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

[PATCH 0/2] Refactor Audio Graph schema
by Sameer Pujar 12 Nov '20

12 Nov '20

There can be custom sound cards reusing most of the audio-graph implementation. To allow this refactor the audio-graph-card schema into following files. * audio-graph.yaml : defines all common bindings * audio-graph-card.yaml : define compatible property Custom sound cards can just reference 'audio-graph.yaml' and define its own compatible and specific properties. This depends on following series: http://patchwork.ozlabs.org/project/devicetree-bindings/patch/2020110220365… Sameer Pujar (2): ASoC: audio-graph-card: Refactor schema ASoC: renesas,rsnd: Update audio graph references .../bindings/sound/audio-graph-card.yaml | 102 +------------------- .../devicetree/bindings/sound/audio-graph.yaml | 107 +++++++++++++++++++++ .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 +- 3 files changed, 112 insertions(+), 103 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph.yaml -- 2.7.4

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[PATCH v2 0/2] ASoC: Fix error handling in arizona
by Zhang Qilong 12 Nov '20

12 Nov '20

The first patch fixed a wrong free in wm8997_probe. The other fixed PM disable depth imbalance on error handling. Change Log v2: - add collected review and ack. Zhang Qilong (2): ASoC: arizona: Fix a wrong free in wm8997_probe ASoC: arizona: Fix PM disable depth imbalance on error sound/soc/codecs/wm8997.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) -- 2.25.4

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[PATCH] ASoC: pcm512x: Add support for data formats RJ and LJ
by Kirill Marinushkin 12 Nov '20

12 Nov '20

Currently, pcm512x driver supports only I2S data format. This commit adds RJ and LJ as well. I don't expect regression WRT existing sound cards, because: * default value in corresponding register of pcm512x codec is 0 == I2S * existing in-tree sound cards with pcm512x codec are configured for I2S * i don't see how existing off-tree sound cards with pcm512x codec could be configured differently - it would not work * tested explicitly, that there is no regression with Raspberry Pi + sound card `sound/soc/bcm/hifiberry_dacplus.c` Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Matthias Reichl <hias(a)horus.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/codecs/pcm512x.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8153d3d01654..c6e975fb4a43 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1167,6 +1167,7 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); int alen; + int afmt; int gpio; int clock_output; int master_mode; @@ -1195,6 +1196,22 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + switch (pcm512x->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + afmt = PCM512x_AFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + afmt = PCM512x_AFMT_RTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + afmt = PCM512x_AFMT_LTJ; + break; + default: + dev_err(component->dev, "unsupported DAI format: 0x%x\n", + pcm512x->fmt); + return -EINVAL; + } + switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: ret = regmap_update_bits(pcm512x->regmap, @@ -1236,6 +1253,13 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, return ret; } + ret = regmap_update_bits(pcm512x->regmap, PCM512x_I2S_1, + PCM512x_AFMT, afmt); + if (ret != 0) { + dev_err(component->dev, "Failed to set data format: %d\n", ret); + return ret; + } + if (pcm512x->pll_out) { ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_A, 0x11); if (ret != 0) { -- 2.13.6

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Alsa Utils + Lib 1.2.2-1ubuntu2, 1.2.2-2.1ubuntu2.1 breaks sound on HP Envy x360 (Mint 20)
by GitHub issues - edited 11 Nov '20

11 Nov '20

alsa-project/alsa-utils issue #71 was edited from kingkode1337: After update, sound no longer works and does not pick up my sound card. ``` Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) HD Audio Controller Subsystem: Hewlett-Packard Company Family 17h (Models 10h-1fh) HD Audio Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel ``` It now shows as a Dummy Device, under the upgrade so I have abstained from doing so. Issue URL : https://github.com/alsa-project/alsa-utils/issues/71 Repository URL: https://github.com/alsa-project/alsa-utils

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Alsa Utils + Lib 1.2.2-1ubuntu2, 1.2.2-2.1ubuntu2.1 breaks sound on HP Envy x360 (Mint 20)
by GitHub issues - opened 11 Nov '20

11 Nov '20

alsa-project/alsa-utils issue #71 was opened from kingkode1337: After update, sound no longer works and does not pick up my sound card. ``` Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) HD Audio Controller Subsystem: Hewlett-Packard Company Family 17h (Models 10h-1fh) HD Audio Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel ``` It now shows as a Dummy Device, under the upgrade so I have abstained from doing so. Issue URL : https://github.com/alsa-project/alsa-utils/issues/71 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: tegra: Don't warn on probe deferral
by Jon Hunter 11 Nov '20

11 Nov '20

Deferred probe is an expected return value for snd_soc_register_card(). Given that the driver deals with it properly, there's no need to output a warning that may potentially confuse users. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/tegra/tegra_alc5632.c | 4 ++-- sound/soc/tegra/tegra_max98090.c | 8 +++----- sound/soc/tegra/tegra_rt5640.c | 8 +++----- sound/soc/tegra/tegra_rt5677.c | 4 ++-- sound/soc/tegra/tegra_sgtl5000.c | 4 ++-- sound/soc/tegra/tegra_wm8753.c | 8 +++----- sound/soc/tegra/tegra_wm8903.c | 8 +++----- sound/soc/tegra/tegra_wm9712.c | 4 ++-- sound/soc/tegra/trimslice.c | 8 +++----- 9 files changed, 23 insertions(+), 33 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 8661877bf4c6..0a0efd24e4b0 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -203,8 +203,8 @@ static int tegra_alc5632_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto err_put_cpu_of_node; } diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 9d8e16473ab9..00c19704057b 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -247,11 +247,9 @@ static int tegra_max98090_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index c73bd23b3d67..9afba37a3b08 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -193,11 +193,9 @@ static int tegra_rt5640_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 7504507dd8b8..d30f8b6deda4 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -268,8 +268,8 @@ static int tegra_rt5677_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto err_put_cpu_of_node; } diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index e1dc8e7d337a..885332170c77 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -154,8 +154,8 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto err_put_cpu_of_node; } diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index fa41fa366daf..efd793886689 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -156,11 +156,9 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index ef6652aaac9b..e4863fa37b0c 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -352,11 +352,9 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "devm_snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 726edfa21a29..4f09a178049d 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -117,8 +117,8 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto codec_unregister; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index baae4cce7fc6..6c1cc3d0ac33 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -144,11 +144,9 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } -- 2.25.1

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[PATCH v2] ASoC: mediatek: mt8192: skip first time data at the beginning of DMIC recording
by Jiaxin Yu 11 Nov '20

11 Nov '20

We can choose to drop away any length of data from the beginning according to project needs. Some projects don't want to throw away any data, because they want to use recorded data to do echo cancellation, so they have to make sure that they are aligned with the reference data as much as possible. Or there are other algorithms in the upper layer to eliminate this noise. Or some projects want to eliminate this noise form the kernel layer. However, the minimum recommended value is 50ms to skip pop noise. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- This seems like something that would apply equally to all DMICs so should be done at a more general level, so we implement it in the DMIC driver. sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +++- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index dc628fb6c1025..31f1431bf5792 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -175,6 +175,7 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 select SND_SOC_MT6359 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C + select SND_SOC_DMIC help This adds ASoC driver for Mediatek MT8192 boards with the MT6359 RT1015 RT5682 audio codec. diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 41d8a8567bb38..b7f42a530d069 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -540,7 +540,9 @@ SND_SOC_DAILINK_DEFS(playback_hdmi, SND_SOC_DAILINK_DEFS(primary_codec, DAILINK_COMP_ARRAY(COMP_CPU("ADDA")), DAILINK_COMP_ARRAY(COMP_CODEC("mt6359-sound", - "mt6359-snd-codec-aif1")), + "mt6359-snd-codec-aif1"), + COMP_CODEC("dmic-codec", + "dmic-hifi")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(primary_codec_ch34, -- 2.18.0

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[PATCH 0/3] Fix PM disable depth imbalance on error
by Zhang Qilong 11 Nov '20

11 Nov '20

The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. This series of patches fixed it. Zhang Qilong (3): ASoC: wm8994: Fix PM disable depth imbalance on error ASoC: wm8997: Fix PM disable depth imbalance on error ASoC: wm8998: Fix PM disable depth imbalance on error sound/soc/codecs/wm8994.c | 6 +++++- sound/soc/codecs/wm8997.c | 9 +++++++-- sound/soc/codecs/wm8998.c | 4 +++- 3 files changed, 15 insertions(+), 4 deletions(-) -- 2.25.4

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Re: 回覆: CLEVO uses the model name and SSID list of Realtek audio CODEC ALC293 (ALC1220)
by Takashi Iwai 11 Nov '20

11 Nov '20

On Wed, 11 Nov 2020 02:28:50 +0100, Pshou wrote: > > Hi Alsa-devel ML: > > Can you help me add some source code to patch_realtek.c? > > Add some CLEVO SSID in the alc293 (alc1220). > > Best regards > Pshou Applied now with minor coding style fixes. Thanks. Takashi > > -----原始郵件----- > 寄件者: Takashi Iwai <tiwai(a)suse.de> > 寄件日期: 2020年11月10日 15:02 > 收件者: Pshou <pshou(a)realtek.com> > 副本: Kailang <kailang(a)realtek.com> > 主旨: Re: CLEVO uses the model name and SSID list of Realtek audio CODEC ALC293 (ALC1220) > > On Tue, 10 Nov 2020 03:03:38 +0100, > Pshou wrote: > > > > Hi Takashi: > > > > Can you help me add some source code to patch_realtek.c? > > > > Add some CLEVO SSID in the alc293 (alc1220). > > > > Best regards > > > > pshou > > Could you just submit to alsa-devel ML and Cc to me? > A patch has to be submitted to the ML for public review. > > > thanks, > > Takashi > > ------Please consider the environment before printing this e-mail. > [2 0001-Add-alc293-and-alc1220-in-clove.patch <application/octet-stream (base64)>] >

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[PATCH] ASoC: fsl_xcvr: fix break condition
by Viorel Suman (OSS) 10 Nov '20

10 Nov '20

From: Viorel Suman <viorel.suman(a)nxp.com> The break condition copied by mistake as same as loop condition in the previous version, but must be the opposite. So fix it. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c055179e6d11..2a28810d0e29 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -247,7 +247,7 @@ static int fsl_xcvr_ai_write(struct fsl_xcvr *xcvr, u8 reg, u32 data, bool phy) regmap_write(xcvr->regmap, FSL_XCVR_PHY_AI_CTRL_TOG, idx); ret = regmap_read_poll_timeout(xcvr->regmap, FSL_XCVR_PHY_AI_CTRL, val, - (val & idx) != ((val & tidx) >> 1), + (val & idx) == ((val & tidx) >> 1), 10, 10000); if (ret) dev_err(dev, "AI timeout: failed to set %s reg 0x%02x=0x%08x\n", -- 2.26.2

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