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[PATCH] ASoC: tegra: Constify static snd_soc_dai_ops structs
by Rikard Falkeborn 27 Sep '21

27 Sep '21

The only usage of these is to assign their address to the ops field in the snd_soc_dai_driver struct, which is a pointer to const. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/tegra/tegra210_adx.c | 4 ++-- sound/soc/tegra/tegra210_amx.c | 4 ++-- sound/soc/tegra/tegra210_mixer.c | 4 ++-- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 4 ++-- 5 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index 78d660447bb1..d7c7849c2f92 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -206,12 +206,12 @@ static int tegra210_adx_put_byte_map(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_soc_dai_ops tegra210_adx_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_adx_in_dai_ops = { .hw_params = tegra210_adx_in_hw_params, .startup = tegra210_adx_startup, }; -static struct snd_soc_dai_ops tegra210_adx_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_adx_out_dai_ops = { .hw_params = tegra210_adx_out_hw_params, }; diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index 83176e1663ab..af9bddfc3120 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -241,12 +241,12 @@ static int tegra210_amx_put_byte_map(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_soc_dai_ops tegra210_amx_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_amx_out_dai_ops = { .hw_params = tegra210_amx_out_hw_params, .startup = tegra210_amx_startup, }; -static struct snd_soc_dai_ops tegra210_amx_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_amx_in_dai_ops = { .hw_params = tegra210_amx_in_hw_params, }; diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 53fcd8f6605a..55e61776c565 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -283,11 +283,11 @@ static int tegra210_mixer_out_hw_params(struct snd_pcm_substream *substream, dai->id - TEGRA210_MIXER_RX_MAX); } -static struct snd_soc_dai_ops tegra210_mixer_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_mixer_out_dai_ops = { .hw_params = tegra210_mixer_out_hw_params, }; -static struct snd_soc_dai_ops tegra210_mixer_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_mixer_in_dai_ops = { .hw_params = tegra210_mixer_in_hw_params, }; diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 3646ce9b0fd1..7b9c7006e419 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -387,7 +387,7 @@ static int tegra210_mvc_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops tegra210_mvc_dai_ops = { +static const struct snd_soc_dai_ops tegra210_mvc_dai_ops = { .hw_params = tegra210_mvc_hw_params, }; diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index 260dca6f6d25..dc477ee1b82c 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3287,12 +3287,12 @@ static int tegra210_sfc_put_control(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_soc_dai_ops tegra210_sfc_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_sfc_in_dai_ops = { .hw_params = tegra210_sfc_in_hw_params, .startup = tegra210_sfc_startup, }; -static struct snd_soc_dai_ops tegra210_sfc_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_sfc_out_dai_ops = { .hw_params = tegra210_sfc_out_hw_params, }; -- 2.33.0

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[PATCH] ASoC: ak4458: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4458 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4458.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 29eb78702bf3..baa9ff5d0ce5 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -464,14 +464,14 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int ret; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: /* Slave Mode */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: /* Consumer Mode */ break; - case SND_SOC_DAIFMT_CBM_CFM: /* Master Mode is not supported */ - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFP: /* Provider Mode is not supported */ + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: default: - dev_err(component->dev, "Master mode unsupported\n"); + dev_err(component->dev, "Clock provider mode unsupported\n"); return -EINVAL; } -- 2.20.1

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[PATCH] ASoC: cs4341: Add SPI device ID table
by Mark Brown 27 Sep '21

27 Sep '21

Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: patches(a)opensource.cirrus.com --- sound/soc/codecs/cs4341.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 7d3e54d8eef3..29d05e32d341 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -305,12 +305,19 @@ static int cs4341_spi_probe(struct spi_device *spi) return cs4341_probe(&spi->dev); } +static const struct spi_device_id cs4341_spi_ids[] = { + { "cs4341a" }, + { } +}; +MODULE_DEVICE_TABLE(spi, cs4341_spi_ids); + static struct spi_driver cs4341_spi_driver = { .driver = { .name = "cs4341-spi", .of_match_table = of_match_ptr(cs4341_dt_ids), }, .probe = cs4341_spi_probe, + .id_table = cs4341_spi_ids, }; #endif -- 2.20.1

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[PATCH v2] ASoC: dt-bindings: wlf,wm8978: Fix I2C address in example
by Geert Uytterhoeven 27 Sep '21

27 Sep '21

According to the WM8978 datasheet, the address of the device is fixed as 0011010 in 2-wire mode. Fixes: 5bd5699c494f42a3 ("ASoC: dt-bindings: Add WM8978 Binding") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- v2: - Fix silly typo (missing 0x prefix). --- Documentation/devicetree/bindings/sound/wlf,wm8978.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml index 944e5859c962e634..96cf9fc9c8b0098b 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -48,10 +48,10 @@ examples: #address-cells = <1>; #size-cells = <0>; - codec@0 { + codec@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8978"; - reg = <0>; + reg = <0x1a>; }; }; -- 2.25.1

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[PATCH v2] ASoC: cpcap: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cpcap driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cpcap.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 05bbacd0d174..598e09024e23 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1168,15 +1168,15 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * "HiFi Playback" should always be configured as - * SND_SOC_DAIFMT_CBM_CFM - codec clk & frm master + * SND_SOC_DAIFMT_CBP_CFP - codec clk & frm provider * SND_SOC_DAIFMT_I2S - I2S mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val &= ~BIT(CPCAP_BIT_SMB_ST_DAC); break; default: - dev_err(dev, "HiFi dai fmt failed: CPCAP should be master"); + dev_err(dev, "HiFi dai fmt failed: CPCAP should be provider"); return -EINVAL; } @@ -1318,15 +1318,15 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * "Voice Playback" and "Voice Capture" should always be - * configured as SND_SOC_DAIFMT_CBM_CFM - codec clk & frm - * master + * configured as SND_SOC_DAIFMT_CBP_CFP - codec clk & frm + * provider */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val &= ~BIT(CPCAP_BIT_SMB_CDC); break; default: - dev_err(component->dev, "Voice dai fmt failed: CPCAP should be the master"); + dev_err(component->dev, "Voice dai fmt failed: CPCAP should be the provider"); val &= ~BIT(CPCAP_BIT_SMB_CDC); break; } -- 2.20.1

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[PATCH] ASoC: pcm179x: Add missing entries SPI to device ID table
by Mark Brown 27 Sep '21

27 Sep '21

Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm179x-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/pcm179x-spi.c b/sound/soc/codecs/pcm179x-spi.c index 0a542924ec5f..ebf63ea90a1c 100644 --- a/sound/soc/codecs/pcm179x-spi.c +++ b/sound/soc/codecs/pcm179x-spi.c @@ -36,6 +36,7 @@ static const struct of_device_id pcm179x_of_match[] = { MODULE_DEVICE_TABLE(of, pcm179x_of_match); static const struct spi_device_id pcm179x_spi_ids[] = { + { "pcm1792a", 0 }, { "pcm179x", 0 }, { }, }; -- 2.20.1

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[PATCH] ASoC: ad1836: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ad1836 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ad1836.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 08a5651bed9f..29e1689da67f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -148,9 +148,9 @@ static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - /* ALCLK,ABCLK are both output, AD1836 can only be master */ - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + /* ALCLK,ABCLK are both output, AD1836 can only be provider */ + case SND_SOC_DAIFMT_CBP_CFP: break; default: return -EINVAL; -- 2.20.1

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[PATCH] ASoC: ak4642: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4642 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4642.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index c49c58eeb476..c284dcc5af76 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -392,13 +392,13 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) data = MCKO | PMPLL; /* use MCKO */ bcko = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set clocking for audio interface */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: data |= MS; bcko = BCKO_64; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

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[PATCH 1/6] ASoC: adau1372: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the adau1372 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau1372.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau1372.c b/sound/soc/codecs/adau1372.c index 6811a8b3866d..1faa4c426365 100644 --- a/sound/soc/codecs/adau1372.c +++ b/sound/soc/codecs/adau1372.c @@ -30,7 +30,7 @@ struct adau1372 { void (*switch_mode)(struct device *dev); bool use_pll; bool enabled; - bool master; + bool clock_provider; struct snd_pcm_hw_constraint_list rate_constraints; unsigned int slot_width; @@ -578,13 +578,13 @@ static int adau1372_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) unsigned int sai0 = 0, sai1 = 0; bool invert_lrclk = false; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - adau1372->master = true; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + adau1372->clock_provider = true; sai1 |= ADAU1372_SAI1_MS; break; - case SND_SOC_DAIFMT_CBS_CFS: - adau1372->master = false; + case SND_SOC_DAIFMT_CBC_CFC: + adau1372->clock_provider = false; break; default: return -EINVAL; @@ -714,7 +714,7 @@ static int adau1372_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 4: sai0 = ADAU1372_SAI0_SAI_TDM4; - if (adau1372->master) + if (adau1372->clock_provider) adau1372->rate_constraints.mask = ADAU1372_RATE_MASK_TDM4_MASTER; else adau1372->rate_constraints.mask = ADAU1372_RATE_MASK_TDM4; -- 2.20.1

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[PATCH] ASoC: dmaengine: Introduce module option prealloc_buffer_size_kbytes
by Sugar Zhang 27 Sep '21

27 Sep '21

Currently, The fixed 512KB prealloc buffer size is too larger for tiny memory kernel (such as 16MB memory). This patch adds the module option "prealloc_buffer_size_kbytes" to specify prealloc buffer size. It's suitable for cards which use the generic dmaengine pcm driver with no config. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/sound/alsa-configuration.rst | 9 +++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +++++- 2 files changed, 14 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 65f6169..34888d4 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -100,6 +100,15 @@ amidi_map MIDI device number maps assigned to the 2st OSS device; Default: 1 +Module snd-soc-core +------------------- + +The soc core module. It is used by all ALSA card drivers. +It takes the following options which have global effects. + +prealloc_buffer_size_kbytes + Specify prealloc buffer size in kbytes (default: 512). + Common parameters for top sound card modules -------------------------------------------- diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 4aa48c7..c54c8ca 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -15,6 +15,10 @@ #include <sound/dmaengine_pcm.h> +static unsigned int prealloc_buffer_size_kbytes = 512; +module_param(prealloc_buffer_size_kbytes, uint, 0444); +MODULE_PARM_DESC(prealloc_buffer_size_kbytes, "Preallocate DMA buffer size (KB)."); + /* * The platforms dmaengine driver does not support reporting the amount of * bytes that are still left to transfer. @@ -237,7 +241,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, prealloc_buffer_size = config->prealloc_buffer_size; max_buffer_size = config->pcm_hardware->buffer_bytes_max; } else { - prealloc_buffer_size = 512 * 1024; + prealloc_buffer_size = prealloc_buffer_size_kbytes * 1024; max_buffer_size = SIZE_MAX; } -- 2.7.4

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