July 2022 - Alsa-devel - mailman.alsa-project.org

Add correct support for Rane SL-1
by GitHub pull_request - edited 28 Jul '22

28 Jul '22

alsa-project/alsa-ucm-conf pull request #187 was edited from christophehenry: Following [this ticket](https://bugzilla.kernel.org/show_bug.cgi?id=216082) on kernel bugzilla, I'm trying to add corect support for [Rane SL-1](https://cdn.korn.eu/pictures/product/400/040506_3.jpg), an old USB soundcard featuring 2 stereo output (RCA) and 3 stereo input (2 RCA, 1 6.3mm jack). Currently, `alsa-info.sh` does show the correct number of inputs and outputs But neither `pavucontrol` nor Mixxx do. I was told complex hardware should be described via UCM but the syntax is unfortunately very poorly documented so I can get my head around it and get `pavucontrol` to show all the inputs and outputs. Here's what I have so far which, to my disappointment, does not work. If I put only 1 `SectionDevice` section in the file, I can display 1 ouput under `pavucontrol`. With 2 `SectionDevice`, nothing shows. Without any UCM file, `pavucontrol` shows 2 outputs, 1 intputs (instead of 3) and Mixxx shows 2 outputs and not input. [The result of `alsa-info.sh` is available here](https://bugzilla.kernel.org/attachment.cgi?id=301351). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/187 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/187.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Biju Das 28 Jul '22

28 Jul '22

We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..c08ef317e3e3 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_dma; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_dma: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.25.1

1 1

[PATCH v3 0/2] ASoC: atmel: one fix and one cleanup
by Claudiu Beznea 28 Jul '22

28 Jul '22

Hi, The series adds one fix for mchp-spdifrx and one cleanups for mchp-spdifrx and mchp-spdifrx drivers. Thank you, Claudiu Beznea Changes in v3: - changed cover letter title s/few/one, s/cleanups/cleanup - fix compilation error and warnings - keep only patch 1/5 and patch 3/5 from previous version as the rest of them were integrated Changes in v2: - s/tag/tab in the title of patch 2/5 Claudiu Beznea (2): ASoC: mchp-spdifrx: disable end of block interrupt on failures ASoC: mchp-spdiftx: remove references to mchp_i2s_caps sound/soc/atmel/mchp-spdifrx.c | 9 ++++++--- sound/soc/atmel/mchp-spdiftx.c | 8 -------- 2 files changed, 6 insertions(+), 11 deletions(-) -- 2.34.1

2 3

[PATCH] ASoC: dt-bindings: use spi-peripheral-props.yaml
by Krzysztof Kozlowski 28 Jul '22

28 Jul '22

Instead of listing directly properties typical for SPI peripherals, reference the spi-peripheral-props.yaml schema. This allows using all properties typical for SPI-connected devices, even these which device bindings author did not tried yet. Remove the spi-* properties which now come via spi-peripheral-props.yaml schema, except for the cases when device schema adds some constraints like maximum frequency. While changing additionalProperties->unevaluatedProperties, put it in typical place, just before example DTS. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Technically, this depends on [1] merged to SPI tree, if we want to preserve existing behavior of not allowing SPI CPHA and CPOL in each of schemas in this patch. If this patch comes independently via different tree, the SPI CPHA and CPOL will be allowed for brief period of time, before [1] is merged. This will not have negative impact, just DT schema checks will be loosened for that period. [1] https://lore.kernel.org/all/20220722191539.90641-2-krzysztof.kozlowski@lina… --- .../devicetree/bindings/sound/adi,adau1977.yaml | 7 ++++--- Documentation/devicetree/bindings/sound/wlf,wm8731.yaml | 9 +++++---- 2 files changed, 9 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml index b80454ad97da..847b83398d3d 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml @@ -32,8 +32,6 @@ properties: reset-gpios: maxItems: 1 - spi-max-frequency: true - AVDD-supply: description: Analog power support for the device. @@ -52,7 +50,10 @@ required: - compatible - AVDD-supply -additionalProperties: false +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml index e7220e8b49f0..15795f63b5a3 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml @@ -52,10 +52,6 @@ properties: DCVDD-supply: description: Digital core supply regulator for the DCVDD pin. - spi-max-frequency: true - -additionalProperties: false - required: - reg - compatible @@ -64,6 +60,11 @@ required: - DBVDD-supply - DCVDD-supply +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + examples: - | spi { -- 2.34.1

3 2

[PATCH] ASoC: codecs: va-macro: use fsgen as clock
by Srinivas Kandagatla 28 Jul '22

28 Jul '22

VA Macro fsgen clock is supplied to other LPASS Macros using proper clock apis, however the internal user uses the registers directly without clk apis. This approch has race condition where in external users of the clock might cut the clock while VA macro is actively using this. Moving the internal usage to clk apis would provide a proper refcounting and avoid such race conditions. This issue was noticed while headset was pulled out while recording is in progress and shifting record patch to DMIC. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index d18b56e60433..1ea10dc70748 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -199,6 +199,7 @@ struct va_macro { struct clk *mclk; struct clk *macro; struct clk *dcodec; + struct clk *fsgen; struct clk_hw hw; struct lpass_macro *pds; @@ -467,9 +468,9 @@ static int va_macro_mclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - return va_macro_mclk_enable(va, true); + return clk_prepare_enable(va->fsgen); case SND_SOC_DAPM_POST_PMD: - return va_macro_mclk_enable(va, false); + clk_disable_unprepare(va->fsgen); } return 0; @@ -1473,6 +1474,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_clkout; + va->fsgen = clk_hw_get_clk(&va->hw, "fsgen"); + if (IS_ERR(va->fsgen)) { + ret = PTR_ERR(va->fsgen); + goto err_clkout; + } + ret = devm_snd_soc_register_component(dev, &va_macro_component_drv, va_macro_dais, ARRAY_SIZE(va_macro_dais)); -- 2.25.1

3 2

snd_soc_skl - No sound on PopOS
by GitHub issues - opened 27 Jul '22

27 Jul '22

alsa-project/alsa-lib issue #255 was opened from runcros: Hello, I have no sound on skylake nau8825 max98357a here is my alsa-info : http://alsa-project.org/db/?f=c1b90f94c1bb072f507636f3979c0bc8a62ae470 Can you help me please If you need others infos, I am here =) thank you ! Issue URL : https://github.com/alsa-project/alsa-lib/issues/255 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: mediatek: mt8186: remove unnecessary judgments
by Jiaxin Yu 27 Jul '22

27 Jul '22

The afe_priv->dai_priv[] is allocated when platform driver probe(), if it failed, the ASoC platform driver probe() will return fail first. Therefore, this is excessive judgment, and the condition will never be established. Bug report: https://www.spinics.net/lists/alsa-devel/msg145609.html This is a semi-automatic email about new static checker warnings. The patch ae92dcbee8b6: "ASoC: mediatek: mt8186: support tdm in platform driver" from May 23, 2022, leads to the following Smatch complaint: sound/soc/mediatek/mt8186/mt8186-dai-tdm.c:424 mtk_dai_tdm_hw_params() warn: variable dereferenced before check 'tdm_priv' (see line 406) sound/soc/mediatek/mt8186/mt8186-dai-tdm.c 405 struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id]; 406 unsigned int tdm_mode = tdm_priv->tdm_mode; ^^^^^^^^^^^^^^^^^^^ Lot's of dereferences 407 unsigned int data_mode = tdm_priv->data_mode; 408 unsigned int rate = params_rate(params); 409 unsigned int channels = params_channels(params); 410 snd_pcm_format_t format = params_format(params); 411 unsigned int bit_width = 412 snd_pcm_format_physical_width(format); 413 unsigned int tdm_channels = (data_mode == TDM_DATA_ONE_PIN) ? 414 get_tdm_ch_per_sdata(tdm_mode, channels) : 2; 415 unsigned int lrck_width = 416 get_tdm_lrck_width(format, tdm_mode); 417 unsigned int tdm_con = 0; 418 bool slave_mode = tdm_priv->slave_mode; 419 bool lrck_inv = tdm_priv->lck_invert; 420 bool bck_inv = tdm_priv->bck_invert; 421 unsigned int tran_rate; 422 unsigned int tran_relatch_rate; 423 424 if (!tdm_priv) { ^^^^^^^^^ Checked too late 425 dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); 426 return -EINVAL; Fixes: ae92dcbee8b6 ("ASoC: mediatek: mt8186: support tdm in platform driver") Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 5 -- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 63 --------------------- sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 5 -- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 55 +----------------- 4 files changed, 1 insertion(+), 127 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index 6be6d4f3b585..f6f23c7c3ff5 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -655,11 +655,6 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", __func__, id, substream->stream, rate); - if (!adda_priv) { - dev_err(afe->dev, "%s(), adda_priv == NULL", __func__); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { unsigned int dl_src2_con0; unsigned int dl_src2_con1; diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index 5c1290b950e8..ec79e2f2a54d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -119,12 +119,6 @@ static int mt8186_i2s_hd_get(struct snd_kcontrol *kcontrol, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - ucontrol->value.integer.value[0] = i2s_priv->low_jitter_en; return 0; @@ -148,12 +142,6 @@ static int mt8186_i2s_hd_set(struct snd_kcontrol *kcontrol, __func__, kcontrol->id.name, hd_en); i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - if (i2s_priv->low_jitter_en == hd_en) return 0; @@ -377,11 +365,6 @@ static int mtk_i2s_en_event(struct snd_soc_dapm_widget *w, i2s_priv = get_i2s_priv_by_name(afe, w->name); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", __func__, w->name, event); @@ -442,11 +425,6 @@ static int mtk_mclk_en_event(struct snd_soc_dapm_widget *w, i2s_priv = get_i2s_priv_by_name(afe, w->name); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - switch (event) { case SND_SOC_DAPM_PRE_PMU: mt8186_mck_enable(afe, i2s_priv->mclk_id, i2s_priv->mclk_rate); @@ -566,12 +544,6 @@ static int mtk_afe_i2s_share_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (i2s_priv->share_i2s_id < 0) return 0; @@ -587,12 +559,6 @@ static int mtk_afe_i2s_hd_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (get_i2s_id_by_name(afe, sink->name) == get_i2s_id_by_name(afe, source->name)) return i2s_priv->low_jitter_en; @@ -618,15 +584,8 @@ static int mtk_afe_i2s_apll_connect(struct snd_soc_dapm_widget *source, int i2s_need_apll; i2s_priv = get_i2s_priv_by_name(afe, w->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); - /* choose APLL from i2s rate */ i2s_need_apll = mt8186_get_apll_by_rate(afe, i2s_priv->rate); @@ -642,12 +601,6 @@ static int mtk_afe_i2s_mclk_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (get_i2s_id_by_name(afe, sink->name) == get_i2s_id_by_name(afe, source->name)) return (i2s_priv->mclk_rate > 0) ? 1 : 0; @@ -672,12 +625,6 @@ static int mtk_afe_mclk_apll_connect(struct snd_soc_dapm_widget *source, int cur_apll; i2s_priv = get_i2s_priv_by_name(afe, w->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); @@ -980,11 +927,6 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, dev_dbg(afe->dev, "%s(), id %d, rate %d, format %d\n", __func__, i2s_id, rate, format); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - i2s_priv->rate = rate; switch (i2s_id) { @@ -1053,11 +995,6 @@ static int mtk_dai_i2s_set_sysclk(struct snd_soc_dai *dai, int apll; int apll_rate; - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - if (dir != SND_SOC_CLOCK_OUT) { dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); return -EINVAL; diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c index 0b0032ecfe6d..41221a66111c 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c @@ -287,11 +287,6 @@ static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_pcm_priv *pcm_priv = afe_priv->dai_priv[dai->id]; - if (!pcm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - /* DAI mode*/ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c index c6ead7c252f0..4148dceb3a4c 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c @@ -106,11 +106,6 @@ static int mtk_tdm_en_event(struct snd_soc_dapm_widget *w, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", __func__, w->name, event); @@ -138,11 +133,6 @@ static int mtk_tdm_mck_en_event(struct snd_soc_dapm_widget *w, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x, dai_id %d\n", __func__, w->name, event, dai_id); @@ -215,11 +205,6 @@ static int mtk_afe_tdm_mclk_connect(struct snd_soc_dapm_widget *source, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - return (tdm_priv->mclk_rate > 0) ? 1 : 0; } @@ -250,11 +235,6 @@ static int mtk_afe_tdm_hd_connect(struct snd_soc_dapm_widget *source, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - return tdm_priv->low_jitter_en; } @@ -270,11 +250,6 @@ static int mtk_afe_tdm_apll_connect(struct snd_soc_dapm_widget *source, int cur_apll; int tdm_need_apll; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); @@ -303,11 +278,6 @@ static int mt8186_tdm_hd_get(struct snd_kcontrol *kcontrol, int dai_id = get_tdm_id_by_name(kcontrol->id.name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - ucontrol->value.integer.value[0] = tdm_priv->low_jitter_en; return 0; @@ -332,11 +302,6 @@ static int mt8186_tdm_hd_set(struct snd_kcontrol *kcontrol, dev_dbg(afe->dev, "%s(), kcontrol name %s, hd_en %d\n", __func__, kcontrol->id.name, hd_en); - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - if (tdm_priv->low_jitter_en == hd_en) return 0; @@ -421,22 +386,14 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, unsigned int tran_rate; unsigned int tran_relatch_rate; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - tdm_priv->rate = rate; - tran_rate = mt8186_rate_transform(afe->dev, rate, dai->id); tran_relatch_rate = mt8186_tdm_relatch_rate_transform(afe->dev, rate); /* calculate mclk_rate, if not set explicitly */ if (!tdm_priv->mclk_rate) { tdm_priv->mclk_rate = rate * tdm_priv->mclk_multiple; - mtk_dai_tdm_cal_mclk(afe, - tdm_priv, - tdm_priv->mclk_rate); + mtk_dai_tdm_cal_mclk(afe, tdm_priv, tdm_priv->mclk_rate); } /* ETDM_IN1_CON0 */ @@ -508,11 +465,6 @@ static int mtk_dai_tdm_set_sysclk(struct snd_soc_dai *dai, struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - if (dir != SND_SOC_CLOCK_IN) { dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); return -EINVAL; @@ -529,11 +481,6 @@ static int mtk_dai_tdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - /* DAI mode*/ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: -- 2.25.1

2 1

[PATCH] ASoC: amd: acp: uninitialized variable in i2s_irq_handler()
by Dan Carpenter 27 Jul '22

27 Jul '22

The "ext_intr_stat1" is used without being initialized. Fixes: 93f53881473c ("ASoC: amd: acp: Modify local variables name to generic") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp/acp-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 10730d33c3b0..f561d39b33e2 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -100,7 +100,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) return IRQ_NONE; if (adata->rsrc->no_of_ctrls == 2) - ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); + ext_intr_stat1 = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); -- 2.35.1

2 1

[PATCH 0/2] Add support for CLSA0101
by Lucas Tanure 27 Jul '22

27 Jul '22

Add Support for the CLSA0101 laptop, an Intel version of CLSA0100. This patch has been tested using the CLSA0100, ensuring it doesn't break the sound for it. We appreciate it if someone with CLSA0101 could verify that this the patch works for them. Lucas Tanure (2): ALSA: hda: cs35l41: Use the CS35L41 HDA internal define ALSA: hda: cs35l41: Support CLSA0101 sound/pci/hda/cs35l41_hda.c | 67 +++++++++++++++++++++-------------- sound/pci/hda/patch_realtek.c | 12 +++++++ 2 files changed, 53 insertions(+), 26 deletions(-) -- 2.37.1

3 6

[PATCH 0/4] ALSA: Defer async signal handling
by Takashi Iwai 27 Jul '22

27 Jul '22

Hi, this series is another attempt to address the potential deadlocks via kill_fasync() calls that have been reported by syzbot for long time. Instead of the previous series to drop the async handler[*], this tries to defer the kill_fasync() call. A few new common helpers are introduced at first, then the actual usages are replaced in the following patches. The patches passed the quick tests with alsa-lib test cases. Takashi [*] https://lore.kernel.org/r/20220717070549.5993-1-tiwai@suse.de === Takashi Iwai (4): ALSA: core: Add async signal helpers ALSA: timer: Use deferred fasync helper ALSA: pcm: Use deferred fasync helper ALSA: control: Use deferred fasync helper include/sound/control.h | 2 +- include/sound/core.h | 8 ++++ include/sound/pcm.h | 2 +- sound/core/control.c | 7 ++-- sound/core/misc.c | 93 +++++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/timer.c | 11 ++--- 9 files changed, 116 insertions(+), 12 deletions(-) -- 2.35.3

2 6