MODULE, + .resume_post = skylake_nau8825_resume_post, .dai_link = skylake_dais, .num_links = ARRAY_SIZE(skylake_dais), .controls = skylake_controls, -- 2.17.1

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[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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