November 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 4.9 01/12] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ba65f4157a7e0..eeb7f4c66d47a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -38,15 +38,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -58,7 +58,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

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[PATCH AUTOSEL 4.14 01/14] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index e69c141d8ed4c..60a6a83aa772c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -38,15 +38,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -58,7 +58,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

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[PATCH AUTOSEL 4.19 01/17] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 7a89b4aad182f..6cae0e4d33710 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

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[PATCH AUTOSEL 5.4 01/25] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ef6f236752867..de3e3a55de4bf 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

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[PATCH AUTOSEL 5.10 01/43] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index c28ebf891cb05..2cbf679f5c74b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

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Dual simulatenous output on single card not recognized
by Eldred HABERT 30 Nov '21

30 Nov '21

Hello, I bought a laptop with integrated sound output (either to speakers or headphones), and a HDMI output. Attached is the output of `aplay -l`. Under both PipeWire and PulseAudio, I only get a single output/sink (by default to the built-in output), but I can get simultaneous output to both: - under PulseAudio, by adding a `load-module module-alsa-sink device=hw:0,7` line to the config; - under PipeWire, by adding a profile to the card to enable both outputs, which correctly provides two outputs that work fine concurrently. I was directed to this mailing list after the following discussion: > It's a kernel ALSA bug if SOF actually must contain enough data for the driver to figure this out (or at least pass that data on to userspace to deal with it) > if [we are] right that hardware itself does not contain the required data, then it can only be fixed in userspace via a card database > which is one thing to do, another would be maybe a heuristic rule that attempts to guess if such cards with analog and digital sub-devices might be independent Please tell me if more details are needed—it would be nice if hardware configurations like mine could be fully supported out of the box ^^ Cheers, ~ ISSOtm

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Channel distortion when converting S16_LE -> S32_LE ('plug' plugin)
by GitHub issues - opened 30 Nov '21

30 Nov '21

alsa-project/alsa-lib issue #195 was opened from dh1dm: When using a Terretec EWX2496 (ICE1712) a sample format conversion is required in some cases as S32_LE is the only supported format. This can be done by using the 'plug' plugin (e.g. 'plughw:0,0'). However, the left audio channel is distorted in a varying intensity, it sounds like a resolution loss in the lower bits. Applications with native S32_LE support (e.g. Audacity) do not show this effect. I'd suppose that the responsible code is in 'alsa-lib/src/pcm/pcm_plug.c', but up to now I could not track down this issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/195 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: meson: use dev_err_probe
by Jerome Brunet 30 Nov '21

30 Nov '21

Use dev_err_probe() helper function to handle probe deferral. It removes the open coded test for -EPROBE_DEFER but more importantly, it sets the deferral reason in debugfs which is great for debugging. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu.c | 17 +++++++---------- sound/soc/meson/axg-fifo.c | 9 +++------ sound/soc/meson/axg-pdm.c | 9 +++------ sound/soc/meson/axg-spdifin.c | 6 ++---- sound/soc/meson/axg-spdifout.c | 6 ++---- sound/soc/meson/axg-tdm-formatter.c | 18 ++++++------------ sound/soc/meson/axg-tdm-interface.c | 9 +++------ sound/soc/meson/meson-card-utils.c | 4 ++-- sound/soc/meson/t9015.c | 8 ++++---- 9 files changed, 32 insertions(+), 54 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index ba15d5762b0b..37036adf14ce 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -219,31 +219,29 @@ static int aiu_clk_get(struct device *dev) aiu->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(aiu->pclk)) { - if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu pclk\n"); + dev_err_probe(dev, PTR_ERR(aiu->pclk), + "Can't get the aiu pclk\n"); return PTR_ERR(aiu->pclk); } aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); if (IS_ERR(aiu->spdif_mclk)) { - if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu spdif master clock\n"); + dev_err_probe(dev, PTR_ERR(aiu->spdif_mclk), + "Can't get the aiu spdif master clock\n"); return PTR_ERR(aiu->spdif_mclk); } ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), &aiu->i2s); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the i2s clocks\n"); + dev_err_probe(dev, ret, "Can't get the i2s clocks\n"); return ret; } ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), &aiu->spdif); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the spdif clocks\n"); + dev_err_probe(dev, ret, "Can't get the spdif clocks\n"); return ret; } @@ -282,8 +280,7 @@ static int aiu_probe(struct platform_device *pdev) ret = device_reset(dev); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Failed to reset device\n"); + dev_err_probe(dev, ret, "Failed to reset device\n"); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b9af2d513e09..8d3184aeba60 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -352,17 +352,14 @@ int axg_fifo_probe(struct platform_device *pdev) fifo->pclk = devm_clk_get(dev, NULL); if (IS_ERR(fifo->pclk)) { - if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %ld\n", - PTR_ERR(fifo->pclk)); + dev_err_probe(dev, PTR_ERR(fifo->pclk), "failed to get pclk\n"); return PTR_ERR(fifo->pclk); } fifo->arb = devm_reset_control_get_exclusive(dev, NULL); if (IS_ERR(fifo->arb)) { - if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) - dev_err(dev, "failed to get arb reset: %ld\n", - PTR_ERR(fifo->arb)); + dev_err_probe(dev, PTR_ERR(fifo->arb), + "failed to get arb reset\n"); return PTR_ERR(fifo->arb); } diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c index bfd37d49a73e..9537fbcf56e0 100644 --- a/sound/soc/meson/axg-pdm.c +++ b/sound/soc/meson/axg-pdm.c @@ -613,24 +613,21 @@ static int axg_pdm_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } priv->dclk = devm_clk_get(dev, "dclk"); if (IS_ERR(priv->dclk)) { ret = PTR_ERR(priv->dclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk\n"); return ret; } priv->sysclk = devm_clk_get(dev, "sysclk"); if (IS_ERR(priv->sysclk)) { ret = PTR_ERR(priv->sysclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk\n"); return ret; } diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d0d09f945b48..e10bc9352667 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -481,16 +481,14 @@ static int axg_spdifin_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } priv->refclk = devm_clk_get(dev, "refclk"); if (IS_ERR(priv->refclk)) { ret = PTR_ERR(priv->refclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk\n"); return ret; } diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c index e769a5ee6e27..179654d76c8f 100644 --- a/sound/soc/meson/axg-spdifout.c +++ b/sound/soc/meson/axg-spdifout.c @@ -424,16 +424,14 @@ static int axg_spdifout_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk)) { ret = PTR_ERR(priv->mclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk\n"); return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index cab7fa2851aa..b8bbeda87a72 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -284,8 +284,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(formatter->pclk)) { ret = PTR_ERR(formatter->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } @@ -293,8 +292,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(formatter->sclk)) { ret = PTR_ERR(formatter->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk\n"); return ret; } @@ -302,8 +300,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(formatter->lrclk)) { ret = PTR_ERR(formatter->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk\n"); return ret; } @@ -311,8 +308,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); if (IS_ERR(formatter->sclk_sel)) { ret = PTR_ERR(formatter->sclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk_sel\n"); return ret; } @@ -320,8 +316,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); if (IS_ERR(formatter->lrclk_sel)) { ret = PTR_ERR(formatter->lrclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk_sel\n"); return ret; } @@ -329,8 +324,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get reset: %d\n", ret); + dev_err_probe(dev, ret, "failed to get reset\n"); return ret; } diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index db077773af7a..8f8b126de158 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -535,8 +535,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(iface->sclk)) { ret = PTR_ERR(iface->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk\n"); return ret; } @@ -544,8 +543,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(iface->lrclk)) { ret = PTR_ERR(iface->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk\n"); return ret; } @@ -561,8 +559,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) if (ret == -ENOENT) { iface->mclk = NULL; } else { - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk\n"); return ret; } } diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 29b0174f4b5c..cc6e16e31f9c 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -86,8 +86,8 @@ int meson_card_parse_dai(struct snd_soc_card *card, ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); + dev_err_probe(card->dev, ret, + "can't parse sound-dai at %pOFn\n", node); return ret; } *dai_of_node = args.np; diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 4c1349dd1e06..7a0a807f494c 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -259,15 +259,15 @@ static int t9015_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { - if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get core clock\n"); + dev_err_probe(dev, PTR_ERR(priv->pclk), + "failed to get core clock\n"); return PTR_ERR(priv->pclk); } priv->avdd = devm_regulator_get(dev, "AVDD"); if (IS_ERR(priv->avdd)) { - if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) - dev_err(dev, "failed to AVDD\n"); + dev_err_probe(dev, PTR_ERR(priv->avdd), + "failed to get AVDD\n"); return PTR_ERR(priv->avdd); } -- 2.34.0

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[PATCH v8 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 30 Nov '21

30 Nov '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 138 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 175 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 70 +++++++ 10 files changed, 457 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

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[PATCH v3] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 29 Nov '21

29 Nov '21

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells', 'num-serializer' and 'port' which were not there in the txt file. Adds 'dmas' and 'dma-names' in the example which were not there in the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v3: - removes maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'port' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 --------- .../bindings/sound/davinci-mcasp-audio.yaml | 178 ++++++++++++++++++ 2 files changed, 178 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..c4d3f56470bf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,178 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: main registers + - description: data port register + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + description: I2S - 0 or DIT - 1 mode + enum: + - 0 + - 1 + + tdm-slots: + maxItems: 1 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 16 + items: + minimum: 0 + maximum: 2 + default: 0 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + maxItems: 1 + + rx-num-evt: + maxItems: 1 + + dismod: + enum: + - 0 + - 2 + - 3 + default: 2 + + sram-size-playback: + maxItems: 1 + + sram-size-capture: + maxItems: 1 + + interrupts: + minItems: 1 + items: + - description: TX FIFO interrupt + - description: RX FIFO interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + description: parent clock for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + description: ratio of AUCLK and FS if applicable + maxItems: 1 + + gpio-controller: true + + "#gpio-cells": + const: 2 + + function-gpios: + maxItems: 1 + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + num-serializer: + maxItems: 1 + + port: + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + - serial-dir + - op-mode + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; -- 2.17.1

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