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

[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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[PATCH] ASoC: ep93xx: Convert to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of the effort to remove our old APIs based on outdated terminology update the EP93xx drivers to use modern terminology. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 10 +++++----- sound/soc/cirrus/snappercl15.c | 2 +- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 7b6cdc9c8a23..385290202912 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -60,7 +60,7 @@ static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ops = &edb93xx_ops, SND_SOC_DAILINK_REG(hifi), }; diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 0d26550d0df8..06c315a4d20a 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -245,14 +245,14 @@ static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* CPU is master */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + /* CPU is provider */ clk_cfg |= EP93XX_I2S_CLKCFG_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFM: - /* Codec is master */ + case SND_SOC_DAIFMT_CBP_CFP: + /* Codec is provider */ clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER; break; diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index c4b112921661..a286f5beeaeb 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ops = &snappercl15_ops, SND_SOC_DAILINK_REG(aic23), }; -- 2.20.1

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[PATCH] ASoC: ak4671: 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 ak4671 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4671.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index eb435235b5a3..e9d1251c4265 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -520,11 +520,11 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ mode = snd_soc_component_read(component, AK4671_PLL_MODE_SELECT1); - 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: mode |= AK4671_M_S; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: mode &= ~(AK4671_M_S); break; default: -- 2.20.1

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[PATCH] ASoC: ak4118: 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 ak4118 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4118.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c index 5d46ae85566c..e0a6451851e8 100644 --- a/sound/soc/codecs/ak4118.c +++ b/sound/soc/codecs/ak4118.c @@ -151,8 +151,8 @@ static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { }; -static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, - unsigned int format) +static int ak4118_set_dai_fmt_provider(struct ak4118_priv *ak4118, + unsigned int format) { int dif; @@ -173,8 +173,8 @@ static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, return dif; } -static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, - unsigned int format) +static int ak4118_set_dai_fmt_consumer(struct ak4118_priv *ak4118, + unsigned int format) { int dif; @@ -201,14 +201,12 @@ static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, int dif; int ret = 0; - switch (format & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - /* component is master */ - dif = ak4118_set_dai_fmt_master(ak4118, format); + switch (format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + dif = ak4118_set_dai_fmt_provider(ak4118, format); break; - case SND_SOC_DAIFMT_CBS_CFS: - /*component is slave */ - dif = ak4118_set_dai_fmt_slave(ak4118, format); + case SND_SOC_DAIFMT_CBC_CFC: + dif = ak4118_set_dai_fmt_consumer(ak4118, format); break; default: ret = -ENOTSUPP; -- 2.20.1

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[PATCH] ASoC: codecs: Fix spelling mistake "Unsupport" -> "Unsupported"
by Colin King 27 Sep '21

27 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in dev_err error messages. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1015.c | 2 +- sound/soc/codecs/rt1016.c | 2 +- sound/soc/codecs/rt1019.c | 2 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/rt5682.c | 6 +++--- 18 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c0c5952cdff7..6a27dfacd81c 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -864,7 +864,7 @@ static int rt1015_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1016.c b/sound/soc/codecs/rt1016.c index 7561d202274c..9845cdddcb4c 100644 --- a/sound/soc/codecs/rt1016.c +++ b/sound/soc/codecs/rt1016.c @@ -490,7 +490,7 @@ static int rt1016_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out * 4, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index 8c0b00242bb8..80b7ca0e4e1e 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -359,7 +359,7 @@ static int rt1019_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 7a0094578e46..a9c473537a91 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -841,7 +841,7 @@ static int rt1305_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index b4e5546e2e21..c555b77b3c5c 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -664,7 +664,7 @@ static int rt1308_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 4b1ad5054e8d..577680df7052 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -936,7 +936,7 @@ static int rt5514_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index fd0d3a08e9dd..8e6414468a87 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1133,7 +1133,7 @@ static int rt5616_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index cd1db5caabad..d01fe73ab9c8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1909,7 +1909,7 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9408ee63cb26..197c56047947 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2969,7 +2969,7 @@ static int rt5645_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index fc0c83b73f09..e78ea101bc8d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1487,7 +1487,7 @@ static int rt5651_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4a50b169fe03..e1503c2eee81 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3509,7 +3509,7 @@ static int rt5659_set_component_pll(struct snd_soc_component *component, int pll ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 33ff9156358b..3b50fb29864e 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1046,7 +1046,7 @@ static int rt5660_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index be9fc58ff681..0389b2bb360e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2941,7 +2941,7 @@ static int rt5663_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index e59323fd5bf2..33e889802ff8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4374,7 +4374,7 @@ static int rt5665_set_component_pll(struct snd_soc_component *component, int pll ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6ab1a8bc3735..fb09715bf932 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2171,7 +2171,7 @@ static int rt5668_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index ecbaf129a6e3..ce7684752bb0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2577,7 +2577,7 @@ static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f655228c8c4b..4a8c267d4fbc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4557,7 +4557,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rt5677_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 4a64cab99c55..12113c2dcae2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2327,7 +2327,7 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, pll2_fout1 = 3840000; ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } @@ -2339,7 +2339,7 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", + dev_err(component->dev, "Unsupported input clock %d\n", pll2_fout1); return ret; } @@ -2390,7 +2390,7 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } -- 2.32.0

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[PATCH] ASoC: cs42l42: Use two thresholds and increased wait time for manual type detection
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Some headsets require very different comparator thresholds for type detection, as well as longer settling times. In order to detect a larger number of headsets, use 2 thresholds to give maximum coverage (1.25V and 1.75V), as well as a longer settling time of 100ms. This will not affect default audotodetect mode and applies to manual mode type detection only. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 84 +++++++++++++++++++++++++++----------- sound/soc/codecs/cs42l42.h | 5 +++ 2 files changed, 66 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 78f8c89410b6..3f202acd4a8e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1048,7 +1048,8 @@ static struct snd_soc_dai_driver cs42l42_dai = { static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) { unsigned int hs_det_status; - unsigned int hs_det_comp; + unsigned int hs_det_comp1; + unsigned int hs_det_comp2; unsigned int hs_det_sw; /* Set hs detect to manual, active mode */ @@ -1063,23 +1064,40 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) (0 << CS42L42_HSBIAS_REF_SHIFT) | (0 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + /* Configure HS DET comparator reference levels. */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL1, + CS42L42_HSDET_COMP1_LVL_MASK | + CS42L42_HSDET_COMP2_LVL_MASK, + (CS42L42_HSDET_COMP1_LVL_VAL << CS42L42_HSDET_COMP1_LVL_SHIFT) | + (CS42L42_HSDET_COMP2_LVL_VAL << CS42L42_HSDET_COMP2_LVL_SHIFT)); + /* Open the SW_HSB_HS3 switch and close SW_HSB_HS4 for a Type 1 headset. */ regmap_write(cs42l42->regmap, CS42L42_HS_SWITCH_CTL, CS42L42_HSDET_SW_COMP1); + msleep(100); + regmap_read(cs42l42->regmap, CS42L42_HS_DET_STATUS, &hs_det_status); - hs_det_comp = (hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> + hs_det_comp1 = (hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> CS42L42_HSDET_COMP1_OUT_SHIFT; + hs_det_comp2 = (hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> + CS42L42_HSDET_COMP2_OUT_SHIFT; /* Close the SW_HSB_HS3 switch for a Type 2 headset. */ regmap_write(cs42l42->regmap, CS42L42_HS_SWITCH_CTL, CS42L42_HSDET_SW_COMP2); + msleep(100); + regmap_read(cs42l42->regmap, CS42L42_HS_DET_STATUS, &hs_det_status); - hs_det_comp |= ((hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> + hs_det_comp1 |= ((hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> + CS42L42_HSDET_COMP1_OUT_SHIFT) << 1; + hs_det_comp2 |= ((hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> CS42L42_HSDET_COMP2_OUT_SHIFT) << 1; - switch (hs_det_comp) { + /* Use Comparator 1 with 1.25V Threshold. */ + switch (hs_det_comp1) { case CS42L42_HSDET_COMP_TYPE1: cs42l42->hs_type = CS42L42_PLUG_CTIA; hs_det_sw = CS42L42_HSDET_SW_TYPE1; @@ -1088,14 +1106,26 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = CS42L42_PLUG_OMTP; hs_det_sw = CS42L42_HSDET_SW_TYPE2; break; - case CS42L42_HSDET_COMP_TYPE3: - cs42l42->hs_type = CS42L42_PLUG_HEADPHONE; - hs_det_sw = CS42L42_HSDET_SW_TYPE3; - break; default: - cs42l42->hs_type = CS42L42_PLUG_INVALID; - hs_det_sw = CS42L42_HSDET_SW_TYPE4; - break; + /* Fallback to Comparator 2 with 1.75V Threshold. */ + switch (hs_det_comp2) { + case CS42L42_HSDET_COMP_TYPE1: + cs42l42->hs_type = CS42L42_PLUG_CTIA; + hs_det_sw = CS42L42_HSDET_SW_TYPE1; + break; + case CS42L42_HSDET_COMP_TYPE2: + cs42l42->hs_type = CS42L42_PLUG_OMTP; + hs_det_sw = CS42L42_HSDET_SW_TYPE2; + break; + case CS42L42_HSDET_COMP_TYPE3: + cs42l42->hs_type = CS42L42_PLUG_HEADPHONE; + hs_det_sw = CS42L42_HSDET_SW_TYPE3; + break; + default: + cs42l42->hs_type = CS42L42_PLUG_INVALID; + hs_det_sw = CS42L42_HSDET_SW_TYPE4; + break; + } } /* Set Switches */ @@ -1112,6 +1142,14 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) (0 << CS42L42_HSDET_SET_SHIFT) | (0 << CS42L42_HSBIAS_REF_SHIFT) | (0 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + + /* Configure HS DET comparator reference levels. */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL1, + CS42L42_HSDET_COMP1_LVL_MASK | + CS42L42_HSDET_COMP2_LVL_MASK, + (CS42L42_HSDET_COMP1_LVL_DEFAULT << CS42L42_HSDET_COMP1_LVL_SHIFT) | + (CS42L42_HSDET_COMP2_LVL_DEFAULT << CS42L42_HSDET_COMP2_LVL_SHIFT)); } static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) @@ -1134,6 +1172,18 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = (hs_det_status & CS42L42_HSDET_TYPE_MASK) >> CS42L42_HSDET_TYPE_SHIFT; + /* Set hs detect to automatic, disabled mode */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL2, + CS42L42_HSDET_CTRL_MASK | + CS42L42_HSDET_SET_MASK | + CS42L42_HSBIAS_REF_MASK | + CS42L42_HSDET_AUTO_TIME_MASK, + (2 << CS42L42_HSDET_CTRL_SHIFT) | + (2 << CS42L42_HSDET_SET_SHIFT) | + (0 << CS42L42_HSBIAS_REF_SHIFT) | + (3 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + /* Run Manual detection if auto detect has not found a headset. * We Re-Run with Manual Detection if the original detection was invalid or headphones, * to ensure that a headset mic is detected in all cases. @@ -1142,18 +1192,6 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type == CS42L42_PLUG_HEADPHONE) { dev_dbg(cs42l42->component->dev, "Running Manual Detection Fallback\n"); cs42l42_manual_hs_type_detect(cs42l42); - } else { - /* Set hs detect to automatic, disabled mode */ - regmap_update_bits(cs42l42->regmap, - CS42L42_HSDET_CTL2, - CS42L42_HSDET_CTRL_MASK | - CS42L42_HSDET_SET_MASK | - CS42L42_HSBIAS_REF_MASK | - CS42L42_HSDET_AUTO_TIME_MASK, - (2 << CS42L42_HSDET_CTRL_SHIFT) | - (2 << CS42L42_HSDET_SET_SHIFT) | - (0 << CS42L42_HSBIAS_REF_SHIFT) | - (3 << CS42L42_HSDET_AUTO_TIME_SHIFT)); } /* Set up button detection */ diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 2aeabba73e05..0704c902475f 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -188,6 +188,11 @@ #define CS42L42_HSDET_COMP2_LVL_SHIFT 4 #define CS42L42_HSDET_COMP2_LVL_MASK (15 << CS42L42_HSDET_COMP2_LVL_SHIFT) +#define CS42L42_HSDET_COMP1_LVL_VAL 12 /* 1.25V Comparator */ +#define CS42L42_HSDET_COMP2_LVL_VAL 2 /* 1.75V Comparator */ +#define CS42L42_HSDET_COMP1_LVL_DEFAULT 7 /* 1V Comparator */ +#define CS42L42_HSDET_COMP2_LVL_DEFAULT 7 /* 2V Comparator */ + #define CS42L42_HSDET_CTL2 (CS42L42_PAGE_11 + 0x20) #define CS42L42_HSDET_AUTO_TIME_SHIFT 0 #define CS42L42_HSDET_AUTO_TIME_MASK (3 << CS42L42_HSDET_AUTO_TIME_SHIFT) -- 2.25.1

2 1

[PATCH] ASoC: ad193x: 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 ad193x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ad193x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 278a55af158b..30b98b4267e1 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -243,22 +243,22 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, if (fmt & SND_SOC_DAIFMT_DSP_A) dac_fmt ^= AD193X_DAC_LEFT_HIGH; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: adc_fmt |= AD193X_ADC_LCR_MASTER; adc_fmt |= AD193X_ADC_BCLK_MASTER; dac_fmt |= AD193X_DAC_LCR_MASTER; dac_fmt |= AD193X_DAC_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ + case SND_SOC_DAIFMT_CBC_CFP: adc_fmt |= AD193X_ADC_LCR_MASTER; dac_fmt |= AD193X_DAC_LCR_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + case SND_SOC_DAIFMT_CBP_CFC: adc_fmt |= AD193X_ADC_BCLK_MASTER; dac_fmt |= AD193X_DAC_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

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[PATCH] ASoC: meson: aiu: Fix spelling mistake "Unsupport" -> "Unsupported"
by Colin King 27 Sep '21

27 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err error message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/meson/aiu-encoder-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-spdif.c b/sound/soc/meson/aiu-encoder-spdif.c index de850913975f..97da60db2c4d 100644 --- a/sound/soc/meson/aiu-encoder-spdif.c +++ b/sound/soc/meson/aiu-encoder-spdif.c @@ -113,7 +113,7 @@ static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream, val |= AIU_958_MISC_MODE_32BITS; break; default: - dev_err(dai->dev, "Unsupport physical width\n"); + dev_err(dai->dev, "Unsupported physical width\n"); return -EINVAL; } -- 2.32.0

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