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

[PATCH v4 0/2] Add power domains support for digital macro codecs
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

This patch set is to add power domains support for RX, TX and VA macros. Changes since v3: -- Add dt-bindings support. Changes since v2: -- Remove redundant local variable. -- Update pds error handling sequence. -- Update module description. -- Clean up pds init function. -- Remove redundant arguments. Changes since v1: -- Add missing macros in Kconfig. Srinivasa Rao Mandadapu (2): ASoC: codecs: Add power domains support in digital macro codecs ASoC: qcom: dt-bindings: Add bindings for power domains in lpass digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-tx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-va-macro.yaml | 8 +++ sound/soc/codecs/Kconfig | 7 +++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 67 ++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 17 ++++++ sound/soc/codecs/lpass-rx-macro.c | 13 ++++- sound/soc/codecs/lpass-tx-macro.c | 10 ++++ sound/soc/codecs/lpass-va-macro.c | 11 +++- sound/soc/qcom/Kconfig | 1 + 11 files changed, 150 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h -- 2.7.4

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[PATCH] ASoC: fsl: Drop unused argument from imx_pcm_dma_init()
by Sascha Hauer 28 Feb '22

28 Feb '22

Since 70d435ba1cd ("ASoC: imx-pcm-dma: simplify pcm_config") the size argument to imx_pcm_dma_init() is unused, so drop it. Also remove the now unused defines that the users of imx_pcm_dma_init() used to pass the size argument Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 +- sound/soc/fsl/imx-pcm.h | 7 ++----- 7 files changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 99ab7f0241cf3..422922146f2a5 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -241,7 +241,7 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev, IMX_DEFAULT_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3a9e2df4e16f6..ed444e8f1d6bf 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1077,7 +1077,7 @@ static int fsl_esai_probe(struct platform_device *pdev) * Register platform component before registering cpu dai for there * is not defer probe for platform component in snd_soc_add_pcm_runtime(). */ - ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) { dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); goto err_pm_get_sync; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 10544fa27dc0b..d7b990c6429ab 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1157,7 +1157,7 @@ static int fsl_sai_probe(struct platform_device *pdev) * is not defer probe for platform component in snd_soc_add_pcm_runtime(). */ if (sai->soc_data->use_imx_pcm) { - ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) goto err_pm_get_sync; } else { diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index d178b479c8bd4..57c41b2f7d177 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1522,7 +1522,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) * Register platform component before registering cpu dai for there * is not defer probe for platform component in snd_soc_add_pcm_runtime(). */ - ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) { dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); goto err_pm_disable; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1169d1104b9e8..ca30a4ede076e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1372,7 +1372,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, if (ret) goto error_pcm; } else { - ret = imx_pcm_dma_init(pdev, IMX_SSI_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) goto error_pcm; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 04a9bc749016d..14e94270911cf 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -34,7 +34,7 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { .compat_filter_fn = filter, }; -int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +int imx_pcm_dma_init(struct platform_device *pdev) { struct snd_dmaengine_pcm_config *config; diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 5dd406774d3ea..5c6cf1ca8c8ab 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -17,9 +17,6 @@ #define IMX_SSI_DMABUF_SIZE (64 * 1024) #define IMX_DEFAULT_DMABUF_SIZE (64 * 1024) -#define IMX_SAI_DMABUF_SIZE (64 * 1024) -#define IMX_SPDIF_DMABUF_SIZE (64 * 1024) -#define IMX_ESAI_DMABUF_SIZE (256 * 1024) static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, @@ -40,9 +37,9 @@ struct imx_pcm_fiq_params { }; #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) -int imx_pcm_dma_init(struct platform_device *pdev, size_t size); +int imx_pcm_dma_init(struct platform_device *pdev); #else -static inline int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +static inline int imx_pcm_dma_init(struct platform_device *pdev) { return -ENODEV; } -- 2.30.2

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[PATCH -next] ASoC: mediatek: mt8195: Remove unnecessary print function dev_err()
by Yang Li 28 Feb '22

28 Feb '22

The print function dev_err() is redundant because platform_get_irq() already prints an error. Eliminate the follow coccicheck warning: ./sound/soc/mediatek/mt8195/mt8195-afe-pcm.c:3126:2-9: line 3126 is redundant because platform_get_irq() already prints an error Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index b77c2ba5a629..550636500949 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3122,10 +3122,8 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); - if (irq_id < 0) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + if (irq_id < 0) return -ENXIO; - } ret = devm_request_irq(dev, irq_id, mt8195_afe_irq_handler, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); -- 2.20.1.7.g153144c

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[PATCH 1/4] ASoC: pcm1681: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the pcm1681 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm1681.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 5b78e9299c95..9eb65f94fc4d 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -136,8 +136,8 @@ static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; struct pcm1681_private *priv = snd_soc_component_get_drvdata(component); - /* The PCM1681 can only be slave to all clocks */ - if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + /* The PCM1681 can only be consumer to all clocks */ + if ((format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { dev_err(component->dev, "Invalid clocking mode\n"); return -EINVAL; } -- 2.30.2

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[PATCH] ASoC: pcm3060: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the pcm3060 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 14 +++++++------- sound/soc/codecs/pcm3060.h | 2 +- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index b2358069cf9b..4e3bfb9fa444 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -68,15 +68,15 @@ static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - priv->dai[dai->id].is_master = true; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + priv->dai[dai->id].is_provider = true; break; - case SND_SOC_DAIFMT_CBS_CFS: - priv->dai[dai->id].is_master = false; + case SND_SOC_DAIFMT_CBC_CFC: + priv->dai[dai->id].is_provider = false; break; default: - dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt); + dev_err(comp->dev, "unsupported DAI mode: 0x%x\n", fmt); return -EINVAL; } @@ -116,7 +116,7 @@ static int pcm3060_hw_params(struct snd_pcm_substream *substream, unsigned int reg; unsigned int val; - if (!priv->dai[dai->id].is_master) { + if (!priv->dai[dai->id].is_provider) { val = PCM3060_REG_MS_S; goto val_ready; } diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index 18d51e5dac2c..5e1185e7b03d 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -23,7 +23,7 @@ extern const struct regmap_config pcm3060_regmap; #define PCM3060_CLK2 2 struct pcm3060_priv_dai { - bool is_master; + bool is_provider; unsigned int sclk_freq; }; -- 2.30.2

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[PATCH] ASoC: wl1273: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the wl1273 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wl1273.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index d8ced4559bf2..02232f64110e 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -414,13 +414,13 @@ int wl1273_get_format(struct snd_soc_component *component, unsigned int *fmt) case WL1273_MODE_FM_TX: *fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; break; case WL1273_MODE_BT: *fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; break; default: -- 2.30.2

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[PATCH] ASoC: uda1380: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the uda1380 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/uda1380.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 89f2bfeeb70e..13060a9a2388 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -435,8 +435,8 @@ static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, iface |= R01_SFORI_MSB | R01_SFORO_MSB; } - /* DATAI is slave only, so in single-link mode, this has to be slave */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + /* DATAI is consumer only */ + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; uda1380_write_reg_cache(component, UDA1380_IFACE, iface); @@ -465,8 +465,8 @@ static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai, iface |= R01_SFORI_MSB; } - /* DATAI is slave only, so this has to be slave */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + /* DATAI is consumer only */ + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; uda1380_write(component, UDA1380_IFACE, iface); @@ -495,7 +495,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, iface |= R01_SFORO_MSB; } - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == SND_SOC_DAIFMT_CBP_CFP) iface |= R01_SIM; uda1380_write(component, UDA1380_IFACE, iface); -- 2.30.2

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[PATCH] ASoC: uda134x: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the uda134x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/uda134x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index bf9182cedb82..037833c509f7 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -272,9 +272,9 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, pr_debug("%s fmt: %08X\n", __func__, fmt); - /* codec supports only full slave mode */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { - printk(KERN_ERR "%s unsupported slave mode\n", __func__); + /* codec supports only full consumer mode */ + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { + printk(KERN_ERR "%s unsupported clocking mode\n", __func__); return -EINVAL; } -- 2.30.2

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[PATCH] ASoC: uda1334: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the uda1334 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/uda1334.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/uda1334.c b/sound/soc/codecs/uda1334.c index 21ab8c5487ba..8670a2a05a56 100644 --- a/sound/soc/codecs/uda1334.c +++ b/sound/soc/codecs/uda1334.c @@ -172,7 +172,7 @@ static int uda1334_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) SND_SOC_DAIFMT_MASTER_MASK); if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS)) { + SND_SOC_DAIFMT_CBC_CFC)) { dev_err(codec_dai->dev, "Invalid DAI format\n"); return -EINVAL; } -- 2.30.2

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[PATCH 1/2] ASoC: tscs42xx: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the tscs42xx driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tscs42xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index fb861baf50e8..5b63e017a43b 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1197,9 +1197,9 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; int ret; - /* Slave mode not supported since it needs always-on frame clock */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* Consumer mode not supported since it needs always-on frame clock */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { -- 2.30.2

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