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

[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

2 1

[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As part of the effort to remove our old APIs based on outdated terminology update the EP93xx drivers to use modern terminology.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

1 1

[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

1 1

[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

1 1

[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

2 1

[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

1 1

[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

3 2

[PATCH] ASoC: samsung: add missing "fallthrough;"
by Arnd Bergmann 27 Sep '21

27 Sep '21

From: Arnd Bergmann <arnd(a)arndb.de> With gcc, we get a warning in this file: In file included from include/linux/io.h:13, from sound/soc/samsung/s3c-i2s-v2.c:16: sound/soc/samsung/s3c-i2s-v2.c: In function 's3c2412_i2s_trigger': arch/arm/include/asm/io.h:92:22: error: this statement may fall through [-Werror=implicit-fallthrough=] #define __raw_writel __raw_writel ^ arch/arm/include/asm/io.h:299:29: note: in expansion of macro '__raw_writel' #define writel_relaxed(v,c) __raw_writel((__force u32) cpu_to_le32(v),c) ^~~~~~~~~~~~ arch/arm/include/asm/io.h:307:36: note: in expansion of macro 'writel_relaxed' #define writel(v,c) ({ __iowmb(); writel_relaxed(v,c); }) ^~~~~~~~~~~~~~ sound/soc/samsung/s3c-i2s-v2.c:398:3: note: in expansion of macro 'writel' writel(0x0, i2s->regs + S3C2412_IISFIC); ^~~~~~ sound/soc/samsung/s3c-i2s-v2.c:400:2: note: here case SNDRV_PCM_TRIGGER_RESUME: ^~~~ >From all I can tell, this was indeed meant to fall through, so add "fallthrough;" statement to avoid the warning. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/s3c-i2s-v2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index e9481187a08c..de66cc422e6e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -397,6 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, /* clear again, just in case */ writel(0x0, i2s->regs + S3C2412_IISFIC); + fallthrough; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!i2s->master) { -- 2.29.2

3 2

[PATCH] ASoC: alc5623: 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 alc5623 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/alc5623.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 54f489837162..b10357a6d655 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -641,12 +641,12 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; u16 iface = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface = ALC5623_DAI_SDP_MASTER_MODE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface = ALC5623_DAI_SDP_SLAVE_MODE; break; default: -- 2.20.1

1 1

[PATCH] ASoC: alc5632: 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 alc5632 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 79813882a955..6d7af3736a91 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -815,12 +815,12 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; u16 iface = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface = ALC5632_DAI_SDP_MASTER_MODE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface = ALC5632_DAI_SDP_SLAVE_MODE; break; default: -- 2.20.1

1 1

[PATCH] ASoC: ak5558: 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 ak5558 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak5558.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 37d4600b6f2c..c94cfde3e4a8 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -198,13 +198,13 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; u8 format; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: default: dev_err(dai->dev, "Clock mode unsupported"); return -EINVAL; -- 2.20.1

1 1

[PATCH] ASoC: cs42l42: Add HP Volume Scale control
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> cs42l42 has a configurable scaling of the maximum volume. Add an ALSA control for this. Note that the datasheet name is "full scale volume" but this conflicts with ALSA naming convention so the control is named "HP Volume Scale". Before this change the FULL_SCALE_VOLUME was set based on the value in RLA_STAT, but as there isn't any load detection result this always set the scaling to -6dB instead of the default 0dB. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5e1e5228b5f..78f8c89410b6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -425,6 +425,14 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); +static const char * const cs42l42_full_scale_vol_text[] = { + "0dB", "-6dB" +}; + +static SOC_ENUM_SINGLE_DECL(cs42l42_full_scale_vol_enum, CS42L42_HP_CTL, + CS42L42_HP_FULL_SCALE_VOL_SHIFT, + cs42l42_full_scale_vol_text); + static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, @@ -450,6 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DACB_INV_SHIFT, true, false), SOC_SINGLE("DAC HPF Switch", CS42L42_DAC_CTL2, CS42L42_DAC_HPF_EN_SHIFT, true, false), + SOC_ENUM("HP Volume Scale", cs42l42_full_scale_vol_enum), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, 0x3f, 1, mixer_tlv) @@ -922,7 +931,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +1001,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.25.1

2 1

Broken Pipe with SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Ryan McClue 27 Sep '21

27 Sep '21

Hello again. I'm using alsa-kernel. My configuration: SNDRV_PCM_HW_PARAM_ACCESS = SNDRV_PCM_ACCESS_RW_INTERLEAVED SNDRV_PCM_HW_PARAM_FORMAT = SNDRV_PCM_FORMAT_S16_LE SNDRV_PCM_HW_PARAM_SUBFORMAT = SNDRV_PCM_SUBFORMAT_STD SNDRV_PCM_HW_PARAM_CHANNELS = 2 SNDRV_PCM_HW_PARAM_RATE = 48000 I have a vsynced game loop that is running at 60fps: > int num_base_samples = 48000 * (1 / 60); > int num_samples = num_base_samples * 2; > int16_t buffer[num_samples] = {}; > > while (true) { > int16_t *samples = buffer; > for (int sample_i = 0; sample_i < num_base_samples; sample_i++) { > *samples++ = 0x33; > *samples++ = 0x33; > } > > struct snd_xferi transfer = {}; > transfer.buf = buffer; > transfer.frames = num_base_samples; > ioctl(fd, SNDRV_PCM_IOCTL_WRITEI_FRAMES, &transfer); > > // ioctl(fd, SNDRV_PCM_IOCTL_DRAIN); --> NECESSARY??? > // ioctl(fd, SNDRV_PCM_IOCTL_PREPARE); --> NECESSARY??? > } On the first iteration of SNDRV_PCM_IOCTL_WRITEI_FRAMES I get no error. All subsequent iterations, I get Broken Pipe error. So, to counteract this at the end of each frame I call SNDRV_PCM_IOCTL_DRAIN and SNDRV_PCM_IOCTL_PREPARE. This removes the Broken Pipe error however slows the program down by half and no sound is heard. How best to solve this issue? Thanks again -- Ryan McClue, Sydney

2 3

[PATCH v2 00/12] Add support for on demand pipeline setup/destroy
by Daniel Baluta 27 Sep '21

27 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries implements the new feature to setup/teardown pipeline as needed when a PCM is open/closed. This only only support single cores, with support for multicore arriving with a future patchseries. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 Changes since v1: - added my own Signed-off-by tag. Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 176 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1034 insertions(+), 538 deletions(-) -- 2.27.0

4 17

[PATCH v3] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 24 Sep '21

24 Sep '21

support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board varients add ALC5682I-VS support Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- v3: - add acpi_dev_present() for additional tweeter condition, otherwise stereo speakers. - add ALC5682I-VS headphone codec support --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 181 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 84 ++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 21 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 312 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3e20c697b569..760d845ac6a1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -466,6 +466,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..ea5058a5b331 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -5,6 +5,7 @@ #include <linux/string.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -133,6 +134,186 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +const struct snd_soc_dapm_route max_98390_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + /* add regular speakers dapm route */ + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "unable to add Left/Right Speaker dapm, ret %d\n", ret); + return ret; + } + + /* add widgets/controls/dapm for tweeter speakers */ + if (acpi_dev_present("MX98390", "3", -1)) { + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + + if (ret) { + dev_err(rtd->dev, "unable to add tweeter dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add tweeter card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, + "unable to add Tweeter Left/Right Speaker dapm, ret %d\n", ret); + } + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..9dd69f5d8721 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3f6f19d9b19e..774e4dd9539c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -59,6 +59,9 @@ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) #define SOF_RT5682S_HEADPHONE_CODEC_PRESENT BIT(23) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) + /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -179,6 +182,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -486,6 +519,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -784,6 +818,20 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + } + links[id].init = max_98390_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -864,6 +912,10 @@ static int sof_audio_probe(struct platform_device *pdev) if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + /* Detect the headset codec variant to support machines in DMI quirk */ + if (acpi_dev_present("RTL5682", NULL, -1)) + sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -920,6 +972,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1046,6 +1106,29 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + .name = "adl_mx98390_rt5682s", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_RT5682S_HEADPHONE_CODEC_PRESENT | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { .name = "jsl_rt5682s_rt1015", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -1095,6 +1178,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..a62a78edc855 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,22 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, + { + .id = "RTL5682", + .drv_name = "adl_mx98390_rt5682s", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

2 1

include a sample `alsamixer.rc` file?
by GitHub issues - opened 24 Sep '21

24 Sep '21

alsa-project/alsa-utils issue #112 was opened from int-e: I think it would be useful to include an example `alsamixer.rc` file; see https://gist.github.com/int-e/25bd77d4e59531fad7d48bfaf1fc1046 for a candidate. In particular this example replicates the default color theme, making it easier to adjust (say, replace `default` by `black` to get sane colors back in terminals with bright backgrounds). The default keybindings are also replicated but that's probably less useful. Issue URL : https://github.com/alsa-project/alsa-utils/issues/112 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2] ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
by Jaroslav Kysela 23 Sep '21

23 Sep '21

The new framing mode causes the user space regression, because the alsa-lib code does not initialize the reserved space in the params structure when the device is opened. This change adds SNDRV_RAWMIDI_IOCTL_USER_PVERSION like we do for the PCM interface for the protocol acknowledgment. Cc: David Henningsson <coding(a)diwic.se> Cc: <stable(a)vger.kernel.org> Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/rawmidi.c | 9 +++++++++ 3 files changed, 11 insertions(+) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 989e1517332d..7a08ed2acd60 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -98,6 +98,7 @@ struct snd_rawmidi_file { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; + unsigned int user_pversion; /* supported protocol version */ }; struct snd_rawmidi_str { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1d84ec9db93b..5859ca0a1439 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -784,6 +784,7 @@ struct snd_rawmidi_status { #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('W', 0x02, int) #define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) #define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) #define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e..6f30231bdb88 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) -- 2.31.1

2 1

[PATCH v7 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 22 Sep '21

22 Sep '21

Many thanks for reviewing v6 This version addresses all the comments raised as part of v7 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v6: - fixed last usage of flexible array in apr - removed leftover comments - added support to derive topology firmware name from soundcard to support different variants, as suggested by Pierre. Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1135 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 836 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1133 ++++++++++++++++ 31 files changed, 7203 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

2 31

[PATCH 01/16] ASoC: eureka-tlv320: Update to modern clocking terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the eureka-tlv320 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e13271ea84de..8b61582753c8 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &eukrea_tlv320_snd_ops, SND_SOC_DAILINK_REG(hifi), }; -- 2.20.1

2 18

[PATCH] ASoC: cros_ec_codec: Use modern ASoC DAI format terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cros_ec_codec driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cros_ec_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index a201d652aca2..9b92e1a0d1a3 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -283,8 +283,8 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_daifmt daifmt; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

2 2

[PATCH v2 01/16] ASoC: eureka-tlv320: Update to modern clocking terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the eureka-tlv320 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e13271ea84de..8b61582753c8 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &eukrea_tlv320_snd_ops, SND_SOC_DAILINK_REG(hifi), }; -- 2.20.1

2 17

Redesign the rawmidi midi read with timestamping API
by GitHub pull_request - opened 22 Sep '21

22 Sep '21

alsa-project/alsa-lib pull request #179 was opened from perexg: It's follow-up for https://github.com/alsa-project/alsa-lib/pull/173 and https://github.com/alsa-project/alsa-lib/issues/172 . I am confident, that the direct framing mechanism for the transfer of the midi stream with the timestamping should be hidden and abstracted in the alsa-lib's API. This version uses a new read function (`snd_rawmidi_tread()`) which decodes and merges the data from the kernel frames. The framing mechanism is very specific to the kernel API and the cost of added data copy is negligible in this case (the kernel code basically does similar thing). I kept the `snd_rawmidi_read()` functionality to read raw frames), but the structure for the parsing must be defined on it's own. This mechanism is unsupported. This version seems more abstract and easy to use for apps than the separate decoder function for `snd_rawmidi_read()`. Request URL : https://github.com/alsa-project/alsa-lib/pull/179 Patch URL : https://github.com/alsa-project/alsa-lib/pull/179.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RFC PATCH] component: do not leave master devres group open after bind
by Kai Vehmanen 22 Sep '21

22 Sep '21

In current code, the devres group for aggregate master is left open after call to component_master_add_*(). This leads to problems when the master does further managed allocations on its own. When any participating driver calls component_del(), this leads to immediate release of resources. This came up when investigating a page fault occurring with i915 DRM driver unbind with 5.15-rc1 kernel. The following sequence occurs: i915_pci_remove() -> intel_display_driver_unregister() -> i915_audio_component_cleanup() -> component_del() -> component.c:take_down_master() -> hdac_component_master_unbind() [via master->ops->unbind()] -> devres_release_group(master->parent, NULL) With older kernels this has not caused issues, but with audio driver moving to use managed interfaces for more of its allocations, this no longer works. Devres log shows following to occur: component_master_add_with_match() [ 126.886032] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000323ccdc5 devm_component_match_release (24 bytes) [ 126.886045] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000865cdb29 grp< (0 bytes) [ 126.886049] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 grp< (0 bytes) audio driver completes its PCI probe() [ 126.892238] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 pcim_iomap_release (48 bytes) component_del() called() at DRM/i915 unbind() [ 137.579422] i915 0000:00:02.0: DEVRES REL 00000000ef44c293 grp< (0 bytes) [ 137.579445] snd_hda_intel 0000:00:1f.3: DEVRES REL 00000000865cdb29 grp< (0 bytes) [ 137.579458] snd_hda_intel 0000:00:1f.3: DEVRES REL 000000001b480725 pcim_iomap_release (48 bytes) So the "devres_release_group(master->parent, NULL)" ends up freeing the pcim_iomap allocation. Upon next runtime resume, the audio driver will cause a page fault as the iomap alloc was released without the driver knowing about it. Fix this issue by using the "struct master" pointer as identifier for the devres group, and by closing the devres group after the master->ops->bind() call is done. This allows devres allocations done by the driver acting as master to be isolated from the binding state of the aggregate driver. This modifies the logic originally introduced in commit 9e1ccb4a7700 ("drivers/base: fix devres handling for master device"). BugLink: https://gitlab.freedesktop.org/drm/intel/-/issues/4136 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- drivers/base/component.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) Hi, I'm sending this as RFC as I'm not sure of the implications of not leaving the devres group open might have to other users of the component framework. For audio, the current behaviour seems very problematic. The display codec is usually just one of many audio codecs attached to the controller, and unbind of the display codec (and the aggregate driver created with DRM), should not bring down the whole audio card. However, now all allocations audio driver does after call to component_master_add_with_match(), will be freed when display driver calls component_del(). Closing the devres group at end of component_master_add_*() would seem the cleanest option. Looking for feedback whether this approach is feasible. One alternative would be for the audio driver to close the "last opened" group after its call to component_master_add(), but this seems messy (audio would make assumptions on component.c internals). diff --git a/drivers/base/component.c b/drivers/base/component.c index 5e79299f6c3f..870485cbbb87 100644 --- a/drivers/base/component.c +++ b/drivers/base/component.c @@ -246,7 +246,7 @@ static int try_to_bring_up_master(struct master *master, return 0; } - if (!devres_open_group(master->parent, NULL, GFP_KERNEL)) + if (!devres_open_group(master->parent, master, GFP_KERNEL)) return -ENOMEM; /* Found all components */ @@ -258,6 +258,7 @@ static int try_to_bring_up_master(struct master *master, return ret; } + devres_close_group(master->parent, NULL); master->bound = true; return 1; } @@ -282,7 +283,7 @@ static void take_down_master(struct master *master) { if (master->bound) { master->ops->unbind(master->parent); - devres_release_group(master->parent, NULL); + devres_release_group(master->parent, master); master->bound = false; } } base-commit: 930e99a51fcc8b1254e0a45fbe0cd5a5b8a704a5 -- 2.32.0

3 2

Re: [LAD] Mackie Onyx 1640i: Long run playback results in xrun and hang.
by AreYouLoco? 21 Sep '21

21 Sep '21

I am going to recompile the kernel with options mentioned here: https://www.alsa-project.org/main/index.php/XRUN_Debug And try to debug why the xrun appears. I also checked with buffer size 1024 and exaclty one xrun and then hang repeats. On 9/13/21 12:23 PM, AreYouLoco? wrote: > Hi, > > I am using custom build real-time kernel 5.12.2-rt3-fw. JACK with ALSA > backend on Mackie Onyx 1640i. > Lenovo Thinkpad T420 with FireWire. Debian Bullseye (11.0 - stable) > JACKD version 1.9.17. > I am using 512 samples 3 periods @48kHz. > > The problem is that long run playback only around 1h (but the time > differs), result in > XRUN detected by JACK logs and the whole audio stops. So I have to > restart JACK in order to make it work. > DMESG is silent about that issue. But sometimes I get: > > snd_oxfw fw1.0: Detect discontinuity of CIP: E8 00 > And different hex values there. > > Here is interesting part from the log: > > Sun Sep 12 22:10:12 2021: Jack: JackRequest::Notification > Sun Sep 12 22:10:12 2021: Jack: **** alsa_pcm: xrun of at least 0.027 msecs > Sun Sep 12 22:10:12 2021: Jack: Repreparing capture > Sun Sep 12 22:10:12 2021: Jack: Repreparing playback > Sun Sep 12 22:10:12 2021: Jack: ALSA XRun wait_status = 0 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackClient::ClientNotify ref = 2 name = > dbusapi notify = 3 > Sun Sep 12 22:10:12 2021: Jack: JackClient::kXRunCallback > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:28 2021: ERROR: ALSA: poll time out, polled for > 15998655 usecs, Retrying with a recovery, retry cnt = 1 > Sun Sep 12 22:10:44 2021: ERROR: ALSA: poll time out, polled for > 15998654 usecs, Retrying with a recovery, retry cnt = 2 > Sun Sep 12 22:11:00 2021: ERROR: ALSA: poll time out, polled for > 15998659 usecs, Retrying with a recovery, retry cnt = 3 > Sun Sep 12 22:11:16 2021: ERROR: ALSA: poll time out, polled for > 15998649 usecs, Retrying with a recovery, retry cnt = 4 > > Anyone have idea what might be the cause of this XRUN? Or how to debug that? > > Its reproducible on my setup. Another log catched the same issue: > > Sun Sep 12 16:50:44 2021: Jack: **** alsa_pcm: xrun of at least 0.030 msecs > Sun Sep 12 16:50:44 2021: Jack: Repreparing capture > Sun Sep 12 16:50:44 2021: Jack: Repreparing playback > Sun Sep 12 16:50:44 2021: Jack: ALSA XRun wait_status = 0 > Sun Sep 12 16:50:44 2021: Jack: JackRequest::Notification > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackClient::ClientNotify ref = 2 name = > dbusapi notify = 3 > Sun Sep 12 16:50:44 2021: Jack: JackClient::kXRunCallback > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:51:00 2021: ERROR: ALSA: poll time out, polled for > 15998366 usecs, Retrying with a recovery, retry cnt = 1 > Sun Sep 12 16:51:16 2021: ERROR: ALSA: poll time out, polled for > 15998353 usecs, Retrying with a recovery, retry cnt = 2 > Sun Sep 12 16:51:32 2021: ERROR: ALSA: poll time out, polled for > 15998352 usecs, Retrying with a recovery, retry cnt = 3 > Sun Sep 12 16:51:48 2021: ERROR: ALSA: poll time out, polled for > 15998367 usecs, Retrying with a recovery, retry cnt = 4 > Sun Sep 12 16:52:04 2021: ERROR: ALSA: poll time out, polled for > 15998353 usecs, Retrying with a recovery, retry cnt = 5 > Sun Sep 12 16:52:20 2021: ERROR: ALSA: poll time out, polled for > 15998371 usecs, Reached max retry cnt = 5, Exiting > Sun Sep 12 16:52:20 2021: ERROR: JackAudioDriver::ProcessAsync: read > error, stopping... > Sun Sep 12 16:52:20 2021: Jack: JackPosixThread::ThreadHandler : exit > > I am willing to provide additional info. > > Thank you in advance, > > AreYouLoco? > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev(a)lists.linuxaudio.org > https://lists.linuxaudio.org/listinfo/linux-audio-dev

1 1

[PATCH] ASoC: Fix warning related to 'sound-name-prefix' binding
by Sameer Pujar 21 Sep '21

21 Sep '21

commit 82d3ec1d89fa ("ASoC: Use schema reference for sound-name-prefix") added name-prefix.yaml schema and the same reference was used in couple of other schemas. But this is causing following warning and the same is fixed in current patch. Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml: audio-codec@34: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml: audio-codec@36: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml Fixes: 82d3ec1d89fa ("ASoC: Use schema reference for sound-name-prefix") Reported-by: Rob Herring <robh+dt(a)kernel.org> Suggested-by: Rob Herring <robh+dt(a)kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 2 ++ Documentation/devicetree/bindings/sound/simple-audio-mux.yaml | 2 ++ 2 files changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index 68e5ad2..7667471 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -24,6 +24,8 @@ properties: '#sound-dai-cells': const: 0 + sound-name-prefix: true + vddd-supply: description: regulator phandle for the VDDD power supply. diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml index c597d9d..b5fc35e 100644 --- a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -24,6 +24,8 @@ properties: description: | GPIOs used to select the input line. + sound-name-prefix: true + required: - compatible - mux-gpios -- 2.7.4

2 1

[PATCH] ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers
by Peter Rosin 21 Sep '21

21 Sep '21

>From 625f858894af2b7e547cc723b97361081438b123 Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Commit 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") breaks the TSE-850 device, which is using a pcm5142 in I2S and CBM_CFS mode (maybe not relevant). Without this fix, the result is: pcm512x 0-004c: Failed to set data format: -16 And after that, no sound. This fix is not 100% correct. The datasheet of at least the pcm5142 states that four bits (0xcc) in the I2S_1 register are "RSV" ("Reserved. Do not access.") and no hint is given as to what the initial values are supposed to be. So, specifying defaults for these bits is wrong. But perhaps better than a broken driver? Fixes: 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kirill Marinushkin <kmarinushkin(a)birdec.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc..60dee41816dc 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -116,6 +116,8 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_FS_SPEED_MODE, 0x00 }, { PCM512x_IDAC_1, 0x01 }, { PCM512x_IDAC_2, 0x00 }, + { PCM512x_I2S_1, 0x02 }, + { PCM512x_I2S_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) -- 2.20.1

3 13

[PATCH 0/7] add internal channels support
by Olivier Moysan 21 Sep '21

21 Sep '21

This patchset adds support of ADC2 internal channels VDDCORE, VREFINT and VBAT on STM32MP15x SoCs. The generic IIO channel bindings is also introduced here to provide this feature. The legacy channel binding is kept for backward compatibility. Olivier Moysan (7): dt-bindings: iio: adc: add generic channel binding dt-bindings: iio: adc: add nvmem support for vrefint internal channel iio: adc stm32-adc: split channel init into several routines iio: adc: stm32-adc: add support of generic channels binding iio: adc: stm32-adc: add support of internal channels iio: adc: stm32-adc: add vrefint calibration support iio: adc: stm32-adc: use generic binding for sample-time .../bindings/iio/adc/st,stm32-adc.yaml | 108 ++++- drivers/iio/adc/stm32-adc-core.h | 8 + drivers/iio/adc/stm32-adc.c | 418 ++++++++++++++++-- 3 files changed, 482 insertions(+), 52 deletions(-) -- 2.17.1

3 14

[PATCH] ALSA: firewire-motu: fix truncated bytes in message tracepoints
by Takashi Sakamoto 21 Sep '21

21 Sep '21

In MOTU protocol v2/v3, first two data chunks across 2nd and 3rd data channels includes message bytes from device. The total size of message is 48 bits per data block. The 'data_block_message' tracepoints event produced by ALSA firewire-motu driver exposes the sequence of messages to userspace in 64 bit storage, however lower 32 bits are actually available since current implementation truncates 16 bits in upper of the message as a result of bit shift operation within 32 bit storage. This commit fixes the bug by perform the bit shift in 64 bit storage. Fixes: c6b0b9e65f09 ("ALSA: firewire-motu: add tracepoints for messages for unique protocol") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 5388b85fb60e..a18c2c033e83 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, /* This is just for v2/v3 protocol. */ for (i = 0; i < data_blocks; ++i) { - *frames = (be32_to_cpu(buffer[1]) << 16) | - (be32_to_cpu(buffer[2]) >> 16); + *frames = be32_to_cpu(buffer[1]); + *frames <<= 16; + *frames |= be32_to_cpu(buffer[2]) >> 16; + ++frames; buffer += data_block_quadlets; - frames++; } } -- 2.30.2

2 1

[GIT PULL] ASoC fixes for v5.15-rc2
by Mark Brown 21 Sep '21

21 Sep '21

The following changes since commit fcb958ee8e832e9cdf43408535207e15f14af755: ASoC: rockchip: i2s: Fix concurrency between tx/rx (2021-09-03 16:19:01 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc2 for you to fetch changes up to cfacfefd382af3b42905108b54f02820dca225c4: ASoC: SOF: trace: Omit error print when waking up trace sleepers (2021-09-17 13:16:36 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A crop of mostly device specific fixes that have been applied since the merge window, nothing particularly standout here. ---------------------------------------------------------------- Arnaud Pouliquen (1): MAINTAINERS: fix update references to stm32 audio bindings Marc Herbert (1): ASoC: SOF: loader: release_firmware() on load failure to avoid batching Mark Brown (1): Merge series "ASoC: fsl: register platform component before registering cpu dai" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Peter Ujfalusi (4): ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication ASoC: SOF: trace: Omit error print when waking up trace sleepers Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic Shengjiu Wang (5): ASoC: fsl_sai: register platform component before registering cpu dai ASoC: fsl_esai: register platform component before registering cpu dai ASoC: fsl_micfil: register platform component before registering cpu dai ASoC: fsl_spdif: register platform component before registering cpu dai ASoC: fsl_xcvr: register platform component before registering cpu dai Trevor Wu (3): ASoC: mediatek: add required config dependency ASoC: mediatek: common: handle NULL case in suspend/resume function ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX Yong Zhi (1): ASoC: SOF: Fix DSP oops stack dump output contents MAINTAINERS | 3 ++- sound/soc/fsl/fsl_esai.c | 16 ++++++++++------ sound/soc/fsl/fsl_micfil.c | 15 ++++++++++----- sound/soc/fsl/fsl_sai.c | 14 +++++++++----- sound/soc/fsl/fsl_spdif.c | 14 +++++++++----- sound/soc/fsl/fsl_xcvr.c | 15 ++++++++++----- sound/soc/intel/boards/sof_sdw.c | 5 +++++ sound/soc/mediatek/Kconfig | 3 +++ sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- .../soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 7 +++---- sound/soc/sof/core.c | 4 +--- sound/soc/sof/imx/imx8.c | 9 ++++++++- sound/soc/sof/imx/imx8m.c | 9 ++++++++- sound/soc/sof/loader.c | 8 +++++--- sound/soc/sof/trace.c | 1 - sound/soc/sof/xtensa/core.c | 4 ++-- 16 files changed, 96 insertions(+), 50 deletions(-)

2 1

Re: [PATCH] ALSA: usx2y: Prefer struct_size over open coded arithmetic
by Takashi Iwai 21 Sep '21

21 Sep '21

On Sun, 19 Sep 2021 15:37:27 +0200, Len Baker wrote: > > As noted in the "Deprecated Interfaces, Language Features, Attributes, > and Conventions" documentation [1], size calculations (especially > multiplication) should not be performed in memory allocator (or similar) > function arguments due to the risk of them overflowing. This could lead > to values wrapping around and a smaller allocation being made than the > caller was expecting. Using those allocations could lead to linear > overflows of heap memory and other misbehaviors. > > In this case this is not actually dynamic size: all the operands > involved in the calculation are constant values. However it is better to > refactor this anyway, just to keep the open-coded math idiom out of > code. > > So, use the struct_size() helper to do the arithmetic instead of the > argument "size + size * count" in the kzalloc() function. > > Also, take the opportunity to refactor the declaration variables to make > it more easy to read. > > [1] https://www.kernel.org/doc/html/latest/process/deprecated.html#open-coded-a… > > Signed-off-by: Len Baker <len.baker(a)gmx.com> Thanks, applied. Takashi

1 0

[PATCH] ASoC: fsl_spdif: Add support for i.MX8ULP
by Shengjiu Wang 21 Sep '21

21 Sep '21

On i.MX8ULP the spdif works with EDMA, so add compatible string and soc specific data for i.MX8ULP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8ffb1a6048d6..61ba08322f23 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -186,6 +186,16 @@ static struct fsl_spdif_soc_data fsl_spdif_imx8mm = { .tx_formats = FSL_SPDIF_FORMATS_PLAYBACK, }; +static struct fsl_spdif_soc_data fsl_spdif_imx8ulp = { + .imx = true, + .shared_root_clock = true, + .raw_capture_mode = false, + .interrupts = 1, + .tx_burst = 2, /* Applied for EDMA */ + .rx_burst = 2, /* Applied for EDMA */ + .tx_formats = SNDRV_PCM_FMTBIT_S24_LE, /* Applied for EDMA */ +}; + /* Check if clk is a root clock that does not share clock source with others */ static inline bool fsl_spdif_can_set_clk_rate(struct fsl_spdif_priv *spdif, int clk) { @@ -1556,6 +1566,7 @@ static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx6sx-spdif", .data = &fsl_spdif_imx6sx, }, { .compatible = "fsl,imx8qm-spdif", .data = &fsl_spdif_imx8qm, }, { .compatible = "fsl,imx8mm-spdif", .data = &fsl_spdif_imx8mm, }, + { .compatible = "fsl,imx8ulp-spdif", .data = &fsl_spdif_imx8ulp, }, {} }; MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); -- 2.17.1

2 1

[PATCH] ASoC: fsl_xcvr: Fix channel swap issue with ARC
by Shengjiu Wang 21 Sep '21

21 Sep '21

With pause and resume test for ARC, there is occasionally channel swap issue. The reason is that currently driver set the DPATH out of reset first, then start the DMA, the first data got from FIFO may not be the Left channel. Moving DPATH out of reset operation after the dma enablement to fix this issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 31c5ee641fe7..6e67033b6cde 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -487,8 +487,9 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return ret; } - /* clear DPATH RESET */ + /* set DPATH RESET */ m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); + v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -590,10 +591,6 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, val |= FSL_XCVR_EXT_CTRL_CMDC_RESET(tx); } - /* set DPATH RESET */ - mask |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - val |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, mask, val); if (ret < 0) { dev_err(dai->dev, "Err setting DPATH RESET: %d\n", ret); @@ -643,6 +640,16 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(dai->dev, "Failed to enable DMA: %d\n", ret); return ret; } + + /* clear DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear DPATH RESET: %d\n", ret); + return ret; + } + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: -- 2.17.1

2 1

[PATCH] ASoC: qdsp6: q6afe-dai: Fix spelling mistake "Fronend" -> "Frontend"
by Colin King 21 Sep '21

21 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the module description. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index ac8f7324e94b..8b664cbf6fa6 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1747,5 +1747,5 @@ static struct platform_driver q6afe_dai_platform_driver = { }; module_platform_driver(q6afe_dai_platform_driver); -MODULE_DESCRIPTION("Q6 Audio Fronend dai driver"); +MODULE_DESCRIPTION("Q6 Audio Frontend dai driver"); MODULE_LICENSE("GPL v2"); -- 2.32.0

3 2

[PATCH] ASoC: fsl: Constify static snd_soc_ops
by Rikard Falkeborn 21 Sep '21

21 Sep '21

These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. 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/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-card.c | 4 ++-- sound/soc/fsl/imx-hdmi.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index a364e2415de0..0d637929bfef 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -132,12 +132,12 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops imx_audmix_fe_ops = { +static const struct snd_soc_ops imx_audmix_fe_ops = { .startup = imx_audmix_fe_startup, .hw_params = imx_audmix_fe_hw_params, }; -static struct snd_soc_ops imx_audmix_be_ops = { +static const struct snd_soc_ops imx_audmix_be_ops = { .hw_params = imx_audmix_be_hw_params, }; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 58fd0639a069..05dff2dc1d19 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -442,12 +442,12 @@ static int imx_aif_startup(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops imx_aif_ops = { +static const struct snd_soc_ops imx_aif_ops = { .hw_params = imx_aif_hw_params, .startup = imx_aif_startup, }; -static struct snd_soc_ops imx_aif_ops_be = { +static const struct snd_soc_ops imx_aif_ops_be = { .hw_params = imx_aif_hw_params, }; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 34a0dceae621..2b663c39edb2 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -59,7 +59,7 @@ static int imx_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops imx_hdmi_ops = { +static const struct snd_soc_ops imx_hdmi_ops = { .hw_params = imx_hdmi_hw_params, }; -- 2.33.0

2 1

[PATCH v3 0/3] Convert name-prefix doc to json-schema
by Sameer Pujar 21 Sep '21

21 Sep '21

Following are the changes: - Add json-schema for 'sound-name-prefix' documentation under 'name-perfix.yaml' - Use schema references wherever needed. - Remove txt based doc Changelog ========= v2 -> v3 -------- * Removed examples in patch 1/3 as suggested by Rob * Added reviewed tag from Jerome for the series * Added reviewed tag from Rob for patch 2/3 v1 -> v2 -------- * Provide top reference to name-prefix.yaml as suggested by Rob for patch 2/3 * Dropped couple of unreachable email ids from Cc list in commit message of patch 2/3 * No changes in remaining patches Sameer Pujar (3): ASoC: Add json-schema documentation for sound-name-prefix ASoC: Use schema reference for sound-name-prefix ASoC: Remove name-prefix.txt .../devicetree/bindings/sound/name-prefix.txt | 24 ---------------------- .../devicetree/bindings/sound/name-prefix.yaml | 21 +++++++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 9 +++----- .../bindings/sound/nvidia,tegra210-dmic.yaml | 9 +++----- .../bindings/sound/nvidia,tegra210-i2s.yaml | 9 +++----- .../devicetree/bindings/sound/nxp,tfa989x.yaml | 9 +++----- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- .../bindings/sound/simple-audio-mux.yaml | 9 +++----- 8 files changed, 37 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.yaml -- 2.7.4

4 8

[PATCH 0/7] Update Lpass digital codec macro drivers
by Srinivasa Rao Mandadapu 21 Sep '21

21 Sep '21

This patch set is to add support for lpass sc7280 based targets. Upadate compatible name and change of bulk clock voting to individual clock voting in digital codec va, rx, tx macro drivers. Srinivasa Rao Mandadapu (7): ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Change mic control registers to volatile ASoC: codecs: lpass-va-macro: Change bulk voting to individual clock voting ASoC: codecs: lpass-rx-macro: Change bulk voting to individual clock voting ASoC: codecs: lpass-tx-macro: Change bulk voting to individual clock voting ASoC: codecs: lpass-va-macro: set mclk clock rate correctly .../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +- sound/soc/codecs/lpass-rx-macro.c | 69 ++++++++++++------ sound/soc/codecs/lpass-tx-macro.c | 81 ++++++++++++++++------ sound/soc/codecs/lpass-va-macro.c | 49 ++++++++----- sound/soc/codecs/lpass-wsa-macro.c | 1 + 8 files changed, 153 insertions(+), 63 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 19

[PATCH] ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
by Jaroslav Kysela 20 Sep '21

20 Sep '21

The new framing mode causes the user space regression, because the alsa-lib code does not initialize the reserved space in the params structure when the device is opened. This change adds SNDRV_RAWMIDI_IOCTL_USER_PVERSION like we do for the PCM interface for the protocol acknowledgment. Cc: David Henningsson <coding(a)diwic.se> Cc: <stable(a)vger.kernel.org> Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/rawmidi.c | 9 +++++++++ 3 files changed, 11 insertions(+) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 989e1517332d..7a08ed2acd60 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -98,6 +98,7 @@ struct snd_rawmidi_file { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; + unsigned int user_pversion; /* supported protocol version */ }; struct snd_rawmidi_str { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1d84ec9db93b..f906e50a7919 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -784,6 +784,7 @@ struct snd_rawmidi_status { #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('A', 0x02, int) #define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) #define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) #define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e..6f30231bdb88 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) -- 2.31.1

2 2

[PATCH] ASoC: rt5682s: make rt5682s_aif2_dai_ops and rt5682s_soc_component_dev
by Jiapeng Chong 20 Sep '21

20 Sep '21

This symbol is not used outside of rt5682s.c, so marks it static. Fix the following sparse warning: sound/soc/codecs/rt5682s.c:2848:39: warning: symbol 'rt5682s_soc_component_dev' was not declared. Should it be static? sound/soc/codecs/rt5682s.c:2842:30: warning: symbol 'rt5682s_aif2_dai_ops' was not declared. Should it be static? Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rt5682s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d878a20..d3e965b 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2839,13 +2839,13 @@ static int rt5682s_resume(struct snd_soc_component *component) .set_bclk_ratio = rt5682s_set_bclk1_ratio, }; -const struct snd_soc_dai_ops rt5682s_aif2_dai_ops = { +static const struct snd_soc_dai_ops rt5682s_aif2_dai_ops = { .hw_params = rt5682s_hw_params, .set_fmt = rt5682s_set_dai_fmt, .set_bclk_ratio = rt5682s_set_bclk2_ratio, }; -const struct snd_soc_component_driver rt5682s_soc_component_dev = { +static const struct snd_soc_component_driver rt5682s_soc_component_dev = { .probe = rt5682s_probe, .remove = rt5682s_remove, .suspend = rt5682s_suspend, -- 1.8.3.1

2 1

[PATCH] ASoC: ab8500: Update to modern clocking terminology
by Mark Brown 20 Sep '21

20 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ab8500 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ab8500-codec.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 5525e1ccab76..aefafb0b7b97 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2104,26 +2104,26 @@ static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) BIT(AB8500_DIGIFCONF3_IF0MASTER); val = 0; - 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: dev_dbg(dai->component->dev, - "%s: IF0 Master-mode: AB8500 master.\n", __func__); + "%s: IF0 Master-mode: AB8500 provider.\n", __func__); val |= BIT(AB8500_DIGIFCONF3_IF0MASTER); break; - case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */ + case SND_SOC_DAIFMT_CBC_CFC: dev_dbg(dai->component->dev, - "%s: IF0 Master-mode: AB8500 slave.\n", __func__); + "%s: IF0 Master-mode: AB8500 consumer.\n", __func__); break; - case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */ - case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: dev_err(dai->component->dev, - "%s: ERROR: The device is either a master or a slave.\n", + "%s: ERROR: The device is either a provider or a consumer.\n", __func__); fallthrough; default: dev_err(dai->component->dev, - "%s: ERROR: Unsupporter master mask 0x%x\n", - __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + "%s: ERROR: Unsupporter clocking mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK); return -EINVAL; } -- 2.20.1

2 2

[PATCH] ASoC: dt-bindings: rt5682s: correct several errors
by Krzysztof Kozlowski 20 Sep '21

20 Sep '21

Correct several errors in rt5682s dtschema: 1. The examples should be under "examples": 'example' is not one of ['$id', '$schema', 'title', 'description', 'examples', ... 2. Missing type for vendor properties 3. clock-names should be an array: properties:clock-names:items: {'const': 'mclk'} is not of type 'array' 4. Example DTS should include headers: [scripts/Makefile.lib:386: Documentation/devicetree/bindings/sound/realtek,rt5682s.example.dt.yaml] Error 1 5. Node name in example DTS misses unit address and does not match DT convention (generic name): Warning (reg_format): /example-0/rt5682s:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) 6. Node address should be in size-cells:0 block in example DTS: Warning (reg_format): /example-0/codec@1a:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) Fixes: 50159fdb144b ("ASoC: dt-bindings: rt5682s: add bindings for rt5682s") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> --- .../bindings/sound/realtek,rt5682s.yaml | 47 ++++++++++++------- 1 file changed, 29 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index fbf23696f1a7..7a5f1d0fd3e2 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -24,18 +24,21 @@ properties: description: The CODEC's interrupt output. realtek,dmic1-data-pin: + $ref: /schemas/types.yaml#/definitions/uint32 enum: - 0 # dmic1 data is not used - 1 # using GPIO2 pin as dmic1 data pin - 2 # using GPIO5 pin as dmic1 data pin realtek,dmic1-clk-pin: + $ref: /schemas/types.yaml#/definitions/uint32 enum: - 0 # dmic1 clk is not used - 1 # using GPIO1 pin as dmic1 clock pin - 2 # using GPIO3 pin as dmic1 clock pin realtek,jd-src: + $ref: /schemas/types.yaml#/definitions/uint32 enum: - 0 # No JD is used - 1 # using JD1 as JD source @@ -63,7 +66,7 @@ properties: clock-names: items: - const: mclk + - const: mclk "#clock-cells": const: 1 @@ -79,22 +82,30 @@ required: - compatible - reg -example: +examples: - | - rt5682s { - compatible = "realtek,rt5682s"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; - realtek,ldo1-en-gpios = - <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; - realtek,dmic1-data-pin = <1>; - realtek,dmic1-clk-pin = <1>; - realtek,jd-src = <1>; - - #clock-cells = <1>; - clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; - - clocks = <&osc>; - clock-names = "mclk"; + #include <dt-bindings/gpio/tegra-gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "realtek,rt5682s"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; + }; }; -- 2.30.2

2 1

[PATCH] ASoC: SOF: Remove struct sof_ops_table and sof_get_ops() macro
by Peter Ujfalusi 20 Sep '21

20 Sep '21

sof_get_ops() is not used and the struct sof_ops_table is only used by that macro. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ops.h | 15 --------------- sound/soc/sof/sof-priv.h | 6 ------ 2 files changed, 21 deletions(-) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index b4121e516585..d40ce2581d94 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -524,21 +524,6 @@ snd_sof_set_mach_params(const struct snd_soc_acpi_mach *mach, sof_ops(sdev)->set_mach_params(mach, sdev); } -static inline const struct snd_sof_dsp_ops -*sof_get_ops(const struct sof_dev_desc *d, - const struct sof_ops_table mach_ops[], int asize) -{ - int i; - - for (i = 0; i < asize; i++) { - if (d == mach_ops[i].desc) - return mach_ops[i].ops; - } - - /* not found */ - return NULL; -} - /** * snd_sof_dsp_register_poll_timeout - Periodically poll an address * until a condition is met or a timeout occurs diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9b1bdba15c74..6b1dbae3344c 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -294,12 +294,6 @@ struct dsp_arch_ops { #define sof_dsp_arch_ops(sdev) ((sdev)->pdata->desc->ops->dsp_arch_ops) -/* DSP device HW descriptor mapping between bus ID and ops */ -struct sof_ops_table { - const struct sof_dev_desc *desc; - const struct snd_sof_dsp_ops *ops; -}; - /* FS entry for debug files that can expose DSP memories, registers */ struct snd_sof_dfsentry { size_t size; -- 2.33.0

2 1

[PATCH] ASoC: Intel: boards: Update to modern clocking terminology
by Peter Ujfalusi 20 Sep '21

20 Sep '21

As part of the effort to remove our old APIs based on outdated terminology update the Intel board drivers to use modern terminology. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 4 ++-- sound/soc/intel/boards/bytcht_da7213.c | 4 ++-- sound/soc/intel/boards/bytcht_es8316.c | 4 ++-- sound/soc/intel/boards/bytcht_nocodec.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- sound/soc/intel/boards/bytcr_wm5102.c | 4 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 4 ++-- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 6 +++--- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- sound/soc/intel/boards/skl_rt286.c | 2 +- 26 files changed, 46 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index c5122d3b0e6c..6cba5552f7a2 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -251,7 +251,7 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = broadwell_ssp0_fixup, .ops = &bdw_rt5650_ops, diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index e01b7a90ca6c..119c441f4c10 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -351,7 +351,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = broadwell_ssp0_fixup, .ops = &bdw_rt5677_ops, diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 3c3aff9c61cc..618d0645ed8d 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -217,7 +217,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .no_pcm = 1, .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = broadwell_ssp0_fixup, .ops = &broadwell_rt286_ops, diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index e67ddfb8e469..b768d9b8ec02 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -572,7 +572,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp_fixup, .dpcm_playback = 1, @@ -585,7 +585,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .no_pcm = 1, .init = broxton_da7219_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp_fixup, .dpcm_playback = 1, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 47f6b1523ae6..920e575b4314 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -468,7 +468,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { .no_pcm = 1, .init = broxton_rt298_codec_init, .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp5_fixup, .ops = &broxton_rt298_ops, diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index a9e51bbf018c..0a736308052a 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -126,7 +126,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; @@ -195,7 +195,7 @@ static struct snd_soc_dai_link byt_cht_cx2072x_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .init = byt_cht_cx2072x_init, .be_hw_params_fixup = byt_cht_cx2072x_fixup, .dpcm_playback = 1, diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index a28773fb7892..fae1e7e785b0 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -81,7 +81,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; @@ -195,7 +195,7 @@ static struct snd_soc_dai_link dailink[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = codec_fixup, .dpcm_playback = 1, .dpcm_capture = 1, diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 055248f104b2..1bb9b8e7bcc7 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -265,7 +265,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS + SND_SOC_DAIFMT_CBC_CFC ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -336,7 +336,7 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = byt_cht_es8316_codec_fixup, .dpcm_playback = 1, .dpcm_capture = 1, diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 9b48fe701a2c..67b3c4e97864 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -61,7 +61,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -141,7 +141,7 @@ static struct snd_soc_dai_link dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = codec_fixup, .ignore_suspend = 1, .dpcm_playback = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a6e837290c7d..c28abe74816f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1336,7 +1336,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; @@ -1411,7 +1411,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = byt_rt5640_codec_fixup, .dpcm_playback = 1, .dpcm_capture = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index e94c9124d4f4..00c347b52863 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -713,7 +713,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS + SND_SOC_DAIFMT_CBC_CFC ); if (ret < 0) { @@ -798,7 +798,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = byt_rt5651_codec_fixup, .dpcm_playback = 1, .dpcm_capture = 1, diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 580d5fddae5a..504ef4cab111 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -265,7 +265,7 @@ static int byt_wm5102_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret) { dev_err(rtd->dev, "Error setting format to I2S: %d\n", ret); return ret; @@ -349,7 +349,7 @@ static struct snd_soc_dai_link byt_wm5102_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = byt_wm5102_codec_fixup, .dpcm_playback = 1, .dpcm_capture = 1, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 131882378a59..1bc21434c9de 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -264,7 +264,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, } fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS; + | SND_SOC_DAIFMT_CBC_CFC; ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret < 0) { @@ -372,7 +372,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, .dpcm_playback = 1, diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index da5a5cbc8759..da534a0b2133 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -214,7 +214,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, .dpcm_playback = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 804dbc7911d5..e182012d0c60 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -362,7 +362,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS + SND_SOC_DAIFMT_CBC_CFC ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -372,7 +372,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS + SND_SOC_DAIFMT_CBC_CFC ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -396,7 +396,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret < 0) { dev_err(rtd->dev, "can't set format to TDM %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 9509b6e161b8..26eb8ad0d262 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -300,7 +300,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_CBC_CFC); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 71fe26a1b701..9d75beec09d1 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -473,7 +473,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = geminilake_ssp_fixup, .dpcm_playback = 1, @@ -486,7 +486,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .no_pcm = 1, .init = geminilake_rt5682_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = geminilake_ssp_fixup, .ops = &geminilake_rt5682_ops, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c763bfeb1f38..36e136acbef5 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = haswell_ssp0_fixup, .ops = &haswell_rt5640_ops, diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 14b625e947f5..a4bdf634e9b9 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -518,7 +518,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .dpcm_playback = 1, @@ -531,7 +531,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .init = kabylake_da7219_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .dpcm_playback = 1, diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 2b43459adc33..620a9fbcb08f 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -764,7 +764,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_pmdown_time = 1, @@ -779,7 +779,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .init = kabylake_da7219_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .dpcm_playback = 1, @@ -907,7 +907,7 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_pmdown_time = 1, diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 289ca39b8206..1cb56ec363e8 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -436,7 +436,7 @@ static struct snd_soc_dai_link kabylake_rt5660_dais[] = { .exit = kabylake_rt5660_codec_exit, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp0_fixup, .ops = &kabylake_rt5660_ops, diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index a3e040a249f6..f24e0ce5d49f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -767,7 +767,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .dpcm_playback = 1, @@ -781,7 +781,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .init = kabylake_rt5663_max98927_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .ops = &kabylake_rt5663_ops, @@ -879,7 +879,7 @@ static struct snd_soc_dai_link kabylake_5663_dais[] = { .no_pcm = 1, .init = kabylake_rt5663_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .ops = &kabylake_rt5663_ops, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dd38fdaf2ff5..6874e981c8df 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -639,7 +639,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .dpcm_playback = 1, @@ -653,7 +653,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .no_pcm = 1, .init = kabylake_rt5663_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = kabylake_ssp_fixup, .ops = &kabylake_rt5663_ops, diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index e3a1f04a8b53..7297eb05613c 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -539,7 +539,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = skylake_ssp_fixup, .dpcm_playback = 1, @@ -552,7 +552,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .no_pcm = 1, .init = skylake_nau8825_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = skylake_ssp_fixup, .ops = &skylake_nau8825_ops, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index adf5992a9ec5..68efde1633b3 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -578,7 +578,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .init = skylake_ssm4567_codec_init, .ignore_pmdown_time = 1, .be_hw_params_fixup = skylake_ssp_fixup, @@ -593,7 +593,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .no_pcm = 1, .init = skylake_nau8825_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = skylake_ssp_fixup, .ops = &skylake_nau8825_ops, diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 75dab5405380..eca4a78668af 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -434,7 +434,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { .init = skylake_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = skylake_ssp0_fixup, .ops = &skylake_rt286_ops, -- 2.33.0

3 2

[PATCH 0/2] ASoC: compress: Support module_get on stream open
by Peter Ujfalusi 20 Sep '21

20 Sep '21

Hi, SOF is marking all componet drivers with module_get_upon_open = 1 which works fine with normal PCM streams, however on compressed side the module get upon open is not supported. The module_get works when module_get_upon_open is not set becasue the snd_soc_component_module_get_when_probe() will pass NULL for the substream parameter of snd_soc_component_module_get(). In order to re-use the existing infrastructure for module_get, the proposal is to convert the mark_module to void pointer (like the pm mark) and implement matching code for the compressed open/free to pcm open/close. Regards, Peter --- Peter Ujfalusi (2): ASoC: soc-component: Convert the mark_module to void* ASoC: compress/component: Use module_get_when_open/put_when_close for cstream include/sound/soc-component.h | 14 ++++---- sound/soc/soc-component.c | 61 +++++++++++++++-------------------- sound/soc/soc-compress.c | 43 +++++++++++++++++++++--- 3 files changed, 71 insertions(+), 47 deletions(-) -- 2.33.0

2 3

[PATCH] ASoC: mediatek: mt8195: force COMMON_CLK dependency
by Arnd Bergmann 20 Sep '21

20 Sep '21

From: Arnd Bergmann <arnd(a)arndb.de> Without CONFIG_COMMON_CLK, this driver fails to link: ERROR: modpost: "clk_unregister_gate" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! ERROR: modpost: "clk_register_gate" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! Add the proper Kconfig dependency for compile testing. Fixes: 6746cc858259 ("ASoC: mediatek: mt8195: add platform driver") Fixes: 940ffa194547 ("ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5a2f4667d50b..268c1f74aa3e 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -187,7 +187,7 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 config SND_SOC_MT8195 tristate "ASoC support for Mediatek MT8195 chip" - depends on ARCH_MEDIATEK || COMPILE_TEST + depends on ARCH_MEDIATEK || (COMPILE_TEST && COMMON_CLK) select SND_SOC_MEDIATEK help This adds ASoC platform driver support for Mediatek MT8195 chip -- 2.29.2

3 3

[PATCH v4 0/4] Rockchip I2S/TDM controller
by Nicolas Frattaroli 20 Sep '21

20 Sep '21

Hello, this is version 4 of the I2S/TDM driver patchset. A big thanks to everyone who has provided their valuable feedback so far. Changes in v4: driver: - factor TDE/RDE enable/disable into their own inlined functions - add an RDE disable in a location where it looks like it was forgotten (rxctrl else), judging by corresponding TDE code - remove parentheses around CLK_PPM_MIN/MAX values - wording + titlecasing in the clock compensation control - use if statement in precious_reg instead - refactor rockchip_i2s_io_multiplex to have the switch statements in a function call to make the function less unwieldy - get rid of IS_ERR checks around clk enable/disable calls where already checked before by the probe - reworded some error message strings - fix potential deadlock in txrxctrl found by Sugar Zhang using spin_lock_irqsave - fix potential deadlock in trcm_mode found by Sugar Zhang using spin_lock_irqsave - use devm_platform_get_and_ioremap_resource in probe - only set DMA things if controller has capture/playback ability. Did not move this into init_dai because I'd then need to pass in the res and probe it earlier in the function, and it's also used elsewhere in the probe function - use _get_optional_exclusive for reset controls, as some controllers only have capture or playback capability bindings: - remove status = "okay" since that's the default - change the path configs to be an enum - rename "foo" to "bus" - make resets optional as controller may lack either playback or capture capability, and therefore also doesn't have a reset. At least one reset is still required, because a controller with no playback and no capture is not very useful Changes in v3: driver: - alphabetically sort includes - check pm_runtime_get_sync return value, act on it - remove unnecessary initialisers in set_fmt - use udelay(15) in retry code: 10 retries * 15 = 150, so at worst we wait the full i2s register access delay - fix some weird returns to return directly - use __maybe_unused instead of #ifdef CONFIG_PM_SLEEP, also put __maybe_unused on the runtime callbacks - use (foo) instead of foo in header macros for precedence reasons - when using mclk-calibrate, also turn off/on those clocks during suspend and resume operations - remove mclk_tx and mclk_rx reenablement code in remove - move hclk enablement further down the probe, and disable it on probe failure - make reset controls mandatory, since the bindings state this too - use _exclusive for getting the reset controls - change reset assert/deassert delays to both be 10 usec (thank you Sugar Zhang!) - properly prepare and enable all mclks in probe, especially before calling clk_get_rate on them - if registering PCM fails, also use the cleanup error path instead of returning directly - bring back playback and capture only but in the way Sugar Zhang suggested it: set those modes depending on dma-names - rework clock enablement in general. Probe now always enables these, instead of relying on the pm resume thing - add myself to MAINTAINERS for this driver dt bindings: - fix a description still mentioning clk-trcm in the schema - document rockchip,io-multiplex, a property that describes the hardware as having multiplexed I2S GPIOs so direction needs to be changed dynamically - document rockchip,mclk-calibrate, which allows specifying different clocks for the two sample rate bases and switch between them as needed - dma-names now doesn't have a set order and items can be absent to indicate that the controller doesn't support this mode - add myself to MAINTAINERS for these bindings Changes in v2: - remove ad-hoc writeq and needless (and broken) optimisation in reset assert/deassert. This wouldn't have worked on Big Endian, and would've been pointless on any other platform, as the overhead for saving one write was comparatively big - fix various checkpatch issues - get rid of leftover clk-trcm in schema - set status = "okay" in example in schema instead of "disabled" - change dma-names so rx is first, adjust device trees as necessary - properly reference uint32-array for rx-route and tx-route instead of uint32 - replace trcm-sync with two boolean properties, adjust DT changes accordingly and also get rid of the header file - get rid of rockchip,no-dmaengine. This was only needed for some downstream driver and shouldn't be in the DT - get rid of rockchip,capture-only/playback-only. Rationale being that I have no way to test whether they're needed, and unconditionally setting channels_min to 0 breaks everything - change hclk description in "clocks" Nicolas Frattaroli (4): ASoC: rockchip: add support for i2s-tdm controller dt-bindings: sound: add rockchip i2s-tdm binding arm64: dts: rockchip: add i2s1 on rk356x arm64: dts: rockchip: add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 218 ++ MAINTAINERS | 7 + .../boot/dts/rockchip/rk3566-quartz64-a.dts | 35 +- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 26 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1832 +++++++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 398 ++++ 8 files changed, 2528 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h -- 2.33.0

3 10

[PATCH RESEND v2 1/2] ASoC: rt5682s: Add driver for ALC5682I-VS codec
by derek.fang＠realtek.com 20 Sep '21

20 Sep '21

From: Derek Fang <derek.fang(a)realtek.com> This is an initial codec driver for Realtek ALC5682I-VS codec. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- include/sound/rt5682s.h | 48 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682s.c | 3188 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5682s.h | 1455 ++++++++++++++++ 5 files changed, 4698 insertions(+) create mode 100644 include/sound/rt5682s.h create mode 100644 sound/soc/codecs/rt5682s.c create mode 100644 sound/soc/codecs/rt5682s.h diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h new file mode 100644 index 000000000000..accfbc2dcdd2 --- /dev/null +++ b/include/sound/rt5682s.h @@ -0,0 +1,48 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt5682s.h -- Platform data for RT5682I-VS + * + * Copyright 2021 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT5682S_H +#define __LINUX_SND_RT5682S_H + +enum rt5682s_dmic1_data_pin { + RT5682S_DMIC1_DATA_NULL, + RT5682S_DMIC1_DATA_GPIO2, + RT5682S_DMIC1_DATA_GPIO5, +}; + +enum rt5682s_dmic1_clk_pin { + RT5682S_DMIC1_CLK_NULL, + RT5682S_DMIC1_CLK_GPIO1, + RT5682S_DMIC1_CLK_GPIO3, +}; + +enum rt5682s_jd_src { + RT5682S_JD_NULL, + RT5682S_JD1, +}; + +enum rt5682s_dai_clks { + RT5682S_DAI_WCLK_IDX, + RT5682S_DAI_BCLK_IDX, + RT5682S_DAI_NUM_CLKS, +}; + +struct rt5682s_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5682s_dmic1_data_pin dmic1_data_pin; + enum rt5682s_dmic1_clk_pin dmic1_clk_pin; + enum rt5682s_jd_src jd_src; + unsigned int dmic_clk_rate; + unsigned int dmic_delay; + bool dmic_clk_driving_high; + + const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..6b3c2cf153db 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,6 +180,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT5677 imply SND_SOC_RT5682_I2C imply SND_SOC_RT5682_SDW + imply SND_SOC_RT5682S imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW imply SND_SOC_RT711_SDCA_SDW @@ -1249,6 +1250,10 @@ config SND_SOC_RT5682_SDW select SND_SOC_RT5682 select REGMAP_SOUNDWIRE +config SND_SOC_RT5682S + tristate + depends on I2C + config SND_SOC_RT700 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..42d00aa4ee46 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -198,6 +198,7 @@ snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt5682-i2c-objs := rt5682-i2c.o +snd-soc-rt5682s-objs := rt5682s.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o @@ -526,6 +527,7 @@ obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_RT5682_I2C) += snd-soc-rt5682-i2c.o obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o +obj-$(CONFIG_SND_SOC_RT5682S) += snd-soc-rt5682s.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT711_SDCA_SDW) += snd-soc-rt711-sdca.o diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c new file mode 100644 index 000000000000..d878a20527f1 --- /dev/null +++ b/sound/soc/codecs/rt5682s.c @@ -0,0 +1,3188 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt5682s.c -- RT5682I-VS ALSA SoC audio component driver +// +// Copyright 2021 Realtek Semiconductor Corp. +// Author: Derek Fang <derek.fang(a)realtek.com> +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5682s.h> + +#include "rt5682s.h" + +#define DEVICE_ID 0x6749 + +static const struct rt5682s_platform_data i2s_default_platform_data = { + .dmic1_data_pin = RT5682S_DMIC1_DATA_GPIO2, + .dmic1_clk_pin = RT5682S_DMIC1_CLK_GPIO3, + .jd_src = RT5682S_JD1, + .dai_clk_names[RT5682S_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682S_DAI_BCLK_IDX] = "rt5682-dai-bclk", +}; + +const char *rt5682s_supply_names[RT5682S_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", +}; + +static const struct reg_sequence patch_list[] = { + {RT5682S_I2C_CTRL, 0x0007}, + {RT5682S_DIG_IN_CTRL_1, 0x0000}, + {RT5682S_CHOP_DAC_2, 0x2020}, + {RT5682S_VREF_REC_OP_FB_CAP_CTRL_2, 0x0101}, + {RT5682S_VREF_REC_OP_FB_CAP_CTRL_1, 0x80c0}, + {RT5682S_HP_CALIB_CTRL_9, 0x0002}, + {RT5682S_DEPOP_1, 0x0000}, + {RT5682S_HP_CHARGE_PUMP_2, 0x3c15}, + {RT5682S_DAC1_DIG_VOL, 0xfefe}, + {RT5682S_SAR_IL_CMD_2, 0xac00}, + {RT5682S_SAR_IL_CMD_3, 0x024c}, + {RT5682S_CBJ_CTRL_6, 0x0804}, +}; + +static void rt5682s_apply_patch_list(struct rt5682s_priv *rt5682s, + struct device *dev) +{ + int ret; + + ret = regmap_multi_reg_write(rt5682s->regmap, patch_list, ARRAY_SIZE(patch_list)); + if (ret) + dev_warn(dev, "Failed to apply regmap patch: %d\n", ret); +} + +const struct reg_default rt5682s_reg[] = { + {0x0000, 0x0001}, + {0x0002, 0x8080}, + {0x0003, 0x0001}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x8007}, + {0x000b, 0x0000}, + {0x000f, 0x4000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x0000}, + {0x0013, 0x1200}, + {0x0014, 0x200a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0017, 0x05a4}, + {0x0019, 0xffff}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x08c0}, + {0x0044, 0x1818}, + {0x004b, 0x00c0}, + {0x004c, 0x0000}, + {0x004d, 0x0000}, + {0x0061, 0x00c0}, + {0x0062, 0x008a}, + {0x0063, 0x0800}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x000c}, + {0x0068, 0x0000}, + {0x0069, 0x0000}, + {0x006a, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0810}, + {0x006f, 0xe4de}, + {0x0070, 0x3320}, + {0x0071, 0x0000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x007f, 0x0000}, + {0x0080, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008e, 0x0060}, + {0x008f, 0x4da1}, + {0x0091, 0x1c15}, + {0x0092, 0x0425}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0095, 0x008f}, + {0x0096, 0x0000}, + {0x0097, 0x0000}, + {0x0098, 0x0000}, + {0x0099, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x0000}, + {0x009f, 0x0009}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0000}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6749}, + {0x0100, 0xa000}, + {0x010b, 0x0066}, + {0x010c, 0x6666}, + {0x010d, 0x2202}, + {0x010e, 0x6666}, + {0x010f, 0xa800}, + {0x0110, 0x0006}, + {0x0111, 0x0460}, + {0x0112, 0x2000}, + {0x0113, 0x0200}, + {0x0117, 0x8000}, + {0x0118, 0x0303}, + {0x0125, 0x0020}, + {0x0132, 0x5026}, + {0x0136, 0x8000}, + {0x0139, 0x0005}, + {0x013a, 0x3030}, + {0x013b, 0xa000}, + {0x013c, 0x4110}, + {0x013f, 0x0000}, + {0x0145, 0x0022}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0156, 0x0022}, + {0x0157, 0x0303}, + {0x0158, 0x2222}, + {0x0159, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0001}, + {0x0190, 0x4131}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01ac, 0x0000}, + {0x01ad, 0x0000}, + {0x01ae, 0x0000}, + {0x01af, 0x2000}, + {0x01b0, 0x0000}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0017}, + {0x01b4, 0x004b}, + {0x01b5, 0x0000}, + {0x01b6, 0x03e8}, + {0x01b7, 0x0000}, + {0x01b8, 0x0000}, + {0x01b9, 0x0400}, + {0x01ba, 0xb5b6}, + {0x01bb, 0x9124}, + {0x01bc, 0x4924}, + {0x01bd, 0x0009}, + {0x01be, 0x0018}, + {0x01bf, 0x002a}, + {0x01c0, 0x004c}, + {0x01c1, 0x0097}, + {0x01c2, 0x01c3}, + {0x01c3, 0x03e9}, + {0x01c4, 0x1389}, + {0x01c5, 0xc351}, + {0x01c6, 0x02a0}, + {0x01c7, 0x0b0f}, + {0x01c8, 0x402f}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0000}, + {0x01d8, 0x0162}, + {0x01d9, 0x0007}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0020}, + {0x01e0, 0x04c1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e5, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa004}, + {0x0212, 0x0365}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x021d, 0x024c}, + {0x02fa, 0x0000}, + {0x02fb, 0x0000}, + {0x02fc, 0x0000}, + {0x03fe, 0x0000}, + {0x03ff, 0x0000}, + {0x0500, 0x0000}, + {0x0600, 0x0000}, + {0x0610, 0x6666}, + {0x0611, 0xa9aa}, + {0x0620, 0x6666}, + {0x0621, 0xa9aa}, + {0x0630, 0x6666}, + {0x0631, 0xa9aa}, + {0x0640, 0x6666}, + {0x0641, 0xa9aa}, + {0x07fa, 0x0000}, + {0x08fa, 0x0000}, + {0x08fb, 0x0000}, + {0x0d00, 0x0000}, + {0x1100, 0x0000}, + {0x1101, 0x0000}, + {0x1102, 0x0000}, + {0x1103, 0x0000}, + {0x1104, 0x0000}, + {0x1105, 0x0000}, + {0x1106, 0x0000}, + {0x1107, 0x0000}, + {0x1108, 0x0000}, + {0x1109, 0x0000}, + {0x110a, 0x0000}, + {0x110b, 0x0000}, + {0x110c, 0x0000}, + {0x1111, 0x0000}, + {0x1112, 0x0000}, + {0x1113, 0x0000}, + {0x1114, 0x0000}, + {0x1115, 0x0000}, + {0x1116, 0x0000}, + {0x1117, 0x0000}, + {0x1118, 0x0000}, + {0x1119, 0x0000}, + {0x111a, 0x0000}, + {0x111b, 0x0000}, + {0x111c, 0x0000}, + {0x1401, 0x0404}, + {0x1402, 0x0007}, + {0x1403, 0x0365}, + {0x1404, 0x0210}, + {0x1405, 0x0365}, + {0x1406, 0x0210}, + {0x1407, 0x0000}, + {0x1408, 0x0000}, + {0x1409, 0x0000}, + {0x140a, 0x0000}, + {0x140b, 0x0000}, + {0x140c, 0x0000}, + {0x140d, 0x0000}, + {0x140e, 0x0000}, + {0x140f, 0x0000}, + {0x1410, 0x0000}, + {0x1411, 0x0000}, + {0x1801, 0x0004}, + {0x1802, 0x0000}, + {0x1803, 0x0000}, + {0x1804, 0x0000}, + {0x1805, 0x00ff}, + {0x2c00, 0x0000}, + {0x3400, 0x0200}, + {0x3404, 0x0000}, + {0x3405, 0x0000}, + {0x3406, 0x0000}, + {0x3407, 0x0000}, + {0x3408, 0x0000}, + {0x3409, 0x0000}, + {0x340a, 0x0000}, + {0x340b, 0x0000}, + {0x340c, 0x0000}, + {0x340d, 0x0000}, + {0x340e, 0x0000}, + {0x340f, 0x0000}, + {0x3410, 0x0000}, + {0x3411, 0x0000}, + {0x3412, 0x0000}, + {0x3413, 0x0000}, + {0x3414, 0x0000}, + {0x3415, 0x0000}, + {0x3424, 0x0000}, + {0x3425, 0x0000}, + {0x3426, 0x0000}, + {0x3427, 0x0000}, + {0x3428, 0x0000}, + {0x3429, 0x0000}, + {0x342a, 0x0000}, + {0x342b, 0x0000}, + {0x342c, 0x0000}, + {0x342d, 0x0000}, + {0x342e, 0x0000}, + {0x342f, 0x0000}, + {0x3430, 0x0000}, + {0x3431, 0x0000}, + {0x3432, 0x0000}, + {0x3433, 0x0000}, + {0x3434, 0x0000}, + {0x3435, 0x0000}, + {0x3440, 0x6319}, + {0x3441, 0x3771}, + {0x3500, 0x0002}, + {0x3501, 0x5728}, + {0x3b00, 0x3010}, + {0x3b01, 0x3300}, + {0x3b02, 0x2200}, + {0x3b03, 0x0100}, +}; + +static bool rt5682s_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682S_RESET: + case RT5682S_CBJ_CTRL_2: + case RT5682S_I2S1_F_DIV_CTRL_2: + case RT5682S_I2S2_F_DIV_CTRL_2: + case RT5682S_INT_ST_1: + case RT5682S_GPIO_ST: + case RT5682S_IL_CMD_1: + case RT5682S_4BTN_IL_CMD_1: + case RT5682S_AJD1_CTRL: + case RT5682S_VERSION_ID...RT5682S_DEVICE_ID: + case RT5682S_STO_NG2_CTRL_1: + case RT5682S_STO_NG2_CTRL_5...RT5682S_STO_NG2_CTRL_7: + case RT5682S_STO1_DAC_SIL_DET: + case RT5682S_HP_IMP_SENS_CTRL_1...RT5682S_HP_IMP_SENS_CTRL_4: + case RT5682S_HP_IMP_SENS_CTRL_13: + case RT5682S_HP_IMP_SENS_CTRL_14: + case RT5682S_HP_IMP_SENS_CTRL_43...RT5682S_HP_IMP_SENS_CTRL_46: + case RT5682S_HP_CALIB_CTRL_1: + case RT5682S_HP_CALIB_CTRL_10: + case RT5682S_HP_CALIB_ST_1...RT5682S_HP_CALIB_ST_11: + case RT5682S_SAR_IL_CMD_2...RT5682S_SAR_IL_CMD_5: + case RT5682S_SAR_IL_CMD_10: + case RT5682S_SAR_IL_CMD_11: + case RT5682S_VERSION_ID_HIDE: + case RT5682S_VERSION_ID_CUS: + case RT5682S_I2C_TRANS_CTRL: + case RT5682S_DMIC_FLOAT_DET: + case RT5682S_HA_CMP_OP_1: + case RT5682S_NEW_CBJ_DET_CTL_10...RT5682S_NEW_CBJ_DET_CTL_16: + case RT5682S_CLK_SW_TEST_1: + case RT5682S_CLK_SW_TEST_2: + case RT5682S_EFUSE_READ_1...RT5682S_EFUSE_READ_18: + case RT5682S_PILOT_DIG_CTL_1: + return true; + default: + return false; + } +} + +static bool rt5682s_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682S_RESET: + case RT5682S_VERSION_ID: + case RT5682S_VENDOR_ID: + case RT5682S_DEVICE_ID: + case RT5682S_HP_CTRL_1: + case RT5682S_HP_CTRL_2: + case RT5682S_HPL_GAIN: + case RT5682S_HPR_GAIN: + case RT5682S_I2C_CTRL: + case RT5682S_CBJ_BST_CTRL: + case RT5682S_CBJ_DET_CTRL: + case RT5682S_CBJ_CTRL_1...RT5682S_CBJ_CTRL_8: + case RT5682S_DAC1_DIG_VOL: + case RT5682S_STO1_ADC_DIG_VOL: + case RT5682S_STO1_ADC_BOOST: + case RT5682S_HP_IMP_GAIN_1: + case RT5682S_HP_IMP_GAIN_2: + case RT5682S_SIDETONE_CTRL: + case RT5682S_STO1_ADC_MIXER: + case RT5682S_AD_DA_MIXER: + case RT5682S_STO1_DAC_MIXER: + case RT5682S_A_DAC1_MUX: + case RT5682S_DIG_INF2_DATA: + case RT5682S_REC_MIXER: + case RT5682S_CAL_REC: + case RT5682S_HP_ANA_OST_CTRL_1...RT5682S_HP_ANA_OST_CTRL_3: + case RT5682S_PWR_DIG_1...RT5682S_PWR_MIXER: + case RT5682S_MB_CTRL: + case RT5682S_CLK_GATE_TCON_1...RT5682S_CLK_GATE_TCON_3: + case RT5682S_CLK_DET...RT5682S_LPF_AD_DMIC: + case RT5682S_I2S1_SDP: + case RT5682S_I2S2_SDP: + case RT5682S_ADDA_CLK_1: + case RT5682S_ADDA_CLK_2: + case RT5682S_I2S1_F_DIV_CTRL_1: + case RT5682S_I2S1_F_DIV_CTRL_2: + case RT5682S_TDM_CTRL: + case RT5682S_TDM_ADDA_CTRL_1: + case RT5682S_TDM_ADDA_CTRL_2: + case RT5682S_DATA_SEL_CTRL_1: + case RT5682S_TDM_TCON_CTRL_1: + case RT5682S_TDM_TCON_CTRL_2: + case RT5682S_GLB_CLK: + case RT5682S_PLL_TRACK_1...RT5682S_PLL_TRACK_6: + case RT5682S_PLL_TRACK_11: + case RT5682S_DEPOP_1: + case RT5682S_HP_CHARGE_PUMP_1: + case RT5682S_HP_CHARGE_PUMP_2: + case RT5682S_HP_CHARGE_PUMP_3: + case RT5682S_MICBIAS_1...RT5682S_MICBIAS_3: + case RT5682S_PLL_TRACK_12...RT5682S_PLL_CTRL_7: + case RT5682S_RC_CLK_CTRL: + case RT5682S_I2S2_M_CLK_CTRL_1: + case RT5682S_I2S2_F_DIV_CTRL_1: + case RT5682S_I2S2_F_DIV_CTRL_2: + case RT5682S_IRQ_CTRL_1...RT5682S_IRQ_CTRL_4: + case RT5682S_INT_ST_1: + case RT5682S_GPIO_CTRL_1: + case RT5682S_GPIO_CTRL_2: + case RT5682S_GPIO_ST: + case RT5682S_HP_AMP_DET_CTRL_1: + case RT5682S_MID_HP_AMP_DET: + case RT5682S_LOW_HP_AMP_DET: + case RT5682S_DELAY_BUF_CTRL: + case RT5682S_SV_ZCD_1: + case RT5682S_SV_ZCD_2: + case RT5682S_IL_CMD_1...RT5682S_IL_CMD_6: + case RT5682S_4BTN_IL_CMD_1...RT5682S_4BTN_IL_CMD_7: + case RT5682S_ADC_STO1_HP_CTRL_1: + case RT5682S_ADC_STO1_HP_CTRL_2: + case RT5682S_AJD1_CTRL: + case RT5682S_JD_CTRL_1: + case RT5682S_DUMMY_1...RT5682S_DUMMY_3: + case RT5682S_DAC_ADC_DIG_VOL1: + case RT5682S_BIAS_CUR_CTRL_2...RT5682S_BIAS_CUR_CTRL_10: + case RT5682S_VREF_REC_OP_FB_CAP_CTRL_1: + case RT5682S_VREF_REC_OP_FB_CAP_CTRL_2: + case RT5682S_CHARGE_PUMP_1: + case RT5682S_DIG_IN_CTRL_1: + case RT5682S_PAD_DRIVING_CTRL: + case RT5682S_CHOP_DAC_1: + case RT5682S_CHOP_DAC_2: + case RT5682S_CHOP_ADC: + case RT5682S_CALIB_ADC_CTRL: + case RT5682S_VOL_TEST: + case RT5682S_SPKVDD_DET_ST: + case RT5682S_TEST_MODE_CTRL_1...RT5682S_TEST_MODE_CTRL_4: + case RT5682S_PLL_INTERNAL_1...RT5682S_PLL_INTERNAL_4: + case RT5682S_STO_NG2_CTRL_1...RT5682S_STO_NG2_CTRL_10: + case RT5682S_STO1_DAC_SIL_DET: + case RT5682S_SIL_PSV_CTRL1: + case RT5682S_SIL_PSV_CTRL2: + case RT5682S_SIL_PSV_CTRL3: + case RT5682S_SIL_PSV_CTRL4: + case RT5682S_SIL_PSV_CTRL5: + case RT5682S_HP_IMP_SENS_CTRL_1...RT5682S_HP_IMP_SENS_CTRL_46: + case RT5682S_HP_LOGIC_CTRL_1...RT5682S_HP_LOGIC_CTRL_3: + case RT5682S_HP_CALIB_CTRL_1...RT5682S_HP_CALIB_CTRL_11: + case RT5682S_HP_CALIB_ST_1...RT5682S_HP_CALIB_ST_11: + case RT5682S_SAR_IL_CMD_1...RT5682S_SAR_IL_CMD_14: + case RT5682S_DUMMY_4...RT5682S_DUMMY_6: + case RT5682S_VERSION_ID_HIDE: + case RT5682S_VERSION_ID_CUS: + case RT5682S_SCAN_CTL: + case RT5682S_HP_AMP_DET: + case RT5682S_BIAS_CUR_CTRL_11: + case RT5682S_BIAS_CUR_CTRL_12: + case RT5682S_BIAS_CUR_CTRL_13: + case RT5682S_BIAS_CUR_CTRL_14: + case RT5682S_BIAS_CUR_CTRL_15: + case RT5682S_BIAS_CUR_CTRL_16: + case RT5682S_BIAS_CUR_CTRL_17: + case RT5682S_BIAS_CUR_CTRL_18: + case RT5682S_I2C_TRANS_CTRL: + case RT5682S_DUMMY_7: + case RT5682S_DUMMY_8: + case RT5682S_DMIC_FLOAT_DET: + case RT5682S_HA_CMP_OP_1...RT5682S_HA_CMP_OP_13: + case RT5682S_HA_CMP_OP_14...RT5682S_HA_CMP_OP_25: + case RT5682S_NEW_CBJ_DET_CTL_1...RT5682S_NEW_CBJ_DET_CTL_16: + case RT5682S_DA_FILTER_1...RT5682S_DA_FILTER_5: + case RT5682S_CLK_SW_TEST_1: + case RT5682S_CLK_SW_TEST_2: + case RT5682S_CLK_SW_TEST_3...RT5682S_CLK_SW_TEST_14: + case RT5682S_EFUSE_MANU_WRITE_1...RT5682S_EFUSE_MANU_WRITE_6: + case RT5682S_EFUSE_READ_1...RT5682S_EFUSE_READ_18: + case RT5682S_EFUSE_TIMING_CTL_1: + case RT5682S_EFUSE_TIMING_CTL_2: + case RT5682S_PILOT_DIG_CTL_1: + case RT5682S_PILOT_DIG_CTL_2: + case RT5682S_HP_AMP_DET_CTL_1...RT5682S_HP_AMP_DET_CTL_4: + return true; + default: + return false; + } +} + +static void rt5682s_reset(struct rt5682s_priv *rt5682s) +{ + regmap_write(rt5682s->regmap, RT5682S_RESET, 0); +} + +static int rt5682s_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read(component, RT5682S_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682S_4BTN_IL_CMD_1, val); + dev_dbg(component->dev, "%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_2, + RT5682S_SAR_ADC_PSV_MASK, RT5682S_SAR_ADC_PSV_ENTRY); + + return btn_type; +} + +enum { + SAR_PWR_OFF, + SAR_PWR_NORMAL, + SAR_PWR_SAVING, +}; + +static void rt5682s_sar_power_mode(struct snd_soc_component *component, + int mode, int jd_step) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + mutex_lock(&rt5682s->sar_mutex); + + switch (mode) { + case SAR_PWR_SAVING: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_REG | RT5682S_CTRL_MB2_REG); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK, RT5682S_SAR_BUTDET_EN); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_2, + RT5682S_SAR_ADC_PSV_MASK, RT5682S_SAR_ADC_PSV_ENTRY); + break; + case SAR_PWR_NORMAL: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_FSM | RT5682S_CTRL_MB2_FSM); + if (!jd_step) { + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_SEL_MB1_2_AUTO); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK, + RT5682S_SAR_BUTDET_EN | RT5682S_SAR_BUTDET_POW_NORM); + } + break; + case SAR_PWR_OFF: + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + break; + default: + dev_err(component->dev, "Invalid SAR Power mode: %d\n", mode); + break; + } + + mutex_unlock(&rt5682s->sar_mutex); +} + +static void rt5682s_enable_push_button_irq(struct snd_soc_component *component) +{ + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682S_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK | RT5682S_4BTN_IL_RST_MASK, + RT5682S_4BTN_IL_EN | RT5682S_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682S_IRQ_CTRL_3, + RT5682S_IL_IRQ_MASK, RT5682S_IL_IRQ_EN); +} + +static void rt5682s_disable_push_button_irq(struct snd_soc_component *component) +{ + snd_soc_component_update_bits(component, RT5682S_IRQ_CTRL_3, + RT5682S_IL_IRQ_MASK, RT5682S_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); +} + +/** + * rt5682s_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_insert) +{ + unsigned int val, count; + int jack_type = 0; + + if (jack_insert) { + rt5682s_disable_push_button_irq(component); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF1 | RT5682S_PWR_VREF2 | RT5682S_PWR_MB, + RT5682S_PWR_VREF1 | RT5682S_PWR_VREF2 | RT5682S_PWR_MB); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_3, + RT5682S_PWR_CBJ, RT5682S_PWR_CBJ); + snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x0365); + snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, + RT5682S_OSW_L_DIS | RT5682S_OSW_R_DIS); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 1); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); + usleep_range(45000, 50000); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_HIGH); + + count = 0; + do { + usleep_range(10000, 15000); + val = snd_soc_component_read(component, RT5682S_CBJ_CTRL_2) + & RT5682S_JACK_TYPE_MASK; + count++; + } while (val == 0 && count < 50); + + pr_debug("%s, val=%d, count=%d\n", __func__, val, count); + + switch (val) { + case 0x1: + case 0x2: + jack_type = SND_JACK_HEADSET; + snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x024c); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_EN); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_MASK, val << RT5682S_SAR_SEL_MB1_2_SFT); + rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 1); + rt5682s_enable_push_button_irq(component); + break; + default: + jack_type = SND_JACK_HEADPHONE; + break; + } + snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, + RT5682S_OSW_L_EN | RT5682S_OSW_R_EN); + } else { + rt5682s_sar_power_mode(component, SAR_PWR_OFF, 1); + rt5682s_disable_push_button_irq(component); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); + + if (!snd_soc_dapm_get_pin_status(&component->dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(&component->dapm, "Vref2")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_VREF2, 0); + + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_3, + RT5682S_PWR_CBJ, 0); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_DIS); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", jack_type); + + return jack_type; +} + +static void rt5682s_jack_detect_handler(struct work_struct *work) +{ + struct rt5682s_priv *rt5682s = + container_of(work, struct rt5682s_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682s->component) + usleep_range(10000, 15000); + + while (!rt5682s->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682s->calibrate_mutex); + + val = snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) + & RT5682S_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682s->jack_type == 0) { + /* jack was out, report jack type */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 1); + rt5682s->irq_work_delay_time = 0; + } else if ((rt5682s->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + rt5682s->jack_type = SND_JACK_HEADSET; + btn_type = rt5682s_button_detect(rt5682s->component); + /** + * rt5682s can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682s->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682s->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682s->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682s->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5682s->component->dev, + "Unexpected button code 0x%04x\n", btn_type); + break; + } + } + } else { + /* jack out */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); + rt5682s->irq_work_delay_time = 50; + } + + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682s->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682s->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + mutex_unlock(&rt5682s->calibrate_mutex); +} + +static void rt5682s_jd_check_handler(struct work_struct *work) +{ + struct rt5682s_priv *rt5682s = + container_of(work, struct rt5682s_priv, jd_check_work.work); + + if (snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) + & RT5682S_JDH_RS_MASK) { + /* jack out */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); + + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682s->jd_check_work, 500); + } +} + +static irqreturn_t rt5682s_irq(int irq, void *data) +{ + struct rt5682s_priv *rt5682s = data; + + mod_delayed_work(system_power_efficient_wq, &rt5682s->jack_detect_work, + msecs_to_jiffies(rt5682s->irq_work_delay_time)); + + return IRQ_HANDLED; +} + +static int rt5682s_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int btndet_delay = 16; + + rt5682s->hs_jack = hs_jack; + + if (!hs_jack) { + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK, RT5682S_JD1_DIS); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_JDH, 0); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + + return 0; + } + + switch (rt5682s->pdata.jd_src) { + case RT5682S_JD1: + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_5, + RT5682S_JD_FAST_OFF_SRC_MASK, RT5682S_JD_FAST_OFF_SRC_JDH); + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_2, + RT5682S_EXT_JD_SRC, RT5682S_EXT_JD_SRC_MANUAL); + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_1, + RT5682S_EMB_JD_MASK | RT5682S_DET_TYPE | + RT5682S_POL_FAST_OFF_MASK | RT5682S_MIC_CAP_MASK, + RT5682S_EMB_JD_EN | RT5682S_DET_TYPE | + RT5682S_POL_FAST_OFF_HIGH | RT5682S_MIC_CAP_HS); + regmap_update_bits(rt5682s->regmap, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_POW_MASK, RT5682S_SAR_POW_EN); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP1_PIN_MASK, RT5682S_GP1_PIN_IRQ); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BGLDO, RT5682S_PWR_BGLDO); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_2, + RT5682S_PWR_JD_MASK, RT5682S_PWR_JD_ENABLE); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_IRQ | RT5682S_POW_JDH, RT5682S_POW_IRQ | RT5682S_POW_JDH); + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK | RT5682S_JD1_POL_MASK, + RT5682S_JD1_EN | RT5682S_JD1_POL_NOR); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_4, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_5, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_6, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_7, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682S_JD_NULL: + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK, RT5682S_JD1_DIS); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_JDH, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); + +static const struct snd_kcontrol_new rt5682s_snd_controls[] = { + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, + RT5682S_L_VOL_SFT + 2, RT5682S_R_VOL_SFT + 2, 63, 0, dac_vol_tlv), + + /* CBJ Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682S_REC_MIXER, + RT5682S_BST_CBJ_SFT, 35, 0, cbj_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE_SFT, RT5682S_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_VOL_SFT + 1, RT5682S_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682S_STO1_ADC_BOOST, + RT5682S_STO1_ADC_L_BST_SFT, RT5682S_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), +}; + +/** + * rt5682s_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682S can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + switch (clk_src) { + case RT5682S_CLK_SEL_SYS: + case RT5682S_CLK_SEL_I2S1_ASRC: + case RT5682S_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682S_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_2, + RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682S_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_3, + RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682s_sel_asrc_clk_src); + +static int rt5682s_div_sel(struct rt5682s_priv *rt5682s, + int target, const int div[], int size) +{ + int i; + + if (rt5682s->sysclk < target) { + dev_err(rt5682s->component->dev, + "sysclk rate %d is too low\n", rt5682s->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + dev_dbg(rt5682s->component->dev, "div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682s->sysclk) + return i; + if (target * div[i + 1] > rt5682s->sysclk) { + dev_dbg(rt5682s->component->dev, + "can't find div for sysclk %d\n", rt5682s->sysclk); + return i; + } + } + + if (target * div[i] < rt5682s->sysclk) + dev_err(rt5682s->component->dev, + "sysclk rate %d is too high\n", rt5682s->sysclk); + + return size - 1; +} + +static int get_clk_info(int sclk, int rate) +{ + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int idx, dmic_clk_rate = 3072000; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + if (rt5682s->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682s->pdata.dmic_clk_rate; + + idx = rt5682s_div_sel(rt5682s, dmic_clk_rate, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_CLK_MASK, idx << RT5682S_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int ref, val, reg, idx; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read(component, RT5682S_GPIO_CTRL_1) + & RT5682S_GP4_PIN_MASK; + + if (w->shift == RT5682S_PWR_ADC_S1F_BIT && val == RT5682S_GP4_PIN_ADCDAT2) + ref = 256 * rt5682s->lrck[RT5682S_AIF2]; + else + ref = 256 * rt5682s->lrck[RT5682S_AIF1]; + + idx = rt5682s_div_sel(rt5682s, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682S_PWR_ADC_S1F_BIT) + reg = RT5682S_PLL_TRACK_3; + else + reg = RT5682S_PLL_TRACK_2; + + snd_soc_component_update_bits(component, reg, + RT5682S_FILTER_CLK_DIV_MASK, idx << RT5682S_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682s->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_1, + RT5682S_ADC_OSR_MASK | RT5682S_DAC_OSR_MASK, + (idx << RT5682S_ADC_OSR_SFT) | (idx << RT5682S_DAC_OSR_SFT)); + + return 0; +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int delay = 50, val; + + if (rt5682s->pdata.dmic_delay) + delay = rt5682s->pdata.dmic_delay; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = (snd_soc_component_read(component, RT5682S_GLB_CLK) + & RT5682S_SCLK_SRC_MASK) >> RT5682S_SCLK_SRC_SFT; + if (val == RT5682S_CLK_SRC_PLL1 || val == RT5682S_CLK_SRC_PLL2) + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_MB, + RT5682S_PWR_VREF2 | RT5682S_PWR_MB); + + /*Add delay to avoid pop noise*/ + msleep(delay); + break; + + case SND_SOC_DAPM_POST_PMD: + if (!rt5682s->jack_type) { + if (!snd_soc_dapm_get_pin_status(w->dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(w->dapm, "Vref2")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_VREF2, 0); + } + break; + } + + return 0; +} + +static int set_i2s_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int pre_div, id; + unsigned int reg, mask, sft; + + if (event != SND_SOC_DAPM_PRE_PMU) + return 0; + + if (w->shift == RT5682S_PWR_I2S2_BIT) { + id = RT5682S_AIF2; + reg = RT5682S_I2S2_M_CLK_CTRL_1; + mask = RT5682S_I2S2_M_D_MASK; + sft = RT5682S_I2S2_M_D_SFT; + } else { + id = RT5682S_AIF1; + reg = RT5682S_ADDA_CLK_1; + mask = RT5682S_I2S_M_D_MASK; + sft = RT5682S_I2S_M_D_SFT; + } + + if (!rt5682s->master[id]) + return 0; + + pre_div = get_clk_info(rt5682s->sysclk, rt5682s->lrck[id]); + if (pre_div < 0) { + dev_err(component->dev, "get pre_div failed\n"); + return -EINVAL; + } + + dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d master\n", + rt5682s->lrck[id], pre_div, id); + snd_soc_component_update_bits(component, reg, mask, pre_div << sft); + + return 0; +} + +static int is_sys_clk_from_plla(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if ((rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL1) || + (rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL2 && rt5682s->pll_comb == USE_PLLAB)) + return 1; + + return 0; +} + +static int is_sys_clk_from_pllb(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if (rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL2) + return 1; + + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, sft, val; + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682S_ADC_STO1_ASRC_SFT: + reg = RT5682S_PLL_TRACK_3; + sft = RT5682S_FILTER_CLK_SEL_SFT; + break; + case RT5682S_DAC_STO1_ASRC_SFT: + reg = RT5682S_PLL_TRACK_2; + sft = RT5682S_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read(component, reg) >> sft) & 0xf; + switch (val) { + case RT5682S_CLK_SEL_I2S1_ASRC: + case RT5682S_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } +} + +static int is_headset_type(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if ((rt5682s->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) + return 1; + + return 0; +} + +static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN); + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(30000, 35000); + snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_11, 0x6666); + snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_12, 0xa82a); + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, + RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | + RT5682S_HPO_SEL_IP_EN_SW, RT5682S_HPO_L_PATH_EN | + RT5682S_HPO_R_PATH_EN | RT5682S_HPO_IP_EN_GATING); + snd_soc_component_write(component, RT5682S_HP_AMP_DET_CTL_1, 0x3050); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, + RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | + RT5682S_HPO_SEL_IP_EN_SW, 0); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, 0); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, 0); + break; + } + + return 0; +} + +static int sar_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 0); + break; + case SND_SOC_DAPM_POST_PMD: + rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 0); + break; + } + + return 0; +} + +/* Interface data select */ +static const char * const rt5682s_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_if2_adc_enum, RT5682S_DIG_INF2_DATA, + RT5682S_IF2_ADC_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_01_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC1_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_23_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC2_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_45_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC3_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_67_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC4_SEL_SFT, rt5682s_data_select); + +static const struct snd_kcontrol_new rt5682s_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682s_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682s_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682s_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682s_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682s_if1_67_adc_enum); + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682s_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682s_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682S_REC_MIXER, + RT5682S_M_CBJ_RM1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_rec1_r_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682S_REC_MIXER, + RT5682S_M_CBJ_RM1_R_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682s_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc1l_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC1L_SRC_SFT, rt5682s_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682s_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc1r_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC1R_SRC_SFT, rt5682s_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682s_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682s_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adcl_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADCL_SRC_SFT, rt5682s_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682s_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adcr_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADCR_SRC_SFT, rt5682s_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682s_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682s_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc2l_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC2L_SRC_SFT, rt5682s_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682s_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc2r_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC2R_SRC_SFT, rt5682s_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682s_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682s_if1_adc_slot_values[] = { + 0, 2, 4, 6, +}; + +static const char * const rt5682s_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_if1_adc_slot_enum, + RT5682S_TDM_CTRL, RT5682S_TDM_ADC_LCA_SFT, RT5682S_TDM_ADC_LCA_MASK, + rt5682s_if1_adc_slot_src, rt5682s_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682s_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682s_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682s_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_alg_dac_l1_enum, RT5682S_A_DAC1_MUX, + RT5682S_A_DACL1_SFT, rt5682s_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682s_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682s_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_alg_dac_r1_enum, RT5682S_A_DAC1_MUX, + RT5682S_A_DACR1_SFT, rt5682s_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682s_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682s_alg_dac_r1_enum); + +static const unsigned int rt5682s_adcdat_pin_values[] = { + 1, 3, +}; + +static const char * const rt5682s_adcdat_pin_select[] = { + "ADCDAT1", "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, + RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, RT5682S_GP4_PIN_MASK, + rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682s_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682s_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO MB1", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO MB2", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_MB2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PLL Powers */ + SND_SOC_DAPM_SUPPLY_S("PLLA_LDO", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_LDO", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_PLLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLA_BIAS", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BIAS_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_BIAS", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BIAS_PLLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLA", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_PLLB_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("PLLA_RST", 1, RT5682S_PWR_ANLG_3, + RT5682S_RSTB_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_RST", 1, RT5682S_PWR_ANLG_3, + RT5682S_RSTB_PLLB_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682S_DMIC_CTRL_1, RT5682S_DMIC_1_EN_SFT, 0, + set_dmic_power, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682s_rec1_l_mix, + ARRAY_SIZE(rt5682s_rec1_l_mix)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5682s_rec1_r_mix, + ARRAY_SIZE(rt5682s_rec1_r_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682S_CAL_REC, + RT5682S_PWR_RM1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5682S_CAL_REC, + RT5682S_PWR_RM1_R_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682S_PWR_DIG_1, + RT5682S_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682S_PWR_DIG_1, + RT5682S_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682S_CHOP_ADC, + RT5682S_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682S_PWR_DIG_2, + RT5682S_PWR_ADC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE_SFT, 1, rt5682s_sto1_adc_l_mix, + ARRAY_SIZE(rt5682s_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_R_MUTE_SFT, 1, rt5682s_sto1_adc_r_mix, + ARRAY_SIZE(rt5682s_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682S_PWR_DIG_1, RT5682S_PWR_I2S1_BIT, + 0, set_i2s_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682S_PWR_DIG_1, RT5682S_PWR_I2S2_BIT, + 0, set_i2s_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, &rt5682s_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, RT5682S_I2S1_SDP, + RT5682S_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, RT5682S_I2S2_SDP, + RT5682S_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_dac_l_mix, ARRAY_SIZE(rt5682s_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682s_dac_r_mix, ARRAY_SIZE(rt5682s_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, &rt5682s_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, &rt5682s_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682S_PWR_DIG_2, + RT5682S_PWR_DAC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_dac_l_mix, ARRAY_SIZE(rt5682s_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_dac_r_mix, ARRAY_SIZE(rt5682s_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682S_PWR_DIG_1, RT5682S_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682S_PWR_DIG_1, RT5682S_PWR_DAC_R1_BIT, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682s_hp_amp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, + RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682S_CLK_DET, + RT5682S_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MCLK0 DET PWR", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MCLK0_WD_BIT, 0, NULL, 0), + + /* SAR */ + SND_SOC_DAPM_SUPPLY("SAR", SND_SOC_NOPM, 0, 0, sar_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLLA", is_sys_clk_from_plla}, + {"ADC Stereo1 Filter", NULL, "PLLB", is_sys_clk_from_pllb}, + {"DAC Stereo1 Filter", NULL, "PLLA", is_sys_clk_from_plla}, + {"DAC Stereo1 Filter", NULL, "PLLB", is_sys_clk_from_pllb}, + {"PLLA", NULL, "PLLA_LDO"}, + {"PLLA", NULL, "PLLA_BIAS"}, + {"PLLA", NULL, "PLLA_RST"}, + {"PLLB", NULL, "PLLB_LDO"}, + {"PLLB", NULL, "PLLB_BIAS"}, + {"PLLB", NULL, "PLLB_RST"}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + + {"CLKDET SYS", NULL, "MCLK0 DET PWR"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "SAR", is_headset_type}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + {"RECMIX1R", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1R", NULL, "RECMIX1R Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + {"ADC1 R", NULL, "RECMIX1R"}, + {"ADC1 R", NULL, "ADC1 R Power"}, + {"ADC1 R", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "I2S1"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "SAR", is_headset_type}, + + {"HPOL", NULL, "HP Amp"}, + {"HPOR", NULL, "HP Amp"}, +}; + +static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, + RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, RT5682S_TDM_EN); + else + snd_soc_component_update_bits(component, + RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682S_TDM_TX_CH_4; + val |= RT5682S_TDM_RX_CH_4; + break; + case 6: + val |= RT5682S_TDM_TX_CH_6; + val |= RT5682S_TDM_RX_CH_6; + break; + case 8: + val |= RT5682S_TDM_TX_CH_8; + val |= RT5682S_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_TDM_CTRL, + RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682S_I2S1_TX_CHL_8 | RT5682S_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682S_TDM_CL_16; + cl = RT5682S_I2S1_TX_CHL_16 | RT5682S_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682S_TDM_CL_20; + cl = RT5682S_I2S1_TX_CHL_20 | RT5682S_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682S_TDM_CL_24; + cl = RT5682S_I2S1_TX_CHL_24 | RT5682S_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682S_TDM_CL_32; + cl = RT5682S_I2S1_TX_CHL_32 | RT5682S_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_TX_CHL_MASK | RT5682S_I2S1_RX_CHL_MASK, cl); + + return 0; +} + +static int rt5682s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int frame_size; + + rt5682s->lrck[dai->id] = params_rate(params); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682S_I2S1_DL_20; + len_2 |= RT5682S_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682S_I2S1_DL_24; + len_2 |= RT5682S_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682S_I2S1_DL_32; + len_2 |= RT5682S_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682S_I2S2_DL_8; + len_2 |= RT5682S_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682S_AIF1: + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_DL_MASK, len_1); + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_MONO_MASK, RT5682S_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_MONO_MASK, RT5682S_I2S1_MONO_DIS); + break; + case RT5682S_AIF2: + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_DL_MASK, len_2); + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MONO_MASK, RT5682S_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MONO_MASK, RT5682S_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682s->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682s->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682S_I2S_BP_INV; + tdm_ctrl |= RT5682S_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682S_AIF1) + tdm_ctrl |= RT5682S_TDM_S_LP_INV | RT5682S_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682S_AIF1) + tdm_ctrl |= RT5682S_TDM_S_BP_INV | RT5682S_TDM_S_LP_INV | + RT5682S_TDM_M_BP_INV | RT5682S_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682S_I2S_DF_LEFT; + tdm_ctrl |= RT5682S_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682S_I2S_DF_PCM_A; + tdm_ctrl |= RT5682S_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682S_I2S_DF_PCM_B; + tdm_ctrl |= RT5682S_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682S_AIF1: + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_MS_MASK | RT5682S_TDM_S_BP_MASK | + RT5682S_TDM_DF_MASK | RT5682S_TDM_M_BP_MASK | + RT5682S_TDM_M_LP_MASK | RT5682S_TDM_S_LP_MASK, + tdm_ctrl | rt5682s->master[dai->id]); + break; + case RT5682S_AIF2: + if (rt5682s->master[dai->id] == 0) + reg_val |= RT5682S_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MS_MASK | RT5682S_I2S_BP_MASK | + RT5682S_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682s_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int src = 0; + + if (freq == rt5682s->sysclk && clk_id == rt5682s->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682S_SCLK_S_MCLK: + src = RT5682S_CLK_SRC_MCLK; + break; + case RT5682S_SCLK_S_PLL1: + src = RT5682S_CLK_SRC_PLL1; + break; + case RT5682S_SCLK_S_PLL2: + src = RT5682S_CLK_SRC_PLL2; + break; + case RT5682S_SCLK_S_RCCLK: + src = RT5682S_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_SCLK_SRC_MASK, src << RT5682S_SCLK_SRC_SFT); + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_1, + RT5682S_I2S_M_CLK_SRC_MASK, src << RT5682S_I2S_M_CLK_SRC_SFT); + snd_soc_component_update_bits(component, RT5682S_I2S2_M_CLK_CTRL_1, + RT5682S_I2S2_M_CLK_SRC_MASK, src << RT5682S_I2S2_M_CLK_SRC_SFT); + + rt5682s->sysclk = freq; + rt5682s->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static const struct pll_calc_map plla_table[] = { + {2048000, 24576000, 0, 46, 2, true, false, false, false}, + {256000, 24576000, 0, 382, 2, true, false, false, false}, + {512000, 24576000, 0, 190, 2, true, false, false, false}, + {4096000, 24576000, 0, 22, 2, true, false, false, false}, + {1024000, 24576000, 0, 94, 2, true, false, false, false}, + {11289600, 22579200, 1, 22, 2, false, false, false, false}, + {1411200, 22579200, 0, 62, 2, true, false, false, false}, + {2822400, 22579200, 0, 30, 2, true, false, false, false}, + {12288000, 24576000, 1, 22, 2, false, false, false, false}, + {1536000, 24576000, 0, 62, 2, true, false, false, false}, + {3072000, 24576000, 0, 30, 2, true, false, false, false}, + {24576000, 49152000, 4, 22, 0, false, false, false, false}, + {3072000, 49152000, 0, 30, 0, true, false, false, false}, + {6144000, 49152000, 0, 30, 0, false, false, false, false}, + {49152000, 98304000, 10, 22, 0, false, true, false, false}, + {6144000, 98304000, 0, 30, 0, false, true, false, false}, + {12288000, 98304000, 1, 22, 0, false, true, false, false}, + {48000000, 3840000, 10, 22, 23, false, false, false, false}, + {24000000, 3840000, 4, 22, 23, false, false, false, false}, + {19200000, 3840000, 3, 23, 23, false, false, false, false}, + {38400000, 3840000, 8, 23, 23, false, false, false, false}, +}; + +static const struct pll_calc_map pllb_table[] = { + {48000000, 24576000, 8, 6, 3, false, false, false, false}, + {48000000, 22579200, 23, 12, 3, false, false, false, true}, + {24000000, 24576000, 3, 6, 3, false, false, false, false}, + {24000000, 22579200, 23, 26, 3, false, false, false, true}, + {19200000, 24576000, 2, 6, 3, false, false, false, false}, + {19200000, 22579200, 3, 5, 3, false, false, false, true}, + {38400000, 24576000, 6, 6, 3, false, false, false, false}, + {38400000, 22579200, 8, 5, 3, false, false, false, true}, + {3840000, 49152000, 0, 6, 0, true, false, false, false}, +}; + +static int find_pll_inter_combination(unsigned int f_in, unsigned int f_out, + struct pll_calc_map *a, struct pll_calc_map *b) +{ + int i, j; + + /* Look at PLLA table */ + for (i = 0; i < ARRAY_SIZE(plla_table); i++) { + if (plla_table[i].freq_in == f_in && plla_table[i].freq_out == f_out) { + memcpy(a, plla_table + i, sizeof(*a)); + return USE_PLLA; + } + } + + /* Look at PLLB table */ + for (i = 0; i < ARRAY_SIZE(pllb_table); i++) { + if (pllb_table[i].freq_in == f_in && pllb_table[i].freq_out == f_out) { + memcpy(b, pllb_table + i, sizeof(*b)); + return USE_PLLB; + } + } + + /* Find a combination of PLLA & PLLB */ + for (i = ARRAY_SIZE(plla_table) - 1; i >= 0; i--) { + if (plla_table[i].freq_in == f_in && plla_table[i].freq_out == 3840000) { + for (j = ARRAY_SIZE(pllb_table) - 1; j >= 0; j--) { + if (pllb_table[j].freq_in == 3840000 && + pllb_table[j].freq_out == f_out) { + memcpy(a, plla_table + i, sizeof(*a)); + memcpy(b, pllb_table + j, sizeof(*b)); + return USE_PLLAB; + } + } + } + } + + return -EINVAL; +} + +static int rt5682s_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct pll_calc_map a_map, b_map; + + if (source == rt5682s->pll_src[pll_id] && freq_in == rt5682s->pll_in[pll_id] && + freq_out == rt5682s->pll_out[pll_id]) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + rt5682s->pll_in[pll_id] = 0; + rt5682s->pll_out[pll_id] = 0; + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_SCLK_SRC_MASK, RT5682S_CLK_SRC_MCLK << RT5682S_SCLK_SRC_SFT); + return 0; + } + + switch (source) { + case RT5682S_PLL_S_MCLK: + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_PLL_SRC_MASK, RT5682S_PLL_SRC_MCLK); + break; + case RT5682S_PLL_S_BCLK1: + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_PLL_SRC_MASK, RT5682S_PLL_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + rt5682s->pll_comb = find_pll_inter_combination(freq_in, freq_out, + &a_map, &b_map); + + if ((pll_id == RT5682S_PLL1 && rt5682s->pll_comb == USE_PLLA) || + (pll_id == RT5682S_PLL2 && (rt5682s->pll_comb == USE_PLLB || + rt5682s->pll_comb == USE_PLLAB))) { + dev_dbg(component->dev, + "Supported freq conversion for PLL%d:(%d->%d): %d\n", + pll_id + 1, freq_in, freq_out, rt5682s->pll_comb); + } else { + dev_err(component->dev, + "Unsupported freq conversion for PLL%d:(%d->%d): %d\n", + pll_id + 1, freq_in, freq_out, rt5682s->pll_comb); + return -EINVAL; + } + + if (rt5682s->pll_comb == USE_PLLA || rt5682s->pll_comb == USE_PLLAB) { + dev_dbg(component->dev, + "PLLA: fin=%d fout=%d m_bp=%d k_bp=%d m=%d n=%d k=%d\n", + a_map.freq_in, a_map.freq_out, a_map.m_bp, a_map.k_bp, + (a_map.m_bp ? 0 : a_map.m), a_map.n, (a_map.k_bp ? 0 : a_map.k)); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_1, + RT5682S_PLLA_N_MASK, a_map.n); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_2, + RT5682S_PLLA_M_MASK | RT5682S_PLLA_K_MASK, + a_map.m << RT5682S_PLLA_M_SFT | a_map.k); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_6, + RT5682S_PLLA_M_BP_MASK | RT5682S_PLLA_K_BP_MASK, + a_map.m_bp << RT5682S_PLLA_M_BP_SFT | + a_map.k_bp << RT5682S_PLLA_K_BP_SFT); + } + + if (rt5682s->pll_comb == USE_PLLB || rt5682s->pll_comb == USE_PLLAB) { + dev_dbg(component->dev, + "PLLB: fin=%d fout=%d m_bp=%d k_bp=%d m=%d n=%d k=%d byp_ps=%d sel_ps=%d\n", + b_map.freq_in, b_map.freq_out, b_map.m_bp, b_map.k_bp, + (b_map.m_bp ? 0 : b_map.m), b_map.n, (b_map.k_bp ? 0 : b_map.k), + b_map.byp_ps, b_map.sel_ps); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_3, + RT5682S_PLLB_N_MASK, b_map.n); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_4, + RT5682S_PLLB_M_MASK | RT5682S_PLLB_K_MASK, + b_map.m << RT5682S_PLLB_M_SFT | b_map.k); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_6, + RT5682S_PLLB_SEL_PS_MASK | RT5682S_PLLB_BYP_PS_MASK | + RT5682S_PLLB_M_BP_MASK | RT5682S_PLLB_K_BP_MASK, + b_map.sel_ps << RT5682S_PLLB_SEL_PS_SFT | + b_map.byp_ps << RT5682S_PLLB_BYP_PS_SFT | + b_map.m_bp << RT5682S_PLLB_M_BP_SFT | + b_map.k_bp << RT5682S_PLLB_K_BP_SFT); + } + + if (rt5682s->pll_comb == USE_PLLB) + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_7, + RT5682S_PLLB_SRC_MASK, RT5682S_PLLB_SRC_DFIN); + + rt5682s->pll_in[pll_id] = freq_in; + rt5682s->pll_out[pll_id] = freq_out; + rt5682s->pll_src[pll_id] = source; + + return 0; +} + +static int rt5682s_set_bclk1_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s->bclk[dai->id] = ratio; + + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_2, + RT5682S_I2S2_BCLK_MS2_MASK, RT5682S_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_2, + RT5682S_I2S2_BCLK_MS2_MASK, RT5682S_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_PWR_LDO, RT5682S_PWR_LDO); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_DIG_GATE_CTRL, RT5682S_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_DIG_GATE_CTRL | RT5682S_PWR_LDO, 0); + break; + case SND_SOC_BIAS_ON: + break; + } + + return 0; +} + +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_48 48000 +#define CLK_44 44100 + +static bool rt5682s_clk_check(struct rt5682s_priv *rt5682s) +{ + if (!rt5682s->master[RT5682S_AIF1]) { + dev_dbg(rt5682s->component->dev, "dai clk fmt not set correctly\n"); + return false; + } + return true; +} + +static int rt5682s_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_MB, RT5682S_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_FV2, RT5682S_PWR_VREF2); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV2, RT5682S_PWR_FV2); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + /* Only need to power PLLB due to the rate set restriction */ + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLLB"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682s_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + if (!rt5682s_clk_check(rt5682s)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); + if (!rt5682s->jack_type) + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_FV2 | RT5682S_PWR_MB, 0); + + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLLB"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682s_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + const char * const clk_name = clk_hw_get_name(hw); + + if (!rt5682s_clk_check(rt5682s)) + return 0; + /* + * Only accept to set wclk rate to 44.1k or 48kHz. + */ + if (rt5682s->lrck[RT5682S_AIF1] != CLK_48 && + rt5682s->lrck[RT5682S_AIF1] != CLK_44) { + dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + __func__, clk_name, CLK_44, CLK_48); + return 0; + } + + return rt5682s->lrck[RT5682S_AIF1]; +} + +static long rt5682s_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + const char * const clk_name = clk_hw_get_name(hw); + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + /* + * Only accept to set wclk rate to 44.1k or 48kHz. + * It will force to 48kHz if not both. + */ + if (rate != CLK_48 && rate != CLK_44) { + dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + __func__, clk_name, CLK_44, CLK_48); + rate = CLK_48; + } + + return rate; +} + +static int rt5682s_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct clk *parent_clk; + const char * const clk_name = clk_hw_get_name(hw); + unsigned int clk_pll2_fout; + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + /* + * Whether the wclk's parent clk (mclk) exists or not, please ensure + * it is fixed or set to 48MHz before setting wclk rate. It's a + * temporary limitation. Only accept 48MHz clk as the clk provider. + * + * It will set the codec anyway by assuming mclk is 48MHz. + */ + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + /* + * To achieve the rate conversion from 48MHz to 44.1k or 48kHz, + * PLL2 is needed. + */ + clk_pll2_fout = rate * 512; + rt5682s_set_component_pll(component, RT5682S_PLL2, RT5682S_PLL_S_MCLK, + CLK_PLL2_FIN, clk_pll2_fout); + + rt5682s_set_component_sysclk(component, RT5682S_SCLK_S_PLL2, 0, + clk_pll2_fout, SND_SOC_CLOCK_IN); + + rt5682s->lrck[RT5682S_AIF1] = rate; + + return 0; +} + +static unsigned long rt5682s_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + unsigned int bclks_per_wclk; + + bclks_per_wclk = snd_soc_component_read(component, RT5682S_TDM_TCON_CTRL_1); + + switch (bclks_per_wclk & RT5682S_TDM_BCLK_MS1_MASK) { + case RT5682S_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682S_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682S_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682S_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682s_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682s_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682s_clk_check(rt5682s)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682s_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682s_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dai *dai; + unsigned long factor; + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + factor = rt5682s_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682S_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682S_AIF1); + return -ENODEV; + } + + return rt5682s_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682s_dai_clk_ops[RT5682S_DAI_NUM_CLKS] = { + [RT5682S_DAI_WCLK_IDX] = { + .prepare = rt5682s_wclk_prepare, + .unprepare = rt5682s_wclk_unprepare, + .recalc_rate = rt5682s_wclk_recalc_rate, + .round_rate = rt5682s_wclk_round_rate, + .set_rate = rt5682s_wclk_set_rate, + }, + [RT5682S_DAI_BCLK_IDX] = { + .recalc_rate = rt5682s_bclk_recalc_rate, + .round_rate = rt5682s_bclk_round_rate, + .set_rate = rt5682s_bclk_set_rate, + }, +}; + +static int rt5682s_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct rt5682s_platform_data *pdata = &rt5682s->pdata; + struct clk_hw *dai_clk_hw; + int i, ret; + + for (i = 0; i < RT5682S_DAI_NUM_CLKS; ++i) { + struct clk_init_data init = { }; + + dai_clk_hw = &rt5682s->dai_clks_hw[i]; + + switch (i) { + case RT5682S_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682s->mclk) { + init.parent_data = &(struct clk_parent_data){ + .fw_name = "mclk", + }; + init.num_parents = 1; + } + break; + case RT5682S_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + init.parent_hws = &(const struct clk_hw *){ + &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX] + }; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + return -EINVAL; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682s_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + ret = devm_clk_hw_register(dev, dai_clk_hw); + if (ret) { + dev_warn(dev, "Failed to register %s: %d\n", init.name, ret); + return ret; + } + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + } else { + ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, + init.name, dev_name(dev)); + if (ret) + return ret; + } + } + + return 0; +} + +static int rt5682s_dai_probe_clks(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int ret; + + /* Check if MCLK provided */ + rt5682s->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682s->mclk)) { + if (PTR_ERR(rt5682s->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682s->mclk); + return ret; + } + rt5682s->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682s_register_dai_clks(component); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682s->lrck[RT5682S_AIF1] = CLK_48; + + return 0; +} +#else +static inline int rt5682s_dai_probe_clks(struct snd_soc_component *component) +{ + return 0; +} +#endif /* CONFIG_COMMON_CLK */ + +static int rt5682s_probe(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; + int ret; + + rt5682s->component = component; + + ret = rt5682s_dai_probe_clks(component); + if (ret) + return ret; + + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "Vref2"); + snd_soc_dapm_sync(dapm); + return 0; +} + +static void rt5682s_remove(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s_reset(rt5682s); +} + +#ifdef CONFIG_PM +static int rt5682s_suspend(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + if (rt5682s->hs_jack && rt5682s->jack_type == SND_JACK_HEADSET) + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + + regcache_cache_only(rt5682s->regmap, true); + regcache_mark_dirty(rt5682s->regmap); + + return 0; +} + +static int rt5682s_resume(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682s->regmap, false); + regcache_sync(rt5682s->regmap); + + if (rt5682s->hs_jack) { + rt5682s->jack_type = 0; + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} +#else +#define rt5682s_suspend NULL +#define rt5682s_resume NULL +#endif + +const struct snd_soc_dai_ops rt5682s_aif1_dai_ops = { + .hw_params = rt5682s_hw_params, + .set_fmt = rt5682s_set_dai_fmt, + .set_tdm_slot = rt5682s_set_tdm_slot, + .set_bclk_ratio = rt5682s_set_bclk1_ratio, +}; + +const struct snd_soc_dai_ops rt5682s_aif2_dai_ops = { + .hw_params = rt5682s_hw_params, + .set_fmt = rt5682s_set_dai_fmt, + .set_bclk_ratio = rt5682s_set_bclk2_ratio, +}; + +const struct snd_soc_component_driver rt5682s_soc_component_dev = { + .probe = rt5682s_probe, + .remove = rt5682s_remove, + .suspend = rt5682s_suspend, + .resume = rt5682s_resume, + .set_bias_level = rt5682s_set_bias_level, + .controls = rt5682s_snd_controls, + .num_controls = ARRAY_SIZE(rt5682s_snd_controls), + .dapm_widgets = rt5682s_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682s_dapm_widgets), + .dapm_routes = rt5682s_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682s_dapm_routes), + .set_sysclk = rt5682s_set_component_sysclk, + .set_pll = rt5682s_set_component_pll, + .set_jack = rt5682s_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682s->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682s->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682s->pdata.jd_src); + device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", + &rt5682s->pdata.dmic_clk_rate); + device_property_read_u32(dev, "realtek,dmic-delay-ms", + &rt5682s->pdata.dmic_delay); + + rt5682s->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + if (device_property_read_string_array(dev, "clock-output-names", + rt5682s->pdata.dai_clk_names, + RT5682S_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682s->pdata.dai_clk_names[RT5682S_DAI_WCLK_IDX], + rt5682s->pdata.dai_clk_names[RT5682S_DAI_BCLK_IDX]); + + rt5682s->pdata.dmic_clk_driving_high = device_property_read_bool(dev, + "realtek,dmic-clk-driving-high"); + + return 0; +} + +static void rt5682s_calibrate(struct rt5682s_priv *rt5682s) +{ + unsigned int count, value; + + mutex_lock(&rt5682s->calibrate_mutex); + + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0xaa80); + usleep_range(15000, 20000); + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0xfa80); + regmap_write(rt5682s->regmap, RT5682S_PWR_DIG_1, 0x01c0); + regmap_write(rt5682s->regmap, RT5682S_MICBIAS_2, 0x0380); + regmap_write(rt5682s->regmap, RT5682S_GLB_CLK, 0x8000); + regmap_write(rt5682s->regmap, RT5682S_ADDA_CLK_1, 0x1001); + regmap_write(rt5682s->regmap, RT5682S_CHOP_DAC_2, 0x3030); + regmap_write(rt5682s->regmap, RT5682S_CHOP_ADC, 0xb000); + regmap_write(rt5682s->regmap, RT5682S_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682s->regmap, RT5682S_CAL_REC, 0x5151); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682s->regmap, RT5682S_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682s->regmap, RT5682S_HP_CALIB_ST_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + dev_err(rt5682s->component->dev, "HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682s->regmap, RT5682S_MICBIAS_2, 0x0180); + regmap_write(rt5682s->regmap, RT5682S_CAL_REC, 0x5858); + regmap_write(rt5682s->regmap, RT5682S_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_2, 0x0320); + regmap_write(rt5682s->regmap, RT5682S_PWR_DIG_1, 0x00c0); + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0x0800); + regmap_write(rt5682s->regmap, RT5682S_GLB_CLK, 0x0000); + + mutex_unlock(&rt5682s->calibrate_mutex); +} + +static const struct regmap_config rt5682s_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682S_MAX_REG, + .volatile_reg = rt5682s_volatile_register, + .readable_reg = rt5682s_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682s_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682s_reg), + .use_single_read = true, + .use_single_write = true, +}; + +static struct snd_soc_dai_driver rt5682s_dai[] = { + { + .name = "rt5682s-aif1", + .id = RT5682S_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .ops = &rt5682s_aif1_dai_ops, + }, + { + .name = "rt5682s-aif2", + .id = RT5682S_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .ops = &rt5682s_aif2_dai_ops, + }, +}; + +static void rt5682s_i2c_disable_regulators(void *data) +{ + struct rt5682s_priv *rt5682s = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); +} + +static int rt5682s_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682s_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682s_priv *rt5682s; + int i, ret; + unsigned int val; + + rt5682s = devm_kzalloc(&i2c->dev, sizeof(struct rt5682s_priv), GFP_KERNEL); + if (!rt5682s) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682s); + + rt5682s->pdata = i2s_default_platform_data; + + if (pdata) + rt5682s->pdata = *pdata; + else + rt5682s_parse_dt(rt5682s, &i2c->dev); + + rt5682s->regmap = devm_regmap_init_i2c(i2c, &rt5682s_regmap); + if (IS_ERR(rt5682s->regmap)) { + ret = PTR_ERR(rt5682s->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682s->supplies); i++) + rt5682s->supplies[i].supply = rt5682s_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = devm_add_action_or_reset(&i2c->dev, rt5682s_i2c_disable_regulators, rt5682s); + if (ret) + return ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682s->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682s->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682s")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 50 ms minimum */ + usleep_range(50000, 55000); + + regmap_read(rt5682s->regmap, RT5682S_DEVICE_ID, &val); + if (val != DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %x is not rt5682s\n", val); + return -ENODEV; + } + + rt5682s_reset(rt5682s); + rt5682s_apply_patch_list(rt5682s, &i2c->dev); + + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_2, + RT5682S_DLDO_I_LIMIT_MASK, RT5682S_DLDO_I_LIMIT_DIS); + usleep_range(20000, 25000); + + mutex_init(&rt5682s->calibrate_mutex); + mutex_init(&rt5682s->sar_mutex); + rt5682s_calibrate(rt5682s); + + regmap_update_bits(rt5682s->regmap, RT5682S_MICBIAS_2, + RT5682S_PWR_CLK25M_MASK | RT5682S_PWR_CLK1M_MASK, + RT5682S_PWR_CLK25M_PD | RT5682S_PWR_CLK1M_PU); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_1, + RT5682S_PWR_BG, RT5682S_PWR_BG); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_LOGIC_CTRL_2, + RT5682S_HP_SIG_SRC_MASK, RT5682S_HP_SIG_SRC_1BIT_CTL); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_PM_HP_MASK, RT5682S_PM_HP_HV); + + /* DMIC data pin */ + switch (rt5682s->pdata.dmic1_data_pin) { + case RT5682S_DMIC1_DATA_NULL: + break; + case RT5682S_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682s->regmap, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_DP_MASK, RT5682S_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP2_PIN_MASK, RT5682S_GP2_PIN_DMIC_SDA); + break; + case RT5682S_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682s->regmap, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_DP_MASK, RT5682S_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP5_PIN_MASK, RT5682S_GP5_PIN_DMIC_SDA); + break; + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + /* DMIC clk pin */ + switch (rt5682s->pdata.dmic1_clk_pin) { + case RT5682S_DMIC1_CLK_NULL: + break; + case RT5682S_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP1_PIN_MASK, RT5682S_GP1_PIN_DMIC_CLK); + break; + case RT5682S_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP3_PIN_MASK, RT5682S_GP3_PIN_DMIC_CLK); + if (rt5682s->pdata.dmic_clk_driving_high) + regmap_update_bits(rt5682s->regmap, RT5682S_PAD_DRIVING_CTRL, + RT5682S_PAD_DRV_GP3_MASK, RT5682S_PAD_DRV_GP3_HIGH); + break; + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + + INIT_DELAYED_WORK(&rt5682s->jack_detect_work, rt5682s_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682s->jd_check_work, rt5682s_jd_check_handler); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682s_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5682s", rt5682s); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5682s_soc_component_dev, + rt5682s_dai, ARRAY_SIZE(rt5682s_dai)); +} + +static void rt5682s_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682s_priv *rt5682s = i2c_get_clientdata(client); + + disable_irq(client->irq); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + rt5682s_reset(rt5682s); +} + +static int rt5682s_i2c_remove(struct i2c_client *client) +{ + rt5682s_i2c_shutdown(client); + + return 0; +} + +static const struct of_device_id rt5682s_of_match[] = { + {.compatible = "realtek,rt5682s"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682s_of_match); + +static const struct acpi_device_id rt5682s_acpi_match[] = { + {"RTL5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682s_acpi_match); + +static const struct i2c_device_id rt5682s_i2c_id[] = { + {"rt5682s", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682s_i2c_id); + +static struct i2c_driver rt5682s_i2c_driver = { + .driver = { + .name = "rt5682s", + .of_match_table = rt5682s_of_match, + .acpi_match_table = rt5682s_acpi_match, + .probe_type = PROBE_PREFER_ASYNCHRONOUS, + }, + .probe = rt5682s_i2c_probe, + .remove = rt5682s_i2c_remove, + .shutdown = rt5682s_i2c_shutdown, + .id_table = rt5682s_i2c_id, +}; +module_i2c_driver(rt5682s_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682I-VS driver"); +MODULE_AUTHOR("Derek Fang <derek.fang(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h new file mode 100644 index 000000000000..7c755e5efb81 --- /dev/null +++ b/sound/soc/codecs/rt5682s.h @@ -0,0 +1,1455 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5682s.h -- RT5682I-VS ALSA SoC audio driver + * + * Copyright 2021 Realtek Microelectronics + * Author: Derek Fang <derek.fang(a)realtek.com> + */ + +#ifndef __RT5682S_H__ +#define __RT5682S_H__ + +#include <sound/rt5682s.h> +#include <linux/regulator/consumer.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> + + +/* Info */ +#define RT5682S_RESET 0x0000 +#define RT5682S_VERSION_ID 0x00fd +#define RT5682S_VENDOR_ID 0x00fe +#define RT5682S_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682S_HP_CTRL_1 0x0002 +#define RT5682S_HP_CTRL_2 0x0003 +#define RT5682S_HPL_GAIN 0x0005 +#define RT5682S_HPR_GAIN 0x0006 + +#define RT5682S_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682S_CBJ_BST_CTRL 0x000b +#define RT5682S_CBJ_DET_CTRL 0x000f +#define RT5682S_CBJ_CTRL_1 0x0010 +#define RT5682S_CBJ_CTRL_2 0x0011 +#define RT5682S_CBJ_CTRL_3 0x0012 +#define RT5682S_CBJ_CTRL_4 0x0013 +#define RT5682S_CBJ_CTRL_5 0x0014 +#define RT5682S_CBJ_CTRL_6 0x0015 +#define RT5682S_CBJ_CTRL_7 0x0016 +#define RT5682S_CBJ_CTRL_8 0x0017 +/* I/O - ADC/DAC/DMIC */ +#define RT5682S_DAC1_DIG_VOL 0x0019 +#define RT5682S_STO1_ADC_DIG_VOL 0x001c +#define RT5682S_STO1_ADC_BOOST 0x001f +#define RT5682S_HP_IMP_GAIN_1 0x0022 +#define RT5682S_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682S_SIDETONE_CTRL 0x0024 +#define RT5682S_STO1_ADC_MIXER 0x0026 +#define RT5682S_AD_DA_MIXER 0x0029 +#define RT5682S_STO1_DAC_MIXER 0x002a +#define RT5682S_A_DAC1_MUX 0x002b +#define RT5682S_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682S_REC_MIXER 0x003c +#define RT5682S_CAL_REC 0x0044 +/* HP Analog Offset Control */ +#define RT5682S_HP_ANA_OST_CTRL_1 0x004b +#define RT5682S_HP_ANA_OST_CTRL_2 0x004c +#define RT5682S_HP_ANA_OST_CTRL_3 0x004d +/* Power */ +#define RT5682S_PWR_DIG_1 0x0061 +#define RT5682S_PWR_DIG_2 0x0062 +#define RT5682S_PWR_ANLG_1 0x0063 +#define RT5682S_PWR_ANLG_2 0x0064 +#define RT5682S_PWR_ANLG_3 0x0065 +#define RT5682S_PWR_MIXER 0x0066 + +#define RT5682S_MB_CTRL 0x0067 +#define RT5682S_CLK_GATE_TCON_1 0x0068 +#define RT5682S_CLK_GATE_TCON_2 0x0069 +#define RT5682S_CLK_GATE_TCON_3 0x006a +/* Clock Detect */ +#define RT5682S_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682S_RESET_LPF_CTRL 0x006c +#define RT5682S_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682S_DMIC_CTRL_1 0x006e +#define RT5682S_LPF_AD_DMIC 0x006f +/* Format - ADC/DAC */ +#define RT5682S_I2S1_SDP 0x0070 +#define RT5682S_I2S2_SDP 0x0071 +#define RT5682S_ADDA_CLK_1 0x0073 +#define RT5682S_ADDA_CLK_2 0x0074 +#define RT5682S_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682S_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682S_TDM_CTRL 0x0079 +#define RT5682S_TDM_ADDA_CTRL_1 0x007a +#define RT5682S_TDM_ADDA_CTRL_2 0x007b +#define RT5682S_DATA_SEL_CTRL_1 0x007c +#define RT5682S_TDM_TCON_CTRL_1 0x007e +#define RT5682S_TDM_TCON_CTRL_2 0x007f +/* Function - Analog */ +#define RT5682S_GLB_CLK 0x0080 +#define RT5682S_PLL_TRACK_1 0x0083 +#define RT5682S_PLL_TRACK_2 0x0084 +#define RT5682S_PLL_TRACK_3 0x0085 +#define RT5682S_PLL_TRACK_4 0x0086 +#define RT5682S_PLL_TRACK_5 0x0087 +#define RT5682S_PLL_TRACK_6 0x0088 +#define RT5682S_PLL_TRACK_11 0x008c +#define RT5682S_DEPOP_1 0x008e +#define RT5682S_HP_CHARGE_PUMP_1 0x008f +#define RT5682S_HP_CHARGE_PUMP_2 0x0091 +#define RT5682S_HP_CHARGE_PUMP_3 0x0092 +#define RT5682S_MICBIAS_1 0x0093 +#define RT5682S_MICBIAS_2 0x0094 +#define RT5682S_MICBIAS_3 0x0095 + +#define RT5682S_PLL_TRACK_12 0x0096 +#define RT5682S_PLL_TRACK_14 0x0097 +#define RT5682S_PLL_CTRL_1 0x0098 +#define RT5682S_PLL_CTRL_2 0x0099 +#define RT5682S_PLL_CTRL_3 0x009a +#define RT5682S_PLL_CTRL_4 0x009b +#define RT5682S_PLL_CTRL_5 0x009c +#define RT5682S_PLL_CTRL_6 0x009d +#define RT5682S_PLL_CTRL_7 0x009e + +#define RT5682S_RC_CLK_CTRL 0x009f +#define RT5682S_I2S2_M_CLK_CTRL_1 0x00a0 +#define RT5682S_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682S_I2S2_F_DIV_CTRL_2 0x00a4 + +#define RT5682S_IRQ_CTRL_1 0x00b6 +#define RT5682S_IRQ_CTRL_2 0x00b7 +#define RT5682S_IRQ_CTRL_3 0x00b8 +#define RT5682S_IRQ_CTRL_4 0x00b9 +#define RT5682S_INT_ST_1 0x00be +#define RT5682S_GPIO_CTRL_1 0x00c0 +#define RT5682S_GPIO_CTRL_2 0x00c1 +#define RT5682S_GPIO_ST 0x00c2 +#define RT5682S_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682S_MID_HP_AMP_DET 0x00d2 +#define RT5682S_LOW_HP_AMP_DET 0x00d3 +#define RT5682S_DELAY_BUF_CTRL 0x00d4 +#define RT5682S_SV_ZCD_1 0x00d9 +#define RT5682S_SV_ZCD_2 0x00da +#define RT5682S_IL_CMD_1 0x00db +#define RT5682S_IL_CMD_2 0x00dc +#define RT5682S_IL_CMD_3 0x00dd +#define RT5682S_IL_CMD_4 0x00de +#define RT5682S_IL_CMD_5 0x00df +#define RT5682S_IL_CMD_6 0x00e0 +#define RT5682S_4BTN_IL_CMD_1 0x00e2 +#define RT5682S_4BTN_IL_CMD_2 0x00e3 +#define RT5682S_4BTN_IL_CMD_3 0x00e4 +#define RT5682S_4BTN_IL_CMD_4 0x00e5 +#define RT5682S_4BTN_IL_CMD_5 0x00e6 +#define RT5682S_4BTN_IL_CMD_6 0x00e7 +#define RT5682S_4BTN_IL_CMD_7 0x00e8 + +#define RT5682S_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682S_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682S_AJD1_CTRL 0x00f0 +#define RT5682S_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682S_DUMMY_1 0x00fa +#define RT5682S_DUMMY_2 0x00fb +#define RT5682S_DUMMY_3 0x00fc + +#define RT5682S_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682S_BIAS_CUR_CTRL_2 0x010b +#define RT5682S_BIAS_CUR_CTRL_3 0x010c +#define RT5682S_BIAS_CUR_CTRL_4 0x010d +#define RT5682S_BIAS_CUR_CTRL_5 0x010e +#define RT5682S_BIAS_CUR_CTRL_6 0x010f +#define RT5682S_BIAS_CUR_CTRL_7 0x0110 +#define RT5682S_BIAS_CUR_CTRL_8 0x0111 +#define RT5682S_BIAS_CUR_CTRL_9 0x0112 +#define RT5682S_BIAS_CUR_CTRL_10 0x0113 +#define RT5682S_VREF_REC_OP_FB_CAP_CTRL_1 0x0117 +#define RT5682S_VREF_REC_OP_FB_CAP_CTRL_2 0x0118 +#define RT5682S_CHARGE_PUMP_1 0x0125 +#define RT5682S_DIG_IN_CTRL_1 0x0132 +#define RT5682S_PAD_DRIVING_CTRL 0x0136 +#define RT5682S_CHOP_DAC_1 0x0139 +#define RT5682S_CHOP_DAC_2 0x013a +#define RT5682S_CHOP_ADC 0x013b +#define RT5682S_CALIB_ADC_CTRL 0x013c +#define RT5682S_VOL_TEST 0x013f +#define RT5682S_SPKVDD_DET_ST 0x0142 +#define RT5682S_TEST_MODE_CTRL_1 0x0145 +#define RT5682S_TEST_MODE_CTRL_2 0x0146 +#define RT5682S_TEST_MODE_CTRL_3 0x0147 +#define RT5682S_TEST_MODE_CTRL_4 0x0148 +#define RT5682S_PLL_INTERNAL_1 0x0156 +#define RT5682S_PLL_INTERNAL_2 0x0157 +#define RT5682S_PLL_INTERNAL_3 0x0158 +#define RT5682S_PLL_INTERNAL_4 0x0159 +#define RT5682S_STO_NG2_CTRL_1 0x0160 +#define RT5682S_STO_NG2_CTRL_2 0x0161 +#define RT5682S_STO_NG2_CTRL_3 0x0162 +#define RT5682S_STO_NG2_CTRL_4 0x0163 +#define RT5682S_STO_NG2_CTRL_5 0x0164 +#define RT5682S_STO_NG2_CTRL_6 0x0165 +#define RT5682S_STO_NG2_CTRL_7 0x0166 +#define RT5682S_STO_NG2_CTRL_8 0x0167 +#define RT5682S_STO_NG2_CTRL_9 0x0168 +#define RT5682S_STO_NG2_CTRL_10 0x0169 +#define RT5682S_STO1_DAC_SIL_DET 0x0190 +#define RT5682S_SIL_PSV_CTRL1 0x0194 +#define RT5682S_SIL_PSV_CTRL2 0x0195 +#define RT5682S_SIL_PSV_CTRL3 0x0197 +#define RT5682S_SIL_PSV_CTRL4 0x0198 +#define RT5682S_SIL_PSV_CTRL5 0x0199 +#define RT5682S_HP_IMP_SENS_CTRL_1 0x01ac +#define RT5682S_HP_IMP_SENS_CTRL_2 0x01ad +#define RT5682S_HP_IMP_SENS_CTRL_3 0x01ae +#define RT5682S_HP_IMP_SENS_CTRL_4 0x01af +#define RT5682S_HP_IMP_SENS_CTRL_5 0x01b0 +#define RT5682S_HP_IMP_SENS_CTRL_6 0x01b1 +#define RT5682S_HP_IMP_SENS_CTRL_7 0x01b2 +#define RT5682S_HP_IMP_SENS_CTRL_8 0x01b3 +#define RT5682S_HP_IMP_SENS_CTRL_9 0x01b4 +#define RT5682S_HP_IMP_SENS_CTRL_10 0x01b5 +#define RT5682S_HP_IMP_SENS_CTRL_11 0x01b6 +#define RT5682S_HP_IMP_SENS_CTRL_12 0x01b7 +#define RT5682S_HP_IMP_SENS_CTRL_13 0x01b8 +#define RT5682S_HP_IMP_SENS_CTRL_14 0x01b9 +#define RT5682S_HP_IMP_SENS_CTRL_15 0x01ba +#define RT5682S_HP_IMP_SENS_CTRL_16 0x01bb +#define RT5682S_HP_IMP_SENS_CTRL_17 0x01bc +#define RT5682S_HP_IMP_SENS_CTRL_18 0x01bd +#define RT5682S_HP_IMP_SENS_CTRL_19 0x01be +#define RT5682S_HP_IMP_SENS_CTRL_20 0x01bf +#define RT5682S_HP_IMP_SENS_CTRL_21 0x01c0 +#define RT5682S_HP_IMP_SENS_CTRL_22 0x01c1 +#define RT5682S_HP_IMP_SENS_CTRL_23 0x01c2 +#define RT5682S_HP_IMP_SENS_CTRL_24 0x01c3 +#define RT5682S_HP_IMP_SENS_CTRL_25 0x01c4 +#define RT5682S_HP_IMP_SENS_CTRL_26 0x01c5 +#define RT5682S_HP_IMP_SENS_CTRL_27 0x01c6 +#define RT5682S_HP_IMP_SENS_CTRL_28 0x01c7 +#define RT5682S_HP_IMP_SENS_CTRL_29 0x01c8 +#define RT5682S_HP_IMP_SENS_CTRL_30 0x01c9 +#define RT5682S_HP_IMP_SENS_CTRL_31 0x01ca +#define RT5682S_HP_IMP_SENS_CTRL_32 0x01cb +#define RT5682S_HP_IMP_SENS_CTRL_33 0x01cc +#define RT5682S_HP_IMP_SENS_CTRL_34 0x01cd +#define RT5682S_HP_IMP_SENS_CTRL_35 0x01ce +#define RT5682S_HP_IMP_SENS_CTRL_36 0x01cf +#define RT5682S_HP_IMP_SENS_CTRL_37 0x01d0 +#define RT5682S_HP_IMP_SENS_CTRL_38 0x01d1 +#define RT5682S_HP_IMP_SENS_CTRL_39 0x01d2 +#define RT5682S_HP_IMP_SENS_CTRL_40 0x01d3 +#define RT5682S_HP_IMP_SENS_CTRL_41 0x01d4 +#define RT5682S_HP_IMP_SENS_CTRL_42 0x01d5 +#define RT5682S_HP_IMP_SENS_CTRL_43 0x01d6 +#define RT5682S_HP_IMP_SENS_CTRL_44 0x01d7 +#define RT5682S_HP_IMP_SENS_CTRL_45 0x01d8 +#define RT5682S_HP_IMP_SENS_CTRL_46 0x01d9 +#define RT5682S_HP_LOGIC_CTRL_1 0x01da +#define RT5682S_HP_LOGIC_CTRL_2 0x01db +#define RT5682S_HP_LOGIC_CTRL_3 0x01dc +#define RT5682S_HP_CALIB_CTRL_1 0x01de +#define RT5682S_HP_CALIB_CTRL_2 0x01df +#define RT5682S_HP_CALIB_CTRL_3 0x01e0 +#define RT5682S_HP_CALIB_CTRL_4 0x01e1 +#define RT5682S_HP_CALIB_CTRL_5 0x01e2 +#define RT5682S_HP_CALIB_CTRL_6 0x01e3 +#define RT5682S_HP_CALIB_CTRL_7 0x01e4 +#define RT5682S_HP_CALIB_CTRL_8 0x01e5 +#define RT5682S_HP_CALIB_CTRL_9 0x01e6 +#define RT5682S_HP_CALIB_CTRL_10 0x01e7 +#define RT5682S_HP_CALIB_CTRL_11 0x01e8 +#define RT5682S_HP_CALIB_ST_1 0x01ea +#define RT5682S_HP_CALIB_ST_2 0x01eb +#define RT5682S_HP_CALIB_ST_3 0x01ec +#define RT5682S_HP_CALIB_ST_4 0x01ed +#define RT5682S_HP_CALIB_ST_5 0x01ee +#define RT5682S_HP_CALIB_ST_6 0x01ef +#define RT5682S_HP_CALIB_ST_7 0x01f0 +#define RT5682S_HP_CALIB_ST_8 0x01f1 +#define RT5682S_HP_CALIB_ST_9 0x01f2 +#define RT5682S_HP_CALIB_ST_10 0x01f3 +#define RT5682S_HP_CALIB_ST_11 0x01f4 +#define RT5682S_SAR_IL_CMD_1 0x0210 +#define RT5682S_SAR_IL_CMD_2 0x0211 +#define RT5682S_SAR_IL_CMD_3 0x0212 +#define RT5682S_SAR_IL_CMD_4 0x0213 +#define RT5682S_SAR_IL_CMD_5 0x0214 +#define RT5682S_SAR_IL_CMD_6 0x0215 +#define RT5682S_SAR_IL_CMD_7 0x0216 +#define RT5682S_SAR_IL_CMD_8 0x0217 +#define RT5682S_SAR_IL_CMD_9 0x0218 +#define RT5682S_SAR_IL_CMD_10 0x0219 +#define RT5682S_SAR_IL_CMD_11 0x021a +#define RT5682S_SAR_IL_CMD_12 0x021b +#define RT5682S_SAR_IL_CMD_13 0x021c +#define RT5682S_SAR_IL_CMD_14 0x021d +#define RT5682S_DUMMY_4 0x02fa +#define RT5682S_DUMMY_5 0x02fb +#define RT5682S_DUMMY_6 0x02fc +#define RT5682S_VERSION_ID_HIDE 0x03fe +#define RT5682S_VERSION_ID_CUS 0x03ff +#define RT5682S_SCAN_CTL 0x0500 +#define RT5682S_HP_AMP_DET 0x0600 +#define RT5682S_BIAS_CUR_CTRL_11 0x0610 +#define RT5682S_BIAS_CUR_CTRL_12 0x0611 +#define RT5682S_BIAS_CUR_CTRL_13 0x0620 +#define RT5682S_BIAS_CUR_CTRL_14 0x0621 +#define RT5682S_BIAS_CUR_CTRL_15 0x0630 +#define RT5682S_BIAS_CUR_CTRL_16 0x0631 +#define RT5682S_BIAS_CUR_CTRL_17 0x0640 +#define RT5682S_BIAS_CUR_CTRL_18 0x0641 +#define RT5682S_I2C_TRANS_CTRL 0x07fa +#define RT5682S_DUMMY_7 0x08fa +#define RT5682S_DUMMY_8 0x08fb +#define RT5682S_DMIC_FLOAT_DET 0x0d00 +#define RT5682S_HA_CMP_OP_1 0x1100 +#define RT5682S_HA_CMP_OP_2 0x1101 +#define RT5682S_HA_CMP_OP_3 0x1102 +#define RT5682S_HA_CMP_OP_4 0x1103 +#define RT5682S_HA_CMP_OP_5 0x1104 +#define RT5682S_HA_CMP_OP_6 0x1105 +#define RT5682S_HA_CMP_OP_7 0x1106 +#define RT5682S_HA_CMP_OP_8 0x1107 +#define RT5682S_HA_CMP_OP_9 0x1108 +#define RT5682S_HA_CMP_OP_10 0x1109 +#define RT5682S_HA_CMP_OP_11 0x110a +#define RT5682S_HA_CMP_OP_12 0x110b +#define RT5682S_HA_CMP_OP_13 0x110c +#define RT5682S_HA_CMP_OP_14 0x1111 +#define RT5682S_HA_CMP_OP_15 0x1112 +#define RT5682S_HA_CMP_OP_16 0x1113 +#define RT5682S_HA_CMP_OP_17 0x1114 +#define RT5682S_HA_CMP_OP_18 0x1115 +#define RT5682S_HA_CMP_OP_19 0x1116 +#define RT5682S_HA_CMP_OP_20 0x1117 +#define RT5682S_HA_CMP_OP_21 0x1118 +#define RT5682S_HA_CMP_OP_22 0x1119 +#define RT5682S_HA_CMP_OP_23 0x111a +#define RT5682S_HA_CMP_OP_24 0x111b +#define RT5682S_HA_CMP_OP_25 0x111c +#define RT5682S_NEW_CBJ_DET_CTL_1 0x1401 +#define RT5682S_NEW_CBJ_DET_CTL_2 0x1402 +#define RT5682S_NEW_CBJ_DET_CTL_3 0x1403 +#define RT5682S_NEW_CBJ_DET_CTL_4 0x1404 +#define RT5682S_NEW_CBJ_DET_CTL_5 0x1406 +#define RT5682S_NEW_CBJ_DET_CTL_6 0x1407 +#define RT5682S_NEW_CBJ_DET_CTL_7 0x1408 +#define RT5682S_NEW_CBJ_DET_CTL_8 0x1409 +#define RT5682S_NEW_CBJ_DET_CTL_9 0x140a +#define RT5682S_NEW_CBJ_DET_CTL_10 0x140b +#define RT5682S_NEW_CBJ_DET_CTL_11 0x140c +#define RT5682S_NEW_CBJ_DET_CTL_12 0x140d +#define RT5682S_NEW_CBJ_DET_CTL_13 0x140e +#define RT5682S_NEW_CBJ_DET_CTL_14 0x140f +#define RT5682S_NEW_CBJ_DET_CTL_15 0x1410 +#define RT5682S_NEW_CBJ_DET_CTL_16 0x1411 +#define RT5682S_DA_FILTER_1 0x1801 +#define RT5682S_DA_FILTER_2 0x1802 +#define RT5682S_DA_FILTER_3 0x1803 +#define RT5682S_DA_FILTER_4 0x1804 +#define RT5682S_DA_FILTER_5 0x1805 +#define RT5682S_CLK_SW_TEST_1 0x2c00 +#define RT5682S_CLK_SW_TEST_2 0x3400 +#define RT5682S_CLK_SW_TEST_3 0x3404 +#define RT5682S_CLK_SW_TEST_4 0x3405 +#define RT5682S_CLK_SW_TEST_5 0x3406 +#define RT5682S_CLK_SW_TEST_6 0x3407 +#define RT5682S_CLK_SW_TEST_7 0x3408 +#define RT5682S_CLK_SW_TEST_8 0x3409 +#define RT5682S_CLK_SW_TEST_9 0x340a +#define RT5682S_CLK_SW_TEST_10 0x340b +#define RT5682S_CLK_SW_TEST_11 0x340c +#define RT5682S_CLK_SW_TEST_12 0x340d +#define RT5682S_CLK_SW_TEST_13 0x340e +#define RT5682S_CLK_SW_TEST_14 0x340f +#define RT5682S_EFUSE_MANU_WRITE_1 0x3410 +#define RT5682S_EFUSE_MANU_WRITE_2 0x3411 +#define RT5682S_EFUSE_MANU_WRITE_3 0x3412 +#define RT5682S_EFUSE_MANU_WRITE_4 0x3413 +#define RT5682S_EFUSE_MANU_WRITE_5 0x3414 +#define RT5682S_EFUSE_MANU_WRITE_6 0x3415 +#define RT5682S_EFUSE_READ_1 0x3424 +#define RT5682S_EFUSE_READ_2 0x3425 +#define RT5682S_EFUSE_READ_3 0x3426 +#define RT5682S_EFUSE_READ_4 0x3427 +#define RT5682S_EFUSE_READ_5 0x3428 +#define RT5682S_EFUSE_READ_6 0x3429 +#define RT5682S_EFUSE_READ_7 0x342a +#define RT5682S_EFUSE_READ_8 0x342b +#define RT5682S_EFUSE_READ_9 0x342c +#define RT5682S_EFUSE_READ_10 0x342d +#define RT5682S_EFUSE_READ_11 0x342e +#define RT5682S_EFUSE_READ_12 0x342f +#define RT5682S_EFUSE_READ_13 0x3430 +#define RT5682S_EFUSE_READ_14 0x3431 +#define RT5682S_EFUSE_READ_15 0x3432 +#define RT5682S_EFUSE_READ_16 0x3433 +#define RT5682S_EFUSE_READ_17 0x3434 +#define RT5682S_EFUSE_READ_18 0x3435 +#define RT5682S_EFUSE_TIMING_CTL_1 0x3440 +#define RT5682S_EFUSE_TIMING_CTL_2 0x3441 +#define RT5682S_PILOT_DIG_CTL_1 0x3500 +#define RT5682S_PILOT_DIG_CTL_2 0x3501 +#define RT5682S_HP_AMP_DET_CTL_1 0x3b00 +#define RT5682S_HP_AMP_DET_CTL_2 0x3b01 +#define RT5682S_HP_AMP_DET_CTL_3 0x3b02 +#define RT5682S_HP_AMP_DET_CTL_4 0x3b03 + +#define RT5682S_MAX_REG (RT5682S_HP_AMP_DET_CTL_4) + +/* global definition */ +#define RT5682S_L_MUTE (0x1 << 15) +#define RT5682S_L_MUTE_SFT 15 +#define RT5682S_R_MUTE (0x1 << 7) +#define RT5682S_R_MUTE_SFT 7 +#define RT5682S_L_VOL_SFT 8 +#define RT5682S_R_VOL_SFT 0 +#define RT5682S_CLK_SRC_MCLK (0x0) +#define RT5682S_CLK_SRC_PLL1 (0x1) +#define RT5682S_CLK_SRC_PLL2 (0x2) +#define RT5682S_CLK_SRC_RCCLK (0x4) /* 25M */ + + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5682S_HPO_L_PATH_MASK (0x1 << 14) +#define RT5682S_HPO_L_PATH_EN (0x1 << 14) +#define RT5682S_HPO_L_PATH_DIS (0x0 << 14) +#define RT5682S_HPO_R_PATH_MASK (0x1 << 13) +#define RT5682S_HPO_R_PATH_EN (0x1 << 13) +#define RT5682S_HPO_R_PATH_DIS (0x0 << 13) +#define RT5682S_HPO_SEL_IP_EN_SW (0x1) +#define RT5682S_HPO_IP_EN_GATING (0x1) +#define RT5682S_HPO_IP_NO_GATING (0x0) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682S_G_HP (0xf << 8) +#define RT5682S_G_HP_SFT 8 +#define RT5682S_G_STO_DA_DMIX (0xf) +#define RT5682S_G_STO_DA_SFT 0 + +/* Embeeded Jack and Type Detection Control 2 (0x0010) */ +#define RT5682S_EMB_JD_MASK (0x1 << 15) +#define RT5682S_EMB_JD_EN (0x1 << 15) +#define RT5682S_EMB_JD_EN_SFT 15 +#define RT5682S_EMB_JD_RST (0x1 << 14) +#define RT5682S_JD_MODE (0x1 << 13) +#define RT5682S_JD_MODE_SFT 13 +#define RT5682S_DET_TYPE (0x1 << 12) +#define RT5682S_DET_TYPE_SFT 12 +#define RT5682S_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682S_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682S_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682S_SEL_FAST_OFF_MASK (0x3 << 9) +#define RT5682S_SEL_FAST_OFF_SFT 9 +#define RT5682S_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682S_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682S_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682S_FAST_OFF_MASK (0x1 << 7) +#define RT5682S_FAST_OFF_EN (0x1 << 7) +#define RT5682S_FAST_OFF_DIS (0x0 << 7) +#define RT5682S_VREF_POW_MASK (0x1 << 6) +#define RT5682S_VREF_POW_FSM (0x0 << 6) +#define RT5682S_VREF_POW_REG (0x1 << 6) +#define RT5682S_MB1_PATH_BIT 5 +#define RT5682S_MB1_PATH_MASK (0x1 << 5) +#define RT5682S_CTRL_MB1_REG (0x1 << 5) +#define RT5682S_CTRL_MB1_FSM (0x0 << 5) +#define RT5682S_MB2_PATH_BIT 4 +#define RT5682S_MB2_PATH_MASK (0x1 << 4) +#define RT5682S_CTRL_MB2_REG (0x1 << 4) +#define RT5682S_CTRL_MB2_FSM (0x0 << 4) +#define RT5682S_TRIG_JD_MASK (0x1 << 3) +#define RT5682S_TRIG_JD_HIGH (0x1 << 3) +#define RT5682S_TRIG_JD_LOW (0x0 << 3) +#define RT5682S_MIC_CAP_MASK (0x1 << 1) +#define RT5682S_MIC_CAP_HS (0x1 << 1) +#define RT5682S_MIC_CAP_HP (0x0 << 1) +#define RT5682S_MIC_CAP_SRC_MASK (0x1) +#define RT5682S_MIC_CAP_SRC_REG (0x1) +#define RT5682S_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 3 (0x0011) */ +#define RT5682S_SEL_CBJ_TYPE_SLOW (0x1 << 15) +#define RT5682S_SEL_CBJ_TYPE_NORM (0x0 << 15) +#define RT5682S_SEL_CBJ_TYPE_MASK (0x1 << 15) +#define RT5682S_POW_BG_MB1_MASK (0x1 << 13) +#define RT5682S_POW_BG_MB1_REG (0x1 << 13) +#define RT5682S_POW_BG_MB1_FSM (0x0 << 13) +#define RT5682S_POW_BG_MB2_MASK (0x1 << 12) +#define RT5682S_POW_BG_MB2_REG (0x1 << 12) +#define RT5682S_POW_BG_MB2_FSM (0x0 << 12) +#define RT5682S_EXT_JD_SRC (0x7 << 4) +#define RT5682S_EXT_JD_SRC_SFT 4 +#define RT5682S_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682S_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682S_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682S_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682S_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682S_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 4 (0x0012) */ +#define RT5682S_CBJ_IN_BUF_MASK (0x1 << 7) +#define RT5682S_CBJ_IN_BUF_EN (0x1 << 7) +#define RT5682S_CBJ_IN_BUF_DIS (0x0 << 7) +#define RT5682S_CBJ_IN_BUF_BIT 7 + +/* Combo Jack and Type Detection Control 5 (0x0013) */ +#define RT5682S_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682S_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682S_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682S_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682S_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682S_CBJ_JD_TEST_MODE (0x1 << 6) + +/* Combo Jack and Type Detection Control 6 (0x0014) */ +#define RT5682S_JD_FAST_OFF_SRC_MASK (0x7 << 8) +#define RT5682S_JD_FAST_OFF_SRC_JDH (0x6 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO6 (0x5 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO5 (0x4 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO4 (0x3 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO3 (0x2 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO2 (0x1 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO1 (0x0 << 8) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682S_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682S_DAC_L1_VOL_SFT 8 +#define RT5682S_DAC_R1_VOL_MASK (0xff) +#define RT5682S_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682S_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682S_ADC_L_VOL_SFT 8 +#define RT5682S_ADC_R_VOL_MASK (0x7f) +#define RT5682S_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682S_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682S_STO1_ADC_L_BST_SFT 14 +#define RT5682S_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682S_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682S_ST_SRC_SEL (0x1 << 8) +#define RT5682S_ST_SRC_SFT 8 +#define RT5682S_ST_EN_MASK (0x1 << 6) +#define RT5682S_ST_DIS (0x0 << 6) +#define RT5682S_ST_EN (0x1 << 6) +#define RT5682S_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682S_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682S_M_STO1_ADC_L1_SFT 15 +#define RT5682S_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682S_M_STO1_ADC_L2_SFT 14 +#define RT5682S_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682S_STO1_ADC1L_SRC_SFT 13 +#define RT5682S_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682S_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682S_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682S_STO1_ADC2L_SRC_SFT 12 +#define RT5682S_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682S_STO1_ADCL_SRC_SFT 10 +#define RT5682S_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682S_M_STO1_ADC_R1_SFT 7 +#define RT5682S_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682S_M_STO1_ADC_R2_SFT 6 +#define RT5682S_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682S_STO1_ADC1R_SRC_SFT 5 +#define RT5682S_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682S_STO1_ADC2R_SRC_SFT 4 +#define RT5682S_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682S_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682S_M_ADCMIX_L (0x1 << 15) +#define RT5682S_M_ADCMIX_L_SFT 15 +#define RT5682S_M_DAC1_L (0x1 << 14) +#define RT5682S_M_DAC1_L_SFT 14 +#define RT5682S_M_ADCMIX_R (0x1 << 7) +#define RT5682S_M_ADCMIX_R_SFT 7 +#define RT5682S_M_DAC1_R (0x1 << 6) +#define RT5682S_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682S_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682S_M_DAC_L1_STO_L_SFT 15 +#define RT5682S_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682S_G_DAC_L1_STO_L_SFT 14 +#define RT5682S_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682S_M_DAC_R1_STO_L_SFT 13 +#define RT5682S_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682S_G_DAC_R1_STO_L_SFT 12 +#define RT5682S_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682S_M_DAC_L1_STO_R_SFT 7 +#define RT5682S_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682S_G_DAC_L1_STO_R_SFT 6 +#define RT5682S_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682S_M_DAC_R1_STO_R_SFT 5 +#define RT5682S_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682S_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682S_M_ST_STO_L (0x1 << 9) +#define RT5682S_M_ST_STO_L_SFT 9 +#define RT5682S_M_ST_STO_R (0x1 << 8) +#define RT5682S_M_ST_STO_R_SFT 8 +#define RT5682S_DAC_L1_SRC_MASK (0x1 << 4) +#define RT5682S_A_DACL1_SFT 4 +#define RT5682S_DAC_R1_SRC_MASK (0x1) +#define RT5682S_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682S_IF2_DAC_SEL_MASK (0x3 << 2) +#define RT5682S_IF2_DAC_SEL_SFT 2 +#define RT5682S_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682S_IF2_ADC_SEL_SFT 0 + +/* REC Left/Right Mixer Control 2 (0x003c) */ +#define RT5682S_BST_CBJ_MASK (0x3f << 8) +#define RT5682S_BST_CBJ_SFT 8 +#define RT5682S_M_CBJ_RM1_L (0x1 << 7) +#define RT5682S_M_CBJ_RM1_L_SFT 7 +#define RT5682S_M_CBJ_RM1_R (0x1 << 6) +#define RT5682S_M_CBJ_RM1_R_SFT 6 + +/* REC Left/Right Mixer Calibration Control(0x0044) */ +#define RT5682S_PWR_RM1_R_BIT 8 +#define RT5682S_PWR_RM1_L_BIT 0 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682S_PWR_I2S1 (0x1 << 15) +#define RT5682S_PWR_I2S1_BIT 15 +#define RT5682S_PWR_I2S2 (0x1 << 14) +#define RT5682S_PWR_I2S2_BIT 14 +#define RT5682S_PRE_CHR_DAC_L1 (0x1 << 13) +#define RT5682S_PRE_CHR_DAC_L1_BIT 13 +#define RT5682S_PRE_CHR_DAC_R1 (0x1 << 12) +#define RT5682S_PRE_CHR_DAC_R1_BIT 12 +#define RT5682S_PWR_DAC_L1 (0x1 << 11) +#define RT5682S_PWR_DAC_L1_BIT 11 +#define RT5682S_PWR_DAC_R1 (0x1 << 10) +#define RT5682S_PWR_DAC_R1_BIT 10 +#define RT5682S_PWR_LDO (0x1 << 8) +#define RT5682S_PWR_LDO_BIT 8 +#define RT5682S_PWR_D2S_L (0x1 << 7) +#define RT5682S_PWR_D2S_L_BIT 7 +#define RT5682S_PWR_D2S_R (0x1 << 6) +#define RT5682S_PWR_D2S_R_BIT 6 +#define RT5682S_PWR_ADC_L1 (0x1 << 4) +#define RT5682S_PWR_ADC_L1_BIT 4 +#define RT5682S_PWR_ADC_R1 (0x1 << 3) +#define RT5682S_PWR_ADC_R1_BIT 3 +#define RT5682S_EFUSE_SW_EN (0x1 << 2) +#define RT5682S_EFUSE_SW_DIS (0x0 << 2) +#define RT5682S_PWR_EFUSE (0x1 << 1) +#define RT5682S_PWR_EFUSE_BIT 1 +#define RT5682S_DIG_GATE_CTRL (0x1 << 0) +#define RT5682S_DIG_GATE_CTRL_SFT 0 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682S_PWR_ADC_S1F (0x1 << 15) +#define RT5682S_PWR_ADC_S1F_BIT 15 +#define RT5682S_PWR_DAC_S1F (0x1 << 10) +#define RT5682S_PWR_DAC_S1F_BIT 10 +#define RT5682S_DLDO_I_LIMIT_MASK (0x1 << 7) +#define RT5682S_DLDO_I_LIMIT_EN (0x1 << 7) +#define RT5682S_DLDO_I_LIMIT_DIS (0x0 << 7) +#define RT5682S_DLDO_I_BIAS_SEL_4 (0x1 << 6) +#define RT5682S_DLDO_I_BIAS_SEL_0 (0x0 << 6) +#define RT5682S_DLDO_REG_TEST_1 (0x1 << 5) +#define RT5682S_DLDO_REG_TEST_0 (0x0 << 5) +#define RT5682S_DLDO_SRC_REG (0x1 << 4) +#define RT5682S_DLDO_SRC_EFUSE (0x0 << 4) + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682S_PWR_VREF1 (0x1 << 15) +#define RT5682S_PWR_VREF1_BIT 15 +#define RT5682S_PWR_FV1 (0x1 << 14) +#define RT5682S_PWR_FV1_BIT 14 +#define RT5682S_PWR_VREF2 (0x1 << 13) +#define RT5682S_PWR_VREF2_BIT 13 +#define RT5682S_PWR_FV2 (0x1 << 12) +#define RT5682S_PWR_FV2_BIT 12 +#define RT5682S_LDO1_DBG_MASK (0x3 << 10) +#define RT5682S_PWR_MB (0x1 << 9) +#define RT5682S_PWR_MB_BIT 9 +#define RT5682S_PWR_BG (0x1 << 7) +#define RT5682S_PWR_BG_BIT 7 +#define RT5682S_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682S_LDO1_BYPASS (0x1 << 6) +#define RT5682S_LDO1_NOT_BYPASS (0x0 << 6) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682S_PWR_MCLK0_WD (0x1 << 15) +#define RT5682S_PWR_MCLK0_WD_BIT 15 +#define RT5682S_PWR_MCLK1_WD (0x1 << 14) +#define RT5682S_PWR_MCLK1_WD_BIT 14 +#define RT5682S_RST_MCLK0 (0x1 << 13) +#define RT5682S_RST_MCLK0_BIT 13 +#define RT5682S_RST_MCLK1 (0x1 << 12) +#define RT5682S_RST_MCLK1_BIT 12 +#define RT5682S_PWR_MB1 (0x1 << 11) +#define RT5682S_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682S_PWR_MB1_BIT 11 +#define RT5682S_PWR_MB2 (0x1 << 10) +#define RT5682S_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682S_PWR_MB2_BIT 10 +#define RT5682S_PWR_JD_MASK (0x1 << 0) +#define RT5682S_PWR_JD_ENABLE (0x1 << 0) +#define RT5682S_PWR_JD_DISABLE (0x0 << 0) + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682S_PWR_LDO_PLLA (0x1 << 15) +#define RT5682S_PWR_LDO_PLLA_BIT 15 +#define RT5682S_PWR_LDO_PLLB (0x1 << 14) +#define RT5682S_PWR_LDO_PLLB_BIT 14 +#define RT5682S_PWR_BIAS_PLLA (0x1 << 13) +#define RT5682S_PWR_BIAS_PLLA_BIT 13 +#define RT5682S_PWR_BIAS_PLLB (0x1 << 12) +#define RT5682S_PWR_BIAS_PLLB_BIT 12 +#define RT5682S_PWR_CBJ (0x1 << 9) +#define RT5682S_PWR_CBJ_BIT 9 +#define RT5682S_RSTB_PLLB (0x1 << 7) +#define RT5682S_RSTB_PLLB_BIT 7 +#define RT5682S_RSTB_PLLA (0x1 << 6) +#define RT5682S_RSTB_PLLA_BIT 6 +#define RT5682S_PWR_PLLB (0x1 << 5) +#define RT5682S_PWR_PLLB_BIT 5 +#define RT5682S_PWR_PLLA (0x1 << 4) +#define RT5682S_PWR_PLLA_BIT 4 +#define RT5682S_PWR_LDO_MB2 (0x1 << 2) +#define RT5682S_PWR_LDO_MB2_BIT 2 +#define RT5682S_PWR_LDO_MB1 (0x1 << 1) +#define RT5682S_PWR_LDO_MB1_BIT 1 +#define RT5682S_PWR_BGLDO (0x1 << 0) +#define RT5682S_PWR_BGLDO_BIT 0 + +/* Power Management for Mixer (0x0066) */ +#define RT5682S_PWR_CLK_COMP_8FS (0x1 << 15) +#define RT5682S_PWR_CLK_COMP_8FS_BIT 15 +#define RT5682S_DBG_BGLDO_MASK (0x3 << 12) +#define RT5682S_DBG_BGLDO_SFT 12 +#define RT5682S_DBG_BGLDO_MB1_MASK (0x3 << 10) +#define RT5682S_DBG_BGLDO_MB1_SFT 10 +#define RT5682S_DBG_BGLDO_MB2_MASK (0x3 << 8) +#define RT5682S_DBG_BGLDO_MB2_SFT 8 +#define RT5682S_DLDO_BGLDO_MASK (0x3 << 6) +#define RT5682S_DLDO_BGLDO_MB2_SFT 6 +#define RT5682S_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682S_PWR_STO1_DAC_L_BIT 5 +#define RT5682S_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682S_PWR_STO1_DAC_R_BIT 4 +#define RT5682S_DVO_BGLDO_MB1_MASK (0x3 << 2) +#define RT5682S_DVO_BGLDO_MB1_SFT 2 +#define RT5682S_DVO_BGLDO_MB2_MASK (0x3 << 0) + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682S_SYS_CLK_DET (0x1 << 15) +#define RT5682S_SYS_CLK_DET_SFT 15 +#define RT5682S_PLL1_CLK_DET (0x1 << 14) +#define RT5682S_PLL1_CLK_DET_SFT 14 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682S_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682S_DMIC_1_EN_SFT 15 +#define RT5682S_DMIC_1_DIS (0x0 << 15) +#define RT5682S_DMIC_1_EN (0x1 << 15) +#define RT5682S_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682S_FIFO_CLK_DIV_2 (0x1 << 12) +#define RT5682S_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682S_DMIC_1_DP_SFT 4 +#define RT5682S_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682S_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682S_DMIC_CLK_MASK (0xf << 0) +#define RT5682S_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682S_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682S_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682S_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682S_SEL_ADCDAT_SFT 15 +#define RT5682S_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682S_I2S1_TX_CHL_SFT 12 +#define RT5682S_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682S_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682S_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682S_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682S_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682S_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682S_I2S1_RX_CHL_SFT 8 +#define RT5682S_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682S_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682S_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682S_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682S_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682S_I2S1_MONO_MASK (0x1 << 7) +#define RT5682S_I2S1_MONO_EN (0x1 << 7) +#define RT5682S_I2S1_MONO_DIS (0x0 << 7) +#define RT5682S_I2S1_DL_MASK (0x7 << 4) +#define RT5682S_I2S1_DL_SFT 4 +#define RT5682S_I2S1_DL_16 (0x0 << 4) +#define RT5682S_I2S1_DL_20 (0x1 << 4) +#define RT5682S_I2S1_DL_24 (0x2 << 4) +#define RT5682S_I2S1_DL_32 (0x3 << 4) +#define RT5682S_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0071) */ +#define RT5682S_I2S2_MS_MASK (0x1 << 15) +#define RT5682S_I2S2_MS_SFT 15 +#define RT5682S_I2S2_MS_M (0x0 << 15) +#define RT5682S_I2S2_MS_S (0x1 << 15) +#define RT5682S_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682S_I2S2_PIN_CFG_SFT 14 +#define RT5682S_I2S2_OUT_MASK (0x1 << 9) +#define RT5682S_I2S2_OUT_SFT 9 +#define RT5682S_I2S2_OUT_UM (0x0 << 9) +#define RT5682S_I2S2_OUT_M (0x1 << 9) +#define RT5682S_I2S_BP_MASK (0x1 << 8) +#define RT5682S_I2S_BP_SFT 8 +#define RT5682S_I2S_BP_NOR (0x0 << 8) +#define RT5682S_I2S_BP_INV (0x1 << 8) +#define RT5682S_I2S2_MONO_MASK (0x1 << 7) +#define RT5682S_I2S2_MONO_EN (0x1 << 7) +#define RT5682S_I2S2_MONO_DIS (0x0 << 7) +#define RT5682S_I2S2_DL_MASK (0x7 << 4) +#define RT5682S_I2S2_DL_SFT 4 +#define RT5682S_I2S2_DL_8 (0x0 << 4) +#define RT5682S_I2S2_DL_16 (0x1 << 4) +#define RT5682S_I2S2_DL_20 (0x2 << 4) +#define RT5682S_I2S2_DL_24 (0x3 << 4) +#define RT5682S_I2S2_DL_32 (0x4 << 4) +#define RT5682S_I2S_DF_MASK (0x7) +#define RT5682S_I2S_DF_SFT 0 +#define RT5682S_I2S_DF_I2S (0x0) +#define RT5682S_I2S_DF_LEFT (0x1) +#define RT5682S_I2S_DF_PCM_A (0x2) +#define RT5682S_I2S_DF_PCM_B (0x3) +#define RT5682S_I2S_DF_PCM_A_N (0x6) +#define RT5682S_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682S_ADC_OSR_MASK (0xf << 12) +#define RT5682S_ADC_OSR_SFT 12 +#define RT5682S_ADC_OSR_D_1 (0x0 << 12) +#define RT5682S_ADC_OSR_D_2 (0x1 << 12) +#define RT5682S_ADC_OSR_D_4 (0x2 << 12) +#define RT5682S_ADC_OSR_D_6 (0x3 << 12) +#define RT5682S_ADC_OSR_D_8 (0x4 << 12) +#define RT5682S_ADC_OSR_D_12 (0x5 << 12) +#define RT5682S_ADC_OSR_D_16 (0x6 << 12) +#define RT5682S_ADC_OSR_D_24 (0x7 << 12) +#define RT5682S_ADC_OSR_D_32 (0x8 << 12) +#define RT5682S_ADC_OSR_D_48 (0x9 << 12) +#define RT5682S_I2S_M_D_MASK (0xf << 8) +#define RT5682S_I2S_M_D_SFT 8 +#define RT5682S_I2S_M_D_1 (0x0 << 8) +#define RT5682S_I2S_M_D_2 (0x1 << 8) +#define RT5682S_I2S_M_D_3 (0x2 << 8) +#define RT5682S_I2S_M_D_4 (0x3 << 8) +#define RT5682S_I2S_M_D_6 (0x4 << 8) +#define RT5682S_I2S_M_D_8 (0x5 << 8) +#define RT5682S_I2S_M_D_12 (0x6 << 8) +#define RT5682S_I2S_M_D_16 (0x7 << 8) +#define RT5682S_I2S_M_D_24 (0x8 << 8) +#define RT5682S_I2S_M_D_32 (0x9 << 8) +#define RT5682S_I2S_M_D_48 (0x10 << 8) +#define RT5682S_I2S_M_CLK_SRC_MASK (0x7 << 4) +#define RT5682S_I2S_M_CLK_SRC_SFT 4 +#define RT5682S_DAC_OSR_MASK (0xf << 0) +#define RT5682S_DAC_OSR_SFT 0 +#define RT5682S_DAC_OSR_D_1 (0x0 << 0) +#define RT5682S_DAC_OSR_D_2 (0x1 << 0) +#define RT5682S_DAC_OSR_D_4 (0x2 << 0) +#define RT5682S_DAC_OSR_D_6 (0x3 << 0) +#define RT5682S_DAC_OSR_D_8 (0x4 << 0) +#define RT5682S_DAC_OSR_D_12 (0x5 << 0) +#define RT5682S_DAC_OSR_D_16 (0x6 << 0) +#define RT5682S_DAC_OSR_D_24 (0x7 << 0) +#define RT5682S_DAC_OSR_D_32 (0x8 << 0) +#define RT5682S_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682S_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682S_I2S2_BCLK_MS2_SFT 11 +#define RT5682S_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682S_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682S_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682S_TDM_TX_CH_2 (0x0 << 12) +#define RT5682S_TDM_TX_CH_4 (0x1 << 12) +#define RT5682S_TDM_TX_CH_6 (0x2 << 12) +#define RT5682S_TDM_TX_CH_8 (0x3 << 12) +#define RT5682S_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682S_TDM_RX_CH_2 (0x0 << 8) +#define RT5682S_TDM_RX_CH_4 (0x1 << 8) +#define RT5682S_TDM_RX_CH_6 (0x2 << 8) +#define RT5682S_TDM_RX_CH_8 (0x3 << 8) +#define RT5682S_TDM_ADC_LCA_MASK (0x7 << 4) +#define RT5682S_TDM_ADC_LCA_SFT 4 +#define RT5682S_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682S_IF1_ADC1_SEL_SFT 14 +#define RT5682S_IF1_ADC2_SEL_SFT 12 +#define RT5682S_IF1_ADC3_SEL_SFT 10 +#define RT5682S_IF1_ADC4_SEL_SFT 8 +#define RT5682S_TDM_ADC_SEL_SFT 3 + +/* TDM control 3 (0x007b) */ +#define RT5682S_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682S_TDM_S_BP_MASK (0x1 << 15) +#define RT5682S_TDM_S_BP_SFT 15 +#define RT5682S_TDM_S_BP_NOR (0x0 << 15) +#define RT5682S_TDM_S_BP_INV (0x1 << 15) +#define RT5682S_TDM_S_LP_MASK (0x1 << 14) +#define RT5682S_TDM_S_LP_SFT 14 +#define RT5682S_TDM_S_LP_NOR (0x0 << 14) +#define RT5682S_TDM_S_LP_INV (0x1 << 14) +#define RT5682S_TDM_DF_MASK (0x7 << 11) +#define RT5682S_TDM_DF_SFT 11 +#define RT5682S_TDM_DF_I2S (0x0 << 11) +#define RT5682S_TDM_DF_LEFT (0x1 << 11) +#define RT5682S_TDM_DF_PCM_A (0x2 << 11) +#define RT5682S_TDM_DF_PCM_B (0x3 << 11) +#define RT5682S_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682S_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682S_TDM_BCLK_MS1_MASK (0x3 << 8) +#define RT5682S_TDM_BCLK_MS1_SFT 8 +#define RT5682S_TDM_BCLK_MS1_32 (0x0 << 8) +#define RT5682S_TDM_BCLK_MS1_64 (0x1 << 8) +#define RT5682S_TDM_BCLK_MS1_128 (0x2 << 8) +#define RT5682S_TDM_BCLK_MS1_256 (0x3 << 8) +#define RT5682S_TDM_BCLK_MS1_16 (0x4 << 8) +#define RT5682S_TDM_CL_MASK (0x3 << 4) +#define RT5682S_TDM_CL_16 (0x0 << 4) +#define RT5682S_TDM_CL_20 (0x1 << 4) +#define RT5682S_TDM_CL_24 (0x2 << 4) +#define RT5682S_TDM_CL_32 (0x3 << 4) +#define RT5682S_TDM_M_BP_MASK (0x1 << 2) +#define RT5682S_TDM_M_BP_SFT 2 +#define RT5682S_TDM_M_BP_NOR (0x0 << 2) +#define RT5682S_TDM_M_BP_INV (0x1 << 2) +#define RT5682S_TDM_M_LP_MASK (0x1 << 1) +#define RT5682S_TDM_M_LP_SFT 1 +#define RT5682S_TDM_M_LP_NOR (0x0 << 1) +#define RT5682S_TDM_M_LP_INV (0x1 << 1) +#define RT5682S_TDM_MS_MASK (0x1 << 0) +#define RT5682S_TDM_MS_SFT 0 +#define RT5682S_TDM_MS_S (0x0 << 0) +#define RT5682S_TDM_MS_M (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682S_SCLK_SRC_MASK (0x7 << 13) +#define RT5682S_SCLK_SRC_SFT 13 +#define RT5682S_PLL_SRC_MASK (0x3 << 8) +#define RT5682S_PLL_SRC_SFT 8 +#define RT5682S_PLL_SRC_MCLK (0x0 << 8) +#define RT5682S_PLL_SRC_BCLK1 (0x1 << 8) +#define RT5682S_PLL_SRC_RC (0x3 << 8) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682S_DA_ASRC_MASK (0x1 << 13) +#define RT5682S_DA_ASRC_SFT 13 +#define RT5682S_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682S_DAC_STO1_ASRC_SFT 12 +#define RT5682S_AD_ASRC_MASK (0x1 << 8) +#define RT5682S_AD_ASRC_SFT 8 +#define RT5682S_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682S_AD_ASRC_SEL_SFT 4 +#define RT5682S_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682S_DMIC_ASRC_SFT 3 +#define RT5682S_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682S_ADC_STO1_ASRC_SFT 2 +#define RT5682S_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682S_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682S_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682S_FILTER_CLK_SEL_SFT 12 +#define RT5682S_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682S_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682S_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682S_ASRCIN_FTK_N1_SFT 14 +#define RT5682S_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682S_ASRCIN_FTK_N2_SFT 12 +#define RT5682S_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682S_ASRCIN_FTK_M1_SFT 8 +#define RT5682S_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682S_ASRCIN_FTK_M2_SFT 4 + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682S_OUT_HP_L_EN (0x1 << 6) +#define RT5682S_OUT_HP_R_EN (0x1 << 5) +#define RT5682S_LDO_PUMP_EN (0x1 << 4) +#define RT5682S_LDO_PUMP_EN_SFT 4 +#define RT5682S_PUMP_EN (0x1 << 3) +#define RT5682S_PUMP_EN_SFT 3 +#define RT5682S_CAPLESS_L_EN (0x1 << 1) +#define RT5682S_CAPLESS_L_EN_SFT 1 +#define RT5682S_CAPLESS_R_EN (0x1 << 0) +#define RT5682S_CAPLESS_R_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682S_RAMP_MASK (0x1 << 12) +#define RT5682S_RAMP_SFT 12 +#define RT5682S_RAMP_DIS (0x0 << 12) +#define RT5682S_RAMP_EN (0x1 << 12) +#define RT5682S_BPS_MASK (0x1 << 11) +#define RT5682S_BPS_SFT 11 +#define RT5682S_BPS_DIS (0x0 << 11) +#define RT5682S_BPS_EN (0x1 << 11) +#define RT5682S_FAST_UPDN_MASK (0x1 << 10) +#define RT5682S_FAST_UPDN_SFT 10 +#define RT5682S_FAST_UPDN_DIS (0x0 << 10) +#define RT5682S_FAST_UPDN_EN (0x1 << 10) +#define RT5682S_VLO_MASK (0x1 << 7) +#define RT5682S_VLO_SFT 7 +#define RT5682S_VLO_3V (0x0 << 7) +#define RT5682S_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682S_OSW_L_MASK (0x1 << 11) +#define RT5682S_OSW_L_SFT 11 +#define RT5682S_OSW_L_DIS (0x0 << 11) +#define RT5682S_OSW_L_EN (0x1 << 11) +#define RT5682S_OSW_R_MASK (0x1 << 10) +#define RT5682S_OSW_R_SFT 10 +#define RT5682S_OSW_R_DIS (0x0 << 10) +#define RT5682S_OSW_R_EN (0x1 << 10) +#define RT5682S_PM_HP_MASK (0x3 << 8) +#define RT5682S_PM_HP_SFT 8 +#define RT5682S_PM_HP_LV (0x0 << 8) +#define RT5682S_PM_HP_MV (0x1 << 8) +#define RT5682S_PM_HP_HV (0x2 << 8) + +/* Micbias Control1 (0x93) */ +#define RT5682S_MIC1_OV_MASK (0x3 << 14) +#define RT5682S_MIC1_OV_SFT 14 +#define RT5682S_MIC1_OV_2V7 (0x0 << 14) +#define RT5682S_MIC1_OV_2V4 (0x1 << 14) +#define RT5682S_MIC1_OV_2V25 (0x3 << 14) +#define RT5682S_MIC1_OV_1V8 (0x4 << 14) +#define RT5682S_MIC2_OV_MASK (0x3 << 8) +#define RT5682S_MIC2_OV_SFT 8 +#define RT5682S_MIC2_OV_2V7 (0x0 << 8) +#define RT5682S_MIC2_OV_2V4 (0x1 << 8) +#define RT5682S_MIC2_OV_2V25 (0x3 << 8) +#define RT5682S_MIC2_OV_1V8 (0x4 << 8) + +/* Micbias Control2 (0x0094) */ +#define RT5682S_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682S_PWR_CLK25M_SFT 9 +#define RT5682S_PWR_CLK25M_PD (0x0 << 9) +#define RT5682S_PWR_CLK25M_PU (0x1 << 9) +#define RT5682S_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682S_PWR_CLK1M_SFT 8 +#define RT5682S_PWR_CLK1M_PD (0x0 << 8) +#define RT5682S_PWR_CLK1M_PU (0x1 << 8) + +/* PLL M/N/K Code Control 1 (0x0098) */ +#define RT5682S_PLLA_N_MASK (0x1ff << 0) + +/* PLL M/N/K Code Control 2 (0x0099) */ +#define RT5682S_PLLA_M_MASK (0x1f << 8) +#define RT5682S_PLLA_M_SFT 8 +#define RT5682S_PLLA_K_MASK (0x1f << 0) + +/* PLL M/N/K Code Control 3 (0x009a) */ +#define RT5682S_PLLB_N_MASK (0x3ff << 0) + +/* PLL M/N/K Code Control 4 (0x009b) */ +#define RT5682S_PLLB_M_MASK (0x1f << 8) +#define RT5682S_PLLB_M_SFT 8 +#define RT5682S_PLLB_K_MASK (0x1f << 0) + +/* PLL M/N/K Code Control 6 (0x009d) */ +#define RT5682S_PLLB_SEL_PS_MASK (0x1 << 13) +#define RT5682S_PLLB_SEL_PS_SFT 13 +#define RT5682S_PLLB_BYP_PS_MASK (0x1 << 12) +#define RT5682S_PLLB_BYP_PS_SFT 12 +#define RT5682S_PLLB_M_BP_MASK (0x1 << 11) +#define RT5682S_PLLB_M_BP_SFT 11 +#define RT5682S_PLLB_K_BP_MASK (0x1 << 10) +#define RT5682S_PLLB_K_BP_SFT 10 +#define RT5682S_PLLA_M_BP_MASK (0x1 << 7) +#define RT5682S_PLLA_M_BP_SFT 7 +#define RT5682S_PLLA_K_BP_MASK (0x1 << 6) +#define RT5682S_PLLA_K_BP_SFT 6 + +/* PLL M/N/K Code Control 7 (0x009e) */ +#define RT5682S_PLLB_SRC_MASK (0x3 << 0) +#define RT5682S_PLLB_SRC_DFIN (0x1) +#define RT5682S_PLLB_SRC_PLLA (0x0) + +/* RC Clock Control (0x009f) */ +#define RT5682S_POW_IRQ (0x1 << 15) +#define RT5682S_POW_JDH (0x1 << 14) + +/* I2S2 Master Mode Clock Control 1 (0x00a0) */ +#define RT5682S_I2S2_M_CLK_SRC_MASK (0x7 << 4) +#define RT5682S_I2S2_M_CLK_SRC_SFT 4 +#define RT5682S_I2S2_M_D_MASK (0xf << 0) +#define RT5682S_I2S2_M_D_1 (0x0) +#define RT5682S_I2S2_M_D_2 (0x1) +#define RT5682S_I2S2_M_D_3 (0x2) +#define RT5682S_I2S2_M_D_4 (0x3) +#define RT5682S_I2S2_M_D_6 (0x4) +#define RT5682S_I2S2_M_D_8 (0x5) +#define RT5682S_I2S2_M_D_12 (0x6) +#define RT5682S_I2S2_M_D_16 (0x7) +#define RT5682S_I2S2_M_D_24 (0x8) +#define RT5682S_I2S2_M_D_32 (0x9) +#define RT5682S_I2S2_M_D_48 (0xa) +#define RT5682S_I2S2_M_D_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682S_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682S_JD1_PULSE_EN_SFT 10 +#define RT5682S_JD1_PULSE_DIS (0x0 << 10) +#define RT5682S_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682S_JD1_EN_MASK (0x1 << 15) +#define RT5682S_JD1_EN_SFT 15 +#define RT5682S_JD1_DIS (0x0 << 15) +#define RT5682S_JD1_EN (0x1 << 15) +#define RT5682S_JD1_POL_MASK (0x1 << 13) +#define RT5682S_JD1_POL_NOR (0x0 << 13) +#define RT5682S_JD1_POL_INV (0x1 << 13) +#define RT5682S_JD1_IRQ_MASK (0x1 << 10) +#define RT5682S_JD1_IRQ_LEV (0x0 << 10) +#define RT5682S_JD1_IRQ_PUL (0x1 << 10) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682S_IL_IRQ_MASK (0x1 << 7) +#define RT5682S_IL_IRQ_DIS (0x0 << 7) +#define RT5682S_IL_IRQ_EN (0x1 << 7) +#define RT5682S_IL_IRQ_TYPE_MASK (0x1 << 4) +#define RT5682S_IL_IRQ_LEV (0x0 << 4) +#define RT5682S_IL_IRQ_PUL (0x1 << 4) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682S_GP1_PIN_MASK (0x3 << 14) +#define RT5682S_GP1_PIN_SFT 14 +#define RT5682S_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682S_GP1_PIN_IRQ (0x1 << 14) +#define RT5682S_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682S_GP2_PIN_MASK (0x3 << 12) +#define RT5682S_GP2_PIN_SFT 12 +#define RT5682S_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682S_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682S_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682S_GP3_PIN_MASK (0x3 << 10) +#define RT5682S_GP3_PIN_SFT 10 +#define RT5682S_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682S_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682S_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682S_GP4_PIN_MASK (0x3 << 8) +#define RT5682S_GP4_PIN_SFT 8 +#define RT5682S_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682S_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682S_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682S_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682S_GP5_PIN_MASK (0x3 << 6) +#define RT5682S_GP5_PIN_SFT 6 +#define RT5682S_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682S_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682S_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682S_GP6_PIN_MASK (0x1 << 5) +#define RT5682S_GP6_PIN_SFT 5 +#define RT5682S_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682S_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682S_GP1_PF_MASK (0x1 << 15) +#define RT5682S_GP1_PF_IN (0x0 << 15) +#define RT5682S_GP1_PF_OUT (0x1 << 15) +#define RT5682S_GP1_OUT_MASK (0x1 << 14) +#define RT5682S_GP1_OUT_L (0x0 << 14) +#define RT5682S_GP1_OUT_H (0x1 << 14) +#define RT5682S_GP2_PF_MASK (0x1 << 13) +#define RT5682S_GP2_PF_IN (0x0 << 13) +#define RT5682S_GP2_PF_OUT (0x1 << 13) +#define RT5682S_GP2_OUT_MASK (0x1 << 12) +#define RT5682S_GP2_OUT_L (0x0 << 12) +#define RT5682S_GP2_OUT_H (0x1 << 12) +#define RT5682S_GP3_PF_MASK (0x1 << 11) +#define RT5682S_GP3_PF_IN (0x0 << 11) +#define RT5682S_GP3_PF_OUT (0x1 << 11) +#define RT5682S_GP3_OUT_MASK (0x1 << 10) +#define RT5682S_GP3_OUT_L (0x0 << 10) +#define RT5682S_GP3_OUT_H (0x1 << 10) +#define RT5682S_GP4_PF_MASK (0x1 << 9) +#define RT5682S_GP4_PF_IN (0x0 << 9) +#define RT5682S_GP4_PF_OUT (0x1 << 9) +#define RT5682S_GP4_OUT_MASK (0x1 << 8) +#define RT5682S_GP4_OUT_L (0x0 << 8) +#define RT5682S_GP4_OUT_H (0x1 << 8) +#define RT5682S_GP5_PF_MASK (0x1 << 7) +#define RT5682S_GP5_PF_IN (0x0 << 7) +#define RT5682S_GP5_PF_OUT (0x1 << 7) +#define RT5682S_GP5_OUT_MASK (0x1 << 6) +#define RT5682S_GP5_OUT_L (0x0 << 6) +#define RT5682S_GP5_OUT_H (0x1 << 6) +#define RT5682S_GP6_PF_MASK (0x1 << 5) +#define RT5682S_GP6_PF_IN (0x0 << 5) +#define RT5682S_GP6_PF_OUT (0x1 << 5) +#define RT5682S_GP6_OUT_MASK (0x1 << 4) +#define RT5682S_GP6_OUT_L (0x0 << 4) +#define RT5682S_GP6_OUT_H (0x1 << 4) + +/* GPIO Status (0x00c2) */ +#define RT5682S_GP6_ST (0x1 << 6) +#define RT5682S_GP5_ST (0x1 << 5) +#define RT5682S_GP4_ST (0x1 << 4) +#define RT5682S_GP3_ST (0x1 << 3) +#define RT5682S_GP2_ST (0x1 << 2) +#define RT5682S_GP1_ST (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682S_ZCD_MASK (0x1 << 10) +#define RT5682S_ZCD_SFT 10 +#define RT5682S_ZCD_PD (0x0 << 10) +#define RT5682S_ZCD_PU (0x1 << 10) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682S_4BTN_IL_MASK (0x1 << 15) +#define RT5682S_4BTN_IL_EN (0x1 << 15) +#define RT5682S_4BTN_IL_DIS (0x0 << 15) +#define RT5682S_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682S_4BTN_IL_NOR (0x1 << 14) +#define RT5682S_4BTN_IL_RST (0x0 << 14) + +/* 4 Button Inline Command Control 3~6 (0x00e5~0x00e8) */ +#define RT5682S_4BTN_IL_HOLD_WIN_MASK (0x7f << 8) +#define RT5682S_4BTN_IL_HOLD_WIN_SFT 8 +#define RT5682S_4BTN_IL_CLICK_WIN_MASK (0x7f) +#define RT5682S_4BTN_IL_CLICK_WIN_SFT 0 + +/* Analog JD Control (0x00f0) */ +#define RT5682S_JDH_RS_MASK (0x1 << 4) +#define RT5682S_JDH_NO_PLUG (0x1 << 4) +#define RT5682S_JDH_PLUG (0x0 << 4) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682S_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682S_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682S_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682S_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682S_CP_CLK_HP_600KHZ (0x3 << 4) + +/* Pad Driving Control (0x0136) */ +#define RT5682S_PAD_DRV_GP1_MASK (0x1 << 14) +#define RT5682S_PAD_DRV_GP1_HIGH (0x1 << 14) +#define RT5682S_PAD_DRV_GP1_LOW (0x0 << 14) +#define RT5682S_PAD_DRV_GP2_MASK (0x1 << 12) +#define RT5682S_PAD_DRV_GP2_HIGH (0x1 << 12) +#define RT5682S_PAD_DRV_GP2_LOW (0x0 << 12) +#define RT5682S_PAD_DRV_GP3_MASK (0x1 << 10) +#define RT5682S_PAD_DRV_GP3_HIGH (0x1 << 10) +#define RT5682S_PAD_DRV_GP3_LOW (0x0 << 10) +#define RT5682S_PAD_DRV_GP4_MASK (0x1 << 8) +#define RT5682S_PAD_DRV_GP4_HIGH (0x1 << 8) +#define RT5682S_PAD_DRV_GP4_LOW (0x0 << 8) +#define RT5682S_PAD_DRV_GP5_MASK (0x1 << 6) +#define RT5682S_PAD_DRV_GP5_HIGH (0x1 << 6) +#define RT5682S_PAD_DRV_GP5_LOW (0x0 << 6) +#define RT5682S_PAD_DRV_GP6_MASK (0x1 << 4) +#define RT5682S_PAD_DRV_GP6_HIGH (0x1 << 4) +#define RT5682S_PAD_DRV_GP6_LOW (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682S_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682S_CKXEN_DAC1_SFT 13 +#define RT5682S_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682S_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682S_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682S_CKXEN_ADC1_SFT 13 +#define RT5682S_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682S_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682S_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682S_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682S_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682S_AD2DA_LB_MASK (0x1 << 10) +#define RT5682S_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682S_NG2_EN_MASK (0x1 << 15) +#define RT5682S_NG2_EN (0x1 << 15) +#define RT5682S_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682S_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682S_DEB_80_MS (0x0 << 4) + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5682S_HP_SIG_SRC_MASK (0x3) +#define RT5682S_HP_SIG_SRC_1BIT_CTL (0x3) +#define RT5682S_HP_SIG_SRC_REG (0x2) +#define RT5682S_HP_SIG_SRC_IMPE_REG (0x1) +#define RT5682S_HP_SIG_SRC_DC_CALI (0x0) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682S_SAR_BUTDET_MASK (0x1 << 15) +#define RT5682S_SAR_BUTDET_EN (0x1 << 15) +#define RT5682S_SAR_BUTDET_DIS (0x0 << 15) +#define RT5682S_SAR_BUTDET_POW_MASK (0x1 << 14) +#define RT5682S_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682S_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682S_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682S_SAR_BUTDET_RST_NORM (0x1 << 13) +#define RT5682S_SAR_BUTDET_RST (0x0 << 13) +#define RT5682S_SAR_POW_MASK (0x1 << 12) +#define RT5682S_SAR_POW_EN (0x1 << 12) +#define RT5682S_SAR_POW_DIS (0x0 << 12) +#define RT5682S_SAR_RST_MASK (0x1 << 11) +#define RT5682S_SAR_RST_NORMAL (0x1 << 11) +#define RT5682S_SAR_RST (0x0 << 11) +#define RT5682S_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682S_SAR_BYPASS_EN (0x1 << 10) +#define RT5682S_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682S_SAR_SEL_MB1_2_MASK (0x3 << 8) +#define RT5682S_SAR_SEL_MB1_2_SFT 8 +#define RT5682S_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682S_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682S_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682S_SAR_SEL_MB1_2_CTL_MASK (0x1 << 5) +#define RT5682S_SAR_SEL_MB1_2_AUTO (0x1 << 5) +#define RT5682S_SAR_SEL_MB1_2_MANU (0x0 << 5) +#define RT5682S_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682S_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682S_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 2 (0x0211) */ +#define RT5682S_SAR_ADC_PSV_MASK (0x1 << 4) +#define RT5682S_SAR_ADC_PSV_ENTRY (0x1 << 4) + + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682S_SAR_SOUR_MASK (0x3f) +#define RT5682S_SAR_SOUR_BTN (0x3f) +#define RT5682S_SAR_SOUR_TYPE (0x0) + + +#define RT5682S_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +/* System Clock Source */ +enum { + RT5682S_SCLK_S_MCLK, + RT5682S_SCLK_S_PLL1, + RT5682S_SCLK_S_PLL2, + RT5682S_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682S_PLL_S_MCLK, + RT5682S_PLL_S_BCLK1, + RT5682S_PLL_S_BCLK2, + RT5682S_PLL_S_RCCLK, +}; + +enum { + RT5682S_PLL1, + RT5682S_PLL2, + RT5682S_PLLS, +}; + +enum { + RT5682S_AIF1, + RT5682S_AIF2, + RT5682S_AIFS +}; + +/* filter mask */ +enum { + RT5682S_DA_STEREO1_FILTER = 0x1, + RT5682S_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682S_CLK_SEL_SYS, + RT5682S_CLK_SEL_I2S1_ASRC, + RT5682S_CLK_SEL_I2S2_ASRC, +}; + +enum { + USE_PLLA, + USE_PLLB, + USE_PLLAB, +}; + +struct pll_calc_map { + unsigned int freq_in; + unsigned int freq_out; + int m; + int n; + int k; + bool m_bp; + bool k_bp; + bool byp_ps; + bool sel_ps; +}; + +#define RT5682S_NUM_SUPPLIES 2 + +struct rt5682s_priv { + struct snd_soc_component *component; + struct rt5682s_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682S_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + struct mutex sar_mutex; + +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682S_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + + int sysclk; + int sysclk_src; + int lrck[RT5682S_AIFS]; + int bclk[RT5682S_AIFS]; + int master[RT5682S_AIFS]; + + int pll_src[RT5682S_PLLS]; + int pll_in[RT5682S_PLLS]; + int pll_out[RT5682S_PLLS]; + int pll_comb; + + int jack_type; + int irq_work_delay_time; +}; + +int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682S_H__ */ -- 2.17.1

3 3

Problems with ALSA MIDI raw devices after upgrading to kernel 5.14
by GitHub issues - opened 19 Sep '21

19 Sep '21

alsa-project/alsa-lib issue #178 was opened from lentferj: I installed 5.14 kernel from this source (https://github.com/geoffreybennett/scarlett-gen2/releases) on Debian Buster to test out the better USB audio latency. I found that both Bitwig and Ardour6 cannot start anymore when using ALSA MIDI raw devices. In Bitwig the sound engine keeps crashing and Ardour6 doesn't start at all when choosing raw devices. In Ardour6 I see this: ``` ALSA lib rawmidi_hw.c:100:(snd_rawmidi_hw_params) SNDRV_RAWMIDI_IOCTL_PARAMS failed: Invalid argument ardour-6.9.0: rawmidi.c:268: snd_rawmidi_open_conf: Assertion `err >= 0' failed. Aborted ``` I also tried to access devices directly with amidi, same result. ``` $ amidi --dump -p hw:5 ALSA lib rawmidi_hw.c:100:(snd_rawmidi_hw_params) SNDRV_RAWMIDI_IOCTL_PARAMS failed: Invalid argument amidi: rawmidi.c:268: snd_rawmidi_open_conf: Assertion `err >= 0' failed. Aborted ``` I manually upgraded alsa-lib and alsa-utils to 1.2.5, but I still have the same problem: ``` $ which amidi /usr/bin/amidi lentferj@ts-d20:~$ ls -l /usr/bin/amidi -rwxr-xr-x 1 root root 57064 Sep 19 18:34 /usr/bin/amidi lentferj@ts-d20:~$ ldd /usr/bin/amidi linux-vdso.so.1 (0x00007ffc50c9b000) libasound.so.2 => /usr/lib/libasound.so.2 (0x00007f9c97c85000) libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f9c97b02000) libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f9c97afd000) libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007f9c97adc000) libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f9c9791b000) librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f9c97911000) /lib64/ld-linux-x86-64.so.2 (0x00007f9c97dc0000) lentferj@ts-d20:~$ ls -l /usr/lib/libasound.so.2 lrwxrwxrwx 1 root root 18 Sep 19 18:29 /usr/lib/libasound.so.2 -> libasound.so.2.0.0 lentferj@ts-d20:~$ ls -l /usr/lib/libasound.so.2.0.0 -rwxr-xr-x 1 root root 5242136 Sep 19 18:29 /usr/lib/libasound.so.2.0.0 lentferj@ts-d20:~$ uname -a Linux ts-d20 5.14.0 0000021 SMP Mon Aug 30 00:46:47 UTC 2021 x86_64 GNU/Linux lentferj@ts-d20:~$ amidi -l Dir Device Name IO hw:2,0,0 Saffire 6USB2.0 MIDI 1 IO hw:3,0,0 Neutron(1) MIDI 1 IO hw:4,0,0 Pulse2 MIDI 1 IO hw:5,0,0 LPK25 MIDI 1 lentferj@ts-d20:~$ amidi --dump -p hw:5 ALSA lib rawmidi_hw.c:100:(snd_rawmidi_hw_params) SNDRV_RAWMIDI_IOCTL_PARAMS failed: Invalid argument amidi: rawmidi.c:256: snd_rawmidi_open_conf: Assertion `err >= 0' failed. Abgebrochen ``` When booting back to a 5.10 kernel the problems disappear. I also reported this on the Ardour Tracker ( https://tracker.ardour.org/view.php?id=8800 ). Issue URL : https://github.com/alsa-project/alsa-lib/issues/178 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH] ASoC: mediatek: SND_SOC_MT8195 remove depends on COMPILE_TEST
by Geert Uytterhoeven 19 Sep '21

19 Sep '21

Hi Jackie, On Sat, Sep 18, 2021 at 7:17 AM Jackie Liu <liu.yun(a)linux.dev> wrote: > From: Jackie Liu <liuyun01(a)kylinos.cn> > > After compiling with COMPILE_TEST, we are most likely to compile on a > different architecture, such as x86. At this time, it is possible that > the HAVE_CLK macro is not selected, resulting in compilation failure. > > Avoid fail like: > > Kernel: arch/x86/boot/bzImage is ready (#17) > ERROR: modpost: "clkdev_add" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! > ERROR: modpost: "clkdev_drop" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! > ERROR: modpost: "clk_unregister_gate" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! > ERROR: modpost: "clk_register_gate" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! > make[1]: *** [scripts/Makefile.modpost:134: modules-only.symvers] Error 1 > make[1]: *** Deleting file 'modules-only.symvers' > make: *** [Makefile:1783: modules] Error 2 > > Cc: Geert Uytterhoeven <geert+renesas(a)glider.be> > Fixes: 940ffa194547 ("ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK") > Signed-off-by: Jackie Liu <liuyun01(a)kylinos.cn> Thanks for your patch! > --- a/sound/soc/mediatek/Kconfig > +++ b/sound/soc/mediatek/Kconfig > @@ -187,7 +187,7 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 > > config SND_SOC_MT8195 > tristate "ASoC support for Mediatek MT8195 chip" > - depends on ARCH_MEDIATEK || COMPILE_TEST > + depends on ARCH_MEDIATEK Instead of dropping support for compile-testing, wouldn't it be better to add the missing real dependencies? HAVE_CLK (clkdev_add, clkdev_drop) COMMON_CLK (clk_unregister_gate, clk_register_gate) As the latter selects the former, adding the line depends on COMMON_CLK Also, the Fixes line is wrong as the link failure happened before. Fixes: 6746cc858259985a ("ASoC: mediatek: mt8195: add platform driver") > select SND_SOC_MEDIATEK > help > This adds ASoC platform driver support for Mediatek MT8195 chip Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

1 0

[PATCH 2/2] ASoC: codecs: max98390: simplify getting the adapter of a client
by Wolfram Sang 19 Sep '21

19 Sep '21

We have a dedicated pointer for that, so use it. Much easier to read and less computation involved. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- Please apply to your subsystem tree. sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b392567c2b3e..d1882cbc9381 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1021,7 +1021,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c, int reg = 0; struct max98390_priv *max98390 = NULL; - struct i2c_adapter *adapter = to_i2c_adapter(i2c->dev.parent); + struct i2c_adapter *adapter = i2c->adapter; ret = i2c_check_functionality(adapter, I2C_FUNC_SMBUS_BYTE -- 2.30.2

1 0

[PATCH v3 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 17 Sep '21

17 Sep '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. This patch set depends on the dt-bindings header patch -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Chnages Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Chnages Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 170 ++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 343 +++++++++++++++++++++ 4 files changed, 527 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH 0/2] ASoC: SOF: ipc: Small cleanups for message handler functions
by Peter Ujfalusi 17 Sep '21

17 Sep '21

Hi, Rename the parameter for ipc_trace_message() to match it's content and use %#x" for hexadecimal prints in remaining places. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: ipc: Clarify the parameter name for ipc_trace_message() ASoC: SOF: ipc: Print 0x prefix for errors in ipc_trace/stream_message() sound/soc/sof/ipc.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) -- 2.33.0

2 3

[PATCH] ASoC: SOF: trace: Omit error print when waking up trace sleepers
by Peter Ujfalusi 17 Sep '21

17 Sep '21

Do not print error message from snd_sof_trace_notify_for_error() when possible sleeping trace work is woken up to flush the remaining debug information. This action by itself is not an error, it is just an action we take when an error occurs to make sure that all information have been fed to the userspace (if we have trace in use). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/trace.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index f72a6e83e6af..58f6ca5cf491 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -530,7 +530,6 @@ void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev) return; if (sdev->dtrace_is_enabled) { - dev_err(sdev->dev, "error: waking up any trace sleepers\n"); sdev->dtrace_error = true; wake_up(&sdev->trace_sleep); } -- 2.33.0

2 1

[PATCH] ASoC: SOF: ipc: Remove redundant error check from sof_ipc_tx_message_unlocked
by Peter Ujfalusi 17 Sep '21

17 Sep '21

If the snd_sof_dsp_send_msg() failed then we have already returned from sof_ipc_tx_message_unlocked() with the error message. There is no need to check if ret is really 0 after this and we can return directly the return value from tx_wait_done() At the same time make the remaining checks for error (ret) to match. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index a4fe007a0e4d..969b017e68d4 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -286,7 +286,7 @@ static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header, spin_unlock_irq(&sdev->ipc_lock); - if (ret < 0) { + if (ret) { dev_err_ratelimited(sdev->dev, "error: ipc tx failed with error %d\n", ret); @@ -296,10 +296,7 @@ static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header, ipc_log_header(sdev->dev, "ipc tx", msg->header); /* now wait for completion */ - if (!ret) - ret = tx_wait_done(ipc, msg, reply_data); - - return ret; + return tx_wait_done(ipc, msg, reply_data); } /* send IPC message from host to DSP */ -- 2.33.0

2 1

[PATCH] ASoC: 88pm860x: Update to modern clocking terminology
by Mark Brown 17 Sep '21

17 Sep '21

As part of moving to remove the old style defines for the bus clocks update the 88pm860x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/88pm860x-codec.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index cac7e557edc8..c6043fa58c74 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -968,16 +968,16 @@ static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_CBP_CFC: if (pm860x->dir == PM860X_CLK_DIR_OUT) { inf |= PCM_INF2_MASTER; ret = 0; } break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: if (pm860x->dir == PM860X_CLK_DIR_IN) { inf &= ~PCM_INF2_MASTER; ret = 0; @@ -1072,15 +1072,15 @@ static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: if (pm860x->dir == PM860X_CLK_DIR_OUT) inf |= PCM_INF2_MASTER; else return -EINVAL; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: if (pm860x->dir == PM860X_CLK_DIR_IN) inf &= ~PCM_INF2_MASTER; else -- 2.20.1

1 1

[PATCH] ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX
by Trevor Wu 17 Sep '21

17 Sep '21

S24_LE params fixup is only required for DPTX. Remove fixup ops assignment for HDMITX. Fixes: 40d605df0a7b ("ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index 8cd8450409e8..a3fa8efc8f81 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -424,8 +424,8 @@ static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); } -static int mt8195_hdmitx_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { /* fix BE i2s format to 32bit, clean param mask first */ @@ -902,7 +902,7 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ops = &mt8195_dptx_ops, - .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup, + .be_hw_params_fixup = mt8195_dptx_hw_params_fixup, SND_SOC_DAILINK_REG(DPTX_BE), }, [DAI_LINK_ETDM1_IN_BE] = { @@ -953,7 +953,6 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_playback = 1, - .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup, SND_SOC_DAILINK_REG(ETDM3_OUT_BE), }, [DAI_LINK_PCM1_BE] = { -- 2.18.0

2 1

[PATCH] ASoC: SOF: Rename sof_arch_ops to dsp_arch_ops
by Peter Ujfalusi 17 Sep '21

17 Sep '21

>From the name sof_arch_ops one can not decipher that these ops are DSP architecture ops. Rename it to dsp_arch_ops and change also the macro to retrieve the DSP architecture specific ops as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx8.c | 4 ++-- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 4 ++-- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/intel/icl.c | 2 +- sound/soc/sof/intel/pci-tng.c | 2 +- sound/soc/sof/intel/tgl.c | 2 +- sound/soc/sof/sof-priv.h | 14 +++++++------- sound/soc/sof/xtensa/core.c | 2 +- 11 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 12fedf0984bd..0a63ab026abb 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -441,7 +441,7 @@ struct snd_sof_dsp_ops sof_imx8_ops = { .dbg_dump = imx8_dump, /* Firmware ops */ - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, /* DAI drivers */ .drv = imx8_dai, @@ -490,7 +490,7 @@ struct snd_sof_dsp_ops sof_imx8x_ops = { .dbg_dump = imx8_dump, /* Firmware ops */ - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, /* DAI drivers */ .drv = imx8_dai, diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index cb822d953767..5d3c05f32c37 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -293,7 +293,7 @@ struct snd_sof_dsp_ops sof_imx8m_ops = { .dbg_dump = imx8_dump, /* Firmware ops */ - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, /* DAI drivers */ .drv = imx8m_dai, diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index c7ed2b3d6abc..740a24052092 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -125,7 +125,7 @@ const struct snd_sof_dsp_ops sof_apl_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; EXPORT_SYMBOL_NS(sof_apl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 89a6c1f04a55..6aaefd4e1ddc 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -657,7 +657,7 @@ static const struct snd_sof_dsp_ops sof_bdw_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; static const struct sof_intel_dsp_desc bdw_chip_info = { diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 8edaf6fdd218..1b180f00c90a 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -275,7 +275,7 @@ static const struct snd_sof_dsp_ops sof_byt_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; static const struct sof_intel_dsp_desc byt_chip_info = { @@ -353,7 +353,7 @@ static const struct snd_sof_dsp_ops sof_cht_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; static const struct sof_intel_dsp_desc cht_chip_info = { diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index e115e12a856f..00fcead5d5e4 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -330,7 +330,7 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; EXPORT_SYMBOL_NS(sof_cnl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index ee095b8f2d01..5b18d81b6791 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -125,7 +125,7 @@ const struct snd_sof_dsp_ops sof_icl_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; EXPORT_SYMBOL_NS(sof_icl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 4bded668b672..31a0f4cf06b4 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -187,7 +187,7 @@ const struct snd_sof_dsp_ops sof_tng_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; const struct sof_intel_dsp_desc tng_chip_info = { diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 199d41a7dc9b..273207e6e71b 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -120,7 +120,7 @@ const struct snd_sof_dsp_ops sof_tgl_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .arch_ops = &sof_xtensa_arch_ops, + .dsp_arch_ops = &sof_xtensa_arch_ops, }; EXPORT_SYMBOL_NS(sof_tgl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index fd8423172d8f..cc3d26c28c2b 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -268,17 +268,17 @@ struct snd_sof_dsp_ops { /* ALSA HW info flags, will be stored in snd_pcm_runtime.hw.info */ u32 hw_info; - const struct sof_arch_ops *arch_ops; + const struct dsp_arch_ops *dsp_arch_ops; }; /* DSP architecture specific callbacks for oops and stack dumps */ -struct sof_arch_ops { +struct dsp_arch_ops { void (*dsp_oops)(struct snd_sof_dev *sdev, void *oops); void (*dsp_stack)(struct snd_sof_dev *sdev, void *oops, u32 *stack, u32 stack_words); }; -#define sof_arch_ops(sdev) ((sdev)->pdata->desc->ops->arch_ops) +#define sof_dsp_arch_ops(sdev) ((sdev)->pdata->desc->ops->dsp_arch_ops) /* DSP device HW descriptor mapping between bus ID and ops */ struct sof_ops_table { @@ -551,16 +551,16 @@ extern struct snd_compress_ops sof_compressed_ops; static inline void sof_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, u32 stack_words) { - sof_arch_ops(sdev)->dsp_stack(sdev, oops, stack, stack_words); + sof_dsp_arch_ops(sdev)->dsp_stack(sdev, oops, stack, stack_words); } static inline void sof_oops(struct snd_sof_dev *sdev, void *oops) { - if (sof_arch_ops(sdev)->dsp_oops) - sof_arch_ops(sdev)->dsp_oops(sdev, oops); + if (sof_dsp_arch_ops(sdev)->dsp_oops) + sof_dsp_arch_ops(sdev)->dsp_oops(sdev, oops); } -extern const struct sof_arch_ops sof_xtensa_arch_ops; +extern const struct dsp_arch_ops sof_xtensa_arch_ops; /* * Utilities diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c index bbb9a2282ed9..85f1eb1d20d2 100644 --- a/sound/soc/sof/xtensa/core.c +++ b/sound/soc/sof/xtensa/core.c @@ -128,7 +128,7 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, } } -const struct sof_arch_ops sof_xtensa_arch_ops = { +const struct dsp_arch_ops sof_xtensa_arch_ops = { .dsp_oops = xtensa_dsp_oops, .dsp_stack = xtensa_stack, }; -- 2.33.0

2 1

[PATCH 00/12] Add support for on demand pipeline setup/destroy
by Daniel Baluta 17 Sep '21

17 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries implements the new feature to setup/teardown pipeline as needed when a PCM is open/closed. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 176 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1034 insertions(+), 538 deletions(-) -- 2.27.0

2 13

[PATCH v6 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 16 Sep '21

16 Sep '21

Many thanks for reviewing v5. This version addresses all the comments raised as part of v5 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v5: - moved usage zero length arrays to flexible arrays as suggested by Pierre. - updated dt-bindings as suggested by Rob. - Fixed various style issues and typos, spotted by Pierre - Replaced usage of spinlock with mutex, as we are mostly dealing with workqueues. - Fixed memory leak in two places spotted by Pierre. - added pga module gain setup into dapm event. - Updated Kconfig with correct ordering. - Fixed build warning for non OF cases. - Added more comments as suggested by Pierre Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 289 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1140 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 836 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1119 ++++++++++++++++ 31 files changed, 7196 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

3 37

[PATCH] ASoC: SOF: loader: release_firmware() on load failure to avoid batching
by Peter Ujfalusi 16 Sep '21

16 Sep '21

From: Marc Herbert <marc.herbert(a)intel.com> Invoke release_firmware() when the firmware fails to boot in sof_probe_continue(). The request_firmware() framework must be informed of failures in sof_probe_continue() otherwise its internal "batching" feature (different from caching) cached the firmware image forever. Attempts to correct the file in /lib/firmware/ were then silently and confusingly ignored until the next reboot. Unloading the drivers did not help because from their disconnected perspective the firmware had failed so there was nothing to release. Also leverage the new snd_sof_fw_unload() function to simplify the snd_sof_device_remove() function. Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/core.c | 4 +--- sound/soc/sof/loader.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 6be4f159ee35..20082c0f2f39 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -370,7 +370,6 @@ int snd_sof_device_remove(struct device *dev) dev_warn(dev, "error: %d failed to prepare DSP for device removal", ret); - snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_free_trace(sdev); @@ -393,8 +392,7 @@ int snd_sof_device_remove(struct device *dev) snd_sof_remove(sdev); /* release firmware */ - release_firmware(pdata->fw); - pdata->fw = NULL; + snd_sof_fw_unload(sdev); return 0; } diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..9c3f251a0dd0 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware); void snd_sof_fw_unload(struct snd_sof_dev *sdev) { /* TODO: support module unloading at runtime */ + release_firmware(sdev->pdata->fw); + sdev->pdata->fw = NULL; } EXPORT_SYMBOL(snd_sof_fw_unload); -- 2.33.0

2 1

[PATCH] ASoC: SOF: core: Move probe work related code under a single if () branch
by Peter Ujfalusi 16 Sep '21

16 Sep '21

Relocate the INIT_WORK() at the same place where we schedule the work to make the code simpler and easier to follow. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 6be4f159ee35..9cbf7d72ae92 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -325,9 +325,6 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) spin_lock_init(&sdev->hw_lock); mutex_init(&sdev->power_state_access); - if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) - INIT_WORK(&sdev->probe_work, sof_probe_work); - /* set default timeouts if none provided */ if (plat_data->desc->ipc_timeout == 0) sdev->ipc_timeout = TIMEOUT_DEFAULT_IPC_MS; @@ -339,6 +336,7 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) sdev->boot_timeout = plat_data->desc->boot_timeout; if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) { + INIT_WORK(&sdev->probe_work, sof_probe_work); schedule_work(&sdev->probe_work); return 0; } -- 2.33.0

2 1

[PATCH v2] ASoC: atmel: Convert to new style DAI format definitions
by Mark Brown 16 Sep '21

16 Sep '21

Convert the Atmel drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 4 ++-- sound/soc/atmel/atmel_ssc_dai.c | 18 +++++++++--------- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++++---- sound/soc/atmel/mikroe-proto.c | 4 ++-- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- 8 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6b3d9c05eaf2..e119abbca099 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -343,7 +343,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* codec is slave, so cpu is master */ mr |= ATMEL_I2SC_MR_MODE_MASTER; ret = atmel_i2s_get_gck_param(dev, params_rate(params)); @@ -351,7 +351,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return ret; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* codec is master, so cpu is slave */ mr |= ATMEL_I2SC_MR_MODE_SLAVE; dev->gck_param = NULL; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6a63e8797a0b..b34e995c59e4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -210,7 +210,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, return frame_size; switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) && ssc->clk_from_rk_pin) /* Receiver Frame Synchro (i.e. capture) @@ -220,7 +220,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, mck_div = 3; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) && !ssc->clk_from_rk_pin) /* Transmit Frame Synchro (i.e. playback) @@ -233,7 +233,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: r.num = ssc_p->mck_rate / mck_div / frame_size; ret = snd_interval_ratnum(i, 1, &r, &num, &den); @@ -243,8 +243,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBP_CFP: t.min = 8000; t.max = ssc_p->mck_rate / mck_div / frame_size; t.openmin = t.openmax = 0; @@ -430,8 +430,8 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, static int atmel_ssc_cfs(struct atmel_ssc_info *ssc_p) { switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBC_CFC: return 1; } return 0; @@ -441,8 +441,8 @@ static int atmel_ssc_cfs(struct atmel_ssc_info *ssc_p) static int atmel_ssc_cbs(struct atmel_ssc_info *ssc_p) { switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBC_CFC: return 1; } return 0; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 9e237580afa9..9c974c4e187d 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -66,7 +66,7 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { .stream_name = "WM8904 PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &atmel_asoc_wm8904_ops, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 8988f024a732..67550e3d9ee3 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -350,7 +350,7 @@ static int mchp_i2s_mcc_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; /* We can't generate only FSYNC */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFS) + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBP_CFC) return -EINVAL; /* We can only reconfigure the IP when it's stopped */ @@ -547,19 +547,19 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* cpu is BCLK and LRC master */ mra |= MCHP_I2SMCC_MRA_MODE_MASTER; if (dev->sysclk) mra |= MCHP_I2SMCC_MRA_IMCKMODE_GEN; set_divs = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: /* cpu is BCLK master */ mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; set_divs = 1; fallthrough; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* cpu is slave */ mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; if (dev->sysclk) diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index 0be7b4221c14..f9331f7e80fe 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -129,9 +129,9 @@ static int snd_proto_probe(struct platform_device *pdev) } if (bitclkmaster) { if (codec_np == bitclkmaster) - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; else - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; } else { dai_fmt |= snd_soc_daifmt_parse_clock_provider_as_flag(np, NULL); } diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index ed1f69b57024..915da92e1ec8 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -126,7 +126,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .stream_name = "WM8731 PCM", .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 7745250fd743..7c45dc4f8c1b 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -115,7 +115,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) dai->codecs->dai_name = "wm8731-hifi"; dai->init = sam9x5_wm8731_init; dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM; + | SND_SOC_DAIFMT_CBP_CFP; ret = snd_soc_of_parse_card_name(card, "atmel,model"); if (ret) { diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 50c3dc6936f9..1b3a31296c9b 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link tse850_dailink = { .stream_name = "TSE-850-PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFS, + | SND_SOC_DAIFMT_CBP_CFC, SND_SOC_DAILINK_REG(pcm), }; -- 2.20.1

3 3

[PATCH 0/6] ASoC: SOF: Clean up the probe support
by Peter Ujfalusi 16 Sep '21

16 Sep '21

Hi, The probe debug feature of SOF can be used to extract streams of data from a given point of a pipeline for analysis. The support is implemented by using the ALSA/ASoC compress support for the capture stream, but the code can not be used by/for a normal compressed data stream. It is a debug feature. Merge the probe implementation in the core (compress.c/h and probe.c/h) into one file: sof-probes.c/h Rename the Intel HDA specific probe implementation from hda-compressc.c to hda-probes.c We also need to add IPC logging support for the probes messages and drop the unused references to SOF compress to have reasonably clean code. Regards, Peter --- Peter Ujfalusi (5): ASoC: SOF: ipc: Add probe message logging to ipc_log_header() ASoC: SOF: pcm: Remove non existent CONFIG_SND_SOC_SOF_COMPRESS reference ASoC: SOF: probe: Merge and clean up the probe and compress files ASoC: SOF: Intel: Rename hda-compress.c to hda-probes.c ASoC: SOF: sof-probes: Correct the function names used for snd_soc_cdai_ops Ranjani Sridharan (1): ASoC: SOF: compress: move and export sof_probe_compr_ops sound/soc/sof/Makefile | 3 +- sound/soc/sof/compress.c | 147 --------- sound/soc/sof/compress.h | 32 -- sound/soc/sof/core.c | 2 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/hda-dai.c | 16 +- .../intel/{hda-compress.c => hda-probes.c} | 0 sound/soc/sof/ipc.c | 23 ++ sound/soc/sof/pcm.c | 6 +- sound/soc/sof/probe.h | 85 ------ sound/soc/sof/sof-priv.h | 5 - sound/soc/sof/{probe.c => sof-probes.c} | 280 +++++++++++------- sound/soc/sof/sof-probes.h | 38 +++ 14 files changed, 248 insertions(+), 393 deletions(-) delete mode 100644 sound/soc/sof/compress.c delete mode 100644 sound/soc/sof/compress.h rename sound/soc/sof/intel/{hda-compress.c => hda-probes.c} (100%) delete mode 100644 sound/soc/sof/probe.h rename sound/soc/sof/{probe.c => sof-probes.c} (52%) create mode 100644 sound/soc/sof/sof-probes.h -- 2.33.0

2 7

[PATCH] ASoC: au1x: Convert to modern terminology for DAI clocking
by Mark Brown 16 Sep '21

16 Sep '21

As part of retiring the old macros defining the DAI clocking mode in the DAI format update the au1x drivers to use the new style macros. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/au1x/db1200.c | 6 +++--- sound/soc/au1x/i2sc.c | 4 ++-- sound/soc/au1x/psc-i2s.c | 8 ++++---- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 5f8baad37a40..400eaf9f8b14 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -117,7 +117,7 @@ static struct snd_soc_dai_link db1200_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &db1200_i2s_wm8731_ops, SND_SOC_DAILINK_REG(db1200_i2s), }; @@ -138,7 +138,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &db1200_i2s_wm8731_ops, SND_SOC_DAILINK_REG(db1300_i2s), }; @@ -159,7 +159,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &db1200_i2s_wm8731_ops, SND_SOC_DAILINK_REG(db1550_i2s), }; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 65bd39f5032d..529284c1f4ad 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -119,9 +119,9 @@ static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) goto out; } - /* I2S controller only supports master */ + /* I2S controller only supports provider */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + case SND_SOC_DAIFMT_CBC_CFC: /* CODEC consumer */ break; default: goto out; diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 767ce950d0da..40bbf83c492f 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -91,11 +91,11 @@ static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */ - ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */ + case SND_SOC_DAIFMT_CBP_CFP: /* CODEC provider */ + ct |= PSC_I2SCFG_MS; /* PSC I2S consumer mode */ break; - case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ - ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */ + case SND_SOC_DAIFMT_CBC_CFC: /* CODEC consumer */ + ct &= ~PSC_I2SCFG_MS; /* PSC I2S provider mode */ break; default: goto out; -- 2.20.1

1 1

[PATCH] ASoC: SOF: imx8m: add SAI1 info
by Daniel Baluta 16 Sep '21

16 Sep '21

From: Viorel Suman <viorel.suman(a)nxp.com> Add SAI1 instance to imx8m_dai array. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/imx/imx8m.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index b9f8a9e8cc3a..6dad87ed0bc8 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -253,6 +253,17 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev, } static struct snd_soc_dai_driver imx8m_dai[] = { +{ + .name = "sai1", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, +}, { .name = "sai3", .playback = { -- 2.27.0

2 1

[PATCH] misc: cs35l41: Remove unused pdn variable
by Charles Keepax 16 Sep '21

16 Sep '21

Remove pdn variable that was made redundant in an earlier patch. Fixes: c2f14cc2bcdd ("ASoC: cs35l41: Fix use of an uninitialised variable") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index ad86c030d9cb5..b16eb6610c0e9 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -564,7 +564,6 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, struct cs35l41_private *cs35l41 = snd_soc_component_get_drvdata(component); unsigned int val; int ret = 0; - bool pdn; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -582,7 +581,6 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs35l41->regmap, CS35L41_PWR_CTRL1, CS35L41_GLOBAL_EN_MASK, 0); - pdn = false; ret = regmap_read_poll_timeout(cs35l41->regmap, CS35L41_IRQ1_STATUS1, val, val & CS35L41_PDN_DONE_MASK, 1000, 100000); -- 2.11.0

2 1

[PATCH] ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication
by Peter Ujfalusi 16 Sep '21

16 Sep '21

In case the firmware is missing we will have the following in the kernel log: 1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2 2 | error: request firmware intel/sof/sof-tgl-h.ri failed err: -2 3 | you may need to download the firmware from https://github.com/thesofproject/sof-bin/ 4 | error: failed to load DSP firmware -2 5 | error: sof_probe_work failed err: -2 The first line is the standard, request_firmware() warning. The second and third line is printed in snd_sof_load_firmware_raw() Note that the first and second line is mostly identical. With this patch the log will be changed to: 1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2 2 | error: sof firmware file is missing, you might need to 3 | download it from https://github.com/thesofproject/sof-bin/ 4 | error: failed to load DSP firmware -2 5 | error: sof_probe_work failed err: -2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/loader.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..46f3900487ee 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev); if (ret < 0) { - dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", - fw_filename, ret); dev_err(sdev->dev, - "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n"); + "error: sof firmware file is missing, you might need to\n"); + dev_err(sdev->dev, + " download it from https://github.com/thesofproject/sof-bin/\n"); goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", -- 2.33.0

2 1

[PATCH v1 0/1] ASoC: cs42l42: Implement Manual Type detection as fallback
by Vitaly Rodionov 16 Sep '21

16 Sep '21

For some headsets CS42L42 autodetect mode is not working correctly. They will be detected as unknown types or as headphones. According to the CS42L42 datasheet, if the headset autodetect failed, then the driver should switch to manual mode and perform a manual steps sequence. These steps were missing in the current driver code. This patch will add manual mode fallback steps in case autodetect failed. The default behavior is not affected, manual mode runs only when autodetect failed. Tested for regression with autodetect with all known headsets - no regression. Tested with all headsets customers reported as false detected: Gumdrop DropTech B1 - detected as headset OK HUAWEI AM115 - detected as headset OK UGREEN EP103 - detected as headset OK HONOR AM116 - detected as headset OK Stefan Binding (1): ASoC: cs42l42: Implement Manual Type detection as fallback sound/soc/codecs/cs42l42.c | 104 ++++++++++++++++++++++++++++++++----- sound/soc/codecs/cs42l42.h | 54 +++++++++++++++++++ 2 files changed, 146 insertions(+), 12 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: cs42l42: Minor fix all errors reported by checkpatch.pl script
by Vitaly Rodionov 16 Sep '21

16 Sep '21

Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index c586ebff45f7..d5e1e5228b5f 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -851,7 +851,7 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 24) cs42l42->bclk = (cs42l42->bclk / 3) * 4; - switch(substream->stream) { + switch (substream->stream) { case SNDRV_PCM_STREAM_CAPTURE: if (channels == 2) { val |= CS42L42_ASP_TX_CH2_AP_MASK; @@ -935,7 +935,7 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) CS42L42_HP_ANA_BMUTE_MASK); cs42l42->stream_use &= ~(1 << stream); - if(!cs42l42->stream_use) { + if (!cs42l42->stream_use) { /* * Switch to the internal oscillator. * SCLK must remain running until after this clock switch. @@ -1015,7 +1015,7 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE ) + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops cs42l42_ops = { .startup = cs42l42_dai_startup, @@ -1572,7 +1572,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) if ((~masks[5]) & irq_params_table[5].mask) { if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) { cs42l42_process_hs_type_detect(cs42l42); - switch(cs42l42->hs_type){ + switch (cs42l42->hs_type) { case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADSET, @@ -1604,7 +1604,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) cs42l42->plug_state = CS42L42_TS_UNPLUG; cs42l42_cancel_hs_type_detect(cs42l42); - switch(cs42l42->hs_type){ + switch (cs42l42->hs_type) { case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: snd_soc_jack_report(cs42l42->jack, 0, SND_JACK_HEADSET); -- 2.25.1

2 1

[PATCH] ASoC: cs35l41: Binding fixes
by David Rhodes 16 Sep '21

16 Sep '21

Fix warnings and errors in DT bindings Add newline at end of file Replace 'unevaluatedProperties' with 'additionalProperties' Add spi context to DT example Add #sound-dai-cells to DT example Rename to 'cirrus,cs35l41.yaml' Signed-off-by: David Rhodes <drhodes(a)opensource.cirrus.com> --- .../{cs35l41.yaml => cirrus,cs35l41.yaml} | 30 +++++++++++-------- 1 file changed, 18 insertions(+), 12 deletions(-) rename Documentation/devicetree/bindings/sound/{cs35l41.yaml => cirrus,cs35l41.yaml} (87%) diff --git a/Documentation/devicetree/bindings/sound/cs35l41.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml similarity index 87% rename from Documentation/devicetree/bindings/sound/cs35l41.yaml rename to Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml index fde78c850286..3235702ce402 100644 --- a/Documentation/devicetree/bindings/sound/cs35l41.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause %YAML 1.2 --- -$id: http://devicetree.org/schemas/sound/cs35l41.yaml# +$id: http://devicetree.org/schemas/sound/cirrus,cs35l41.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# title: Cirrus Logic CS35L41 Speaker Amplifier @@ -135,17 +135,23 @@ required: - cirrus,boost-ind-nanohenry - cirrus,boost-cap-microfarad -unevaluatedProperties: false +additionalProperties: false examples: - | - cs35l41: cs35l41@2 { - compatible = "cirrus,cs35l41"; - reg = <2>; - VA-supply = <&dummy_vreg>; - VP-supply = <&dummy_vreg>; - reset-gpios = <&gpio 110 0>; - cirrus,boost-peak-milliamp = <4500>; - cirrus,boost-ind-nanohenry = <1000>; - cirrus,boost-cap-microfarad = <15>; - }; \ No newline at end of file + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l41: cs35l41@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; + }; + }; -- 2.25.1

2 1

[PATCH] ALSA: hda/cs8409: Setup Dolphin Headset Mic as Phantom Jack
by Vitaly Rodionov 16 Sep '21

16 Sep '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Dell's requirement to have headset mic as phantom jack on this specific dolphin hardware platform. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 3c7ef55d016e..31ff11ab868e 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true, SND_JACK_HEADPHONE, NULL); + snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true, + SND_JACK_MICROPHONE, NULL); + cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID); -- 2.25.1

2 1

[PATCH] pcxhr: "fix" PCXHR_REG_TO_PORT definition
by Alexey Dobriyan 16 Sep '21

16 Sep '21

The following preprocessor directive is non-compliant: #undef PCXHR_REG_TO_PORT(x) gcc warns about extra tokens but nobody sees them as they are under if branch which is never parsed. Make it an #error, it is not clear to me what the author meant. Signed-off-by: Alexey Dobriyan <adobriyan(a)gmail.com> --- sound/pci/pcxhr/pcxhr_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -52,7 +52,7 @@ #define PCXHR_DSP 2 #if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN) -#undef PCXHR_REG_TO_PORT(x) +#error PCXHR_REG_TO_PORT(x) #else #define PCXHR_REG_TO_PORT(x) ((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP) #endif

2 1

[PATCH] ASoC: nau8821: Add driver for Nuvoton codec NAU88L21
by Seven Lee 16 Sep '21

16 Sep '21

The driver is for codec NAU88L21 of Nuvoton Technology Corporation. The NAU88L21 is an ultra-low power high performance audio codec that supportsboth analog and digital audio functions. In addition, DMIC function is added to calculate the clock threshold value by device properties. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8821.c | 1804 +++++++++++++++++ sound/soc/codecs/nau8821.h | 533 +++++ 5 files changed, 2399 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 sound/soc/codecs/nau8821.c create mode 100644 sound/soc/codecs/nau8821.h diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt new file mode 100644 index 000000000000..6c3baf7a5f21 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -0,0 +1,55 @@ +Nuvoton NAU88L21 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8821" + + - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. + + +Example: + + headset: nau8821@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,dmic-clk-threshold = 3072000; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..11bcf17b5f91 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -136,6 +136,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 + imply SND_SOC_NAU8821 imply SND_SOC_NAU8822 imply SND_SOC_NAU8824 imply SND_SOC_NAU8825 @@ -1904,6 +1905,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8821 + tristate "Nuvoton Technology Corporation NAU88L21 CODEC" + depends on I2C + config SND_SOC_NAU8822 tristate "Nuvoton Technology Corporation NAU88C22 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..55daf5fbc1b7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -142,6 +142,7 @@ snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o +snd-soc-nau8821-objs := nau8821.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -468,6 +469,7 @@ obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8821) += snd-soc-nau8821.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c new file mode 100644 index 000000000000..1090da91bafd --- /dev/null +++ b/sound/soc/codecs/nau8821.c @@ -0,0 +1,1804 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * nau8821.c -- Nuvoton NAU88L21 audio codec driver + * + * Copyright 2021 Nuvoton Technology Corp. + * Author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Lee <wtli(a)nuvoton.com> + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/acpi.h> +#include <linux/math64.h> +#include <linux/semaphore.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "nau8821.h" + +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MIN 90000000 + +/* the maximum frequency of CLK_ADC and CLK_DAC */ +#define CLK_DA_AD_MAX 6144000 + +static int nau8821_configure_sysclk(struct nau8821 *nau8821, + int clk_id, unsigned int freq); + +struct nau8821_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau8821_fll_attr { + unsigned int param; + unsigned int val; +}; + +/* scaling for mclk from sysclk_src output */ +static const struct nau8821_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, + { 48, 0xd }, + { 96, 0xe }, + { 5, 0xf }, +}; + +/* ratio for input clk freq */ +static const struct nau8821_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau8821_fll_attr fll_pre_scalar[] = { + { 0, 0x0 }, + { 1, 0x1 }, + { 2, 0x2 }, + { 3, 0x3 }, +}; + +/* over sampling rate */ +struct nau8821_osr_attr { + unsigned int osr; + unsigned int clk_src; +}; + +static const struct nau8821_osr_attr osr_dac_sel[] = { + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 256, 0 }, /* OSR 256, SRC 1 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 0, 0 }, + { 32, 3 }, /* OSR 32, SRC 1/8 */ +}; + +static const struct nau8821_osr_attr osr_adc_sel[] = { + { 32, 3 }, /* OSR 32, SRC 1/8 */ + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 256, 0 }, /* OSR 256, SRC 1 */ +}; + +struct nau8821_dmic_speed { + unsigned int param; + unsigned int val; +}; + +static const struct nau8821_dmic_speed dmic_speed_sel[] = { + { 0, 0x0 }, /*SPEED 1, SRC 1 */ + { 1, 0x1 }, /*SPEED 2, SRC 1/2 */ + { 2, 0x2 }, /*SPEED 4, SRC 1/4 */ + { 3, 0x3 }, /*SPEED 8, SRC 1/8 */ +}; + +static const struct reg_default nau8821_reg_defaults[] = { + { NAU8821_R01_ENA_CTRL, 0x00ff }, + { NAU8821_R03_CLK_DIVIDER, 0x0050 }, + { NAU8821_R04_FLL1, 0x0 }, + { NAU8821_R05_FLL2, 0x00bc }, + { NAU8821_R06_FLL3, 0x0008 }, + { NAU8821_R07_FLL4, 0x0010 }, + { NAU8821_R08_FLL5, 0x4000 }, + { NAU8821_R09_FLL6, 0x6900 }, + { NAU8821_R0A_FLL7, 0x0031 }, + { NAU8821_R0B_FLL8, 0x26e9 }, + { NAU8821_R0D_JACK_DET_CTRL, 0x0 }, + { NAU8821_R0F_INTERRUPT_MASK, 0x0 }, + { NAU8821_R12_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8821_R13_DMIC_CTRL, 0x0 }, + { NAU8821_R1A_GPIO12_CTRL, 0x0 }, + { NAU8821_R1B_TDM_CTRL, 0x0 }, + { NAU8821_R1C_I2S_PCM_CTRL1, 0x000a }, + { NAU8821_R1D_I2S_PCM_CTRL2, 0x8010 }, + { NAU8821_R1E_LEFT_TIME_SLOT, 0x0 }, + { NAU8821_R1F_RIGHT_TIME_SLOT, 0x0 }, + { NAU8821_R21_BIQ0_COF1, 0x0 }, + { NAU8821_R22_BIQ0_COF2, 0x0 }, + { NAU8821_R23_BIQ0_COF3, 0x0 }, + { NAU8821_R24_BIQ0_COF4, 0x0 }, + { NAU8821_R25_BIQ0_COF5, 0x0 }, + { NAU8821_R26_BIQ0_COF6, 0x0 }, + { NAU8821_R27_BIQ0_COF7, 0x0 }, + { NAU8821_R28_BIQ0_COF8, 0x0 }, + { NAU8821_R29_BIQ0_COF9, 0x0 }, + { NAU8821_R2A_BIQ0_COF10, 0x0 }, + { NAU8821_R2B_ADC_RATE, 0x0002 }, + { NAU8821_R2C_DAC_CTRL1, 0x0082 }, + { NAU8821_R2D_DAC_CTRL2, 0x0 }, + { NAU8821_R2F_DAC_DGAIN_CTRL, 0x0 }, + { NAU8821_R30_ADC_DGAIN_CTRL, 0x0 }, + { NAU8821_R31_MUTE_CTRL, 0x0 }, + { NAU8821_R32_HSVOL_CTRL, 0x0 }, + { NAU8821_R34_DACR_CTRL, 0xcfcf }, + { NAU8821_R35_ADC_DGAIN_CTRL1, 0xcfcf }, + { NAU8821_R36_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8821_R37_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8821_R38_ADC_DRC_SLOPES, 0x25ff }, + { NAU8821_R39_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8821_R3A_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8821_R3B_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8821_R3C_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8821_R3D_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8821_R41_BIQ1_COF1, 0x0 }, + { NAU8821_R42_BIQ1_COF2, 0x0 }, + { NAU8821_R43_BIQ1_COF3, 0x0 }, + { NAU8821_R44_BIQ1_COF4, 0x0 }, + { NAU8821_R45_BIQ1_COF5, 0x0 }, + { NAU8821_R46_BIQ1_COF6, 0x0 }, + { NAU8821_R47_BIQ1_COF7, 0x0 }, + { NAU8821_R48_BIQ1_COF8, 0x0 }, + { NAU8821_R49_BIQ1_COF9, 0x0 }, + { NAU8821_R4A_BIQ1_COF10, 0x0 }, + { NAU8821_R4B_CLASSG_CTRL, 0x0 }, + { NAU8821_R4C_IMM_MODE_CTRL, 0x0 }, + { NAU8821_R4D_IMM_RMS_L, 0x0 }, + { NAU8821_R53_OTPDOUT_1, 0xaad8 }, + { NAU8821_R54_OTPDOUT_2, 0x0002 }, + { NAU8821_R55_MISC_CTRL, 0x0 }, + { NAU8821_R66_BIAS_ADJ, 0x0 }, + { NAU8821_R68_TRIM_SETTINGS, 0x0 }, + { NAU8821_R69_ANALOG_CONTROL_1, 0x0 }, + { NAU8821_R6A_ANALOG_CONTROL_2, 0x0 }, + { NAU8821_R6B_PGA_MUTE, 0x0 }, + { NAU8821_R71_ANALOG_ADC_1, 0x0011 }, + { NAU8821_R72_ANALOG_ADC_2, 0x0020 }, + { NAU8821_R73_RDAC, 0x0008 }, + { NAU8821_R74_MIC_BIAS, 0x0006 }, + { NAU8821_R76_BOOST, 0x0 }, + { NAU8821_R77_FEPGA, 0x0 }, + { NAU8821_R7E_PGA_GAIN, 0x0 }, + { NAU8821_R7F_POWER_UP_CONTROL, 0x0 }, + { NAU8821_R80_CHARGE_PUMP, 0x0 }, +}; + +/** + * nau8821_sema_acquire - acquire the semaphore of nau8821 + * @nau8821: component to register the codec private data with + * @timeout: how long in jiffies to wait before failure or zero to wait + * until release + * + * Attempts to acquire the semaphore with number of jiffies. If no more + * tasks are allowed to acquire the semaphore, calling this function will + * put the task to sleep. If the semaphore is not released within the + * specified number of jiffies, this function returns. + * If the semaphore is not released within the specified number of jiffies, + * this function returns -ETIME. If the sleep is interrupted by a signal, + * this function will return -EINTR. It returns 0 if the semaphore was + * acquired successfully. + * + * Acquires the semaphore without jiffies. Try to acquire the semaphore + * atomically. Returns 0 if the semaphore has been acquired successfully + * or 1 if it cannot be acquired. + */ +static int nau8821_sema_acquire(struct nau8821 *nau8821, long timeout) +{ + int ret; + + if (!nau8821->irq) + return 1; + + if (timeout) { + ret = down_timeout(&nau8821->jd_sem, timeout); + if (ret < 0) + dev_dbg(nau8821->dev, "Acquire semaphore timeout\n"); + } else { + ret = down_trylock(&nau8821->jd_sem); + if (ret) + dev_dbg(nau8821->dev, "Acquire semaphore fail\n"); + } + + return ret; +} + +/** + * nau8821_sema_release - release the semaphore of nau8821 + * @nau8821: component to register the codec private data with + * + * Release the semaphore which may be called from any context and + * even by tasks which have never called down(). + */ +static inline void nau8821_sema_release(struct nau8821 *nau8821) +{ + if (!nau8821->irq) + return; + up(&nau8821->jd_sem); +} + +/** + * nau8821_sema_reset - reset the semaphore for nau8821 + * @nau8821: component to register the codec private data with + * + * Reset the counter of the semaphore. Call this function to restart + * a new round task management. + */ +static inline void nau8821_sema_reset(struct nau8821 *nau8821) +{ + nau8821->jd_sem.count = 1; +} + +static bool nau8821_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET ... NAU8821_R01_ENA_CTRL: + case NAU8821_R03_CLK_DIVIDER ... NAU8821_R0B_FLL8: + case NAU8821_R0D_JACK_DET_CTRL: + case NAU8821_R0F_INTERRUPT_MASK ... NAU8821_R13_DMIC_CTRL: + case NAU8821_R1A_GPIO12_CTRL ... NAU8821_R1F_RIGHT_TIME_SLOT: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2D_DAC_CTRL2: + case NAU8821_R2F_DAC_DGAIN_CTRL ... NAU8821_R32_HSVOL_CTRL: + case NAU8821_R34_DACR_CTRL ... NAU8821_R3D_DAC_DRC_ATKDCY: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4F_FUSE_CTRL3: + case NAU8821_R51_FUSE_CTRL1: + case NAU8821_R53_OTPDOUT_1 ... NAU8821_R55_MISC_CTRL: + case NAU8821_R58_I2C_DEVICE_ID ... NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R66_BIAS_ADJ: + case NAU8821_R68_TRIM_SETTINGS ... NAU8821_R6B_PGA_MUTE: + case NAU8821_R71_ANALOG_ADC_1 ... NAU8821_R74_MIC_BIAS: + case NAU8821_R76_BOOST ... NAU8821_R77_FEPGA: + case NAU8821_R7E_PGA_GAIN ... NAU8821_R82_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static bool nau8821_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET ... NAU8821_R01_ENA_CTRL: + case NAU8821_R03_CLK_DIVIDER ... NAU8821_R0B_FLL8: + case NAU8821_R0D_JACK_DET_CTRL: + case NAU8821_R0F_INTERRUPT_MASK: + case NAU8821_R11_INT_CLR_KEY_STATUS ... NAU8821_R13_DMIC_CTRL: + case NAU8821_R1A_GPIO12_CTRL ... NAU8821_R1F_RIGHT_TIME_SLOT: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2D_DAC_CTRL2: + case NAU8821_R2F_DAC_DGAIN_CTRL ... NAU8821_R32_HSVOL_CTRL: + case NAU8821_R34_DACR_CTRL ... NAU8821_R3D_DAC_DRC_ATKDCY: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4C_IMM_MODE_CTRL: + case NAU8821_R4E_FUSE_CTRL2 ... NAU8821_R4F_FUSE_CTRL3: + case NAU8821_R51_FUSE_CTRL1: + case NAU8821_R55_MISC_CTRL: + case NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R66_BIAS_ADJ: + case NAU8821_R68_TRIM_SETTINGS ... NAU8821_R6B_PGA_MUTE: + case NAU8821_R71_ANALOG_ADC_1 ... NAU8821_R74_MIC_BIAS: + case NAU8821_R76_BOOST ... NAU8821_R77_FEPGA: + case NAU8821_R7E_PGA_GAIN ... NAU8821_R80_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8821_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET: + case NAU8821_R10_IRQ_STATUS ... NAU8821_R11_INT_CLR_KEY_STATUS: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2A_BIQ0_COF10: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4A_BIQ1_COF10: + case NAU8821_R4D_IMM_RMS_L: + case NAU8821_R53_OTPDOUT_1 ... NAU8821_R54_OTPDOUT_2: + case NAU8821_R58_I2C_DEVICE_ID ... NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R81_CHARGE_PUMP_INPUT_READ ... NAU8821_R82_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8821_biq_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + if (!component->regmap) + return -EINVAL; + + regmap_raw_read(component->regmap, NAU8821_R21_BIQ0_COF1, + ucontrol->value.bytes.data, params->max); + + return 0; +} + +static int nau8821_biq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + + if (!component->regmap) + return -EINVAL; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + regmap_raw_write(component->regmap, NAU8821_R21_BIQ0_COF1, + data, params->max); + + kfree(data); + + return 0; +} + +static const char * const nau8821_adc_decimation[] = { + "32", "64", "128", "256" }; + +static const struct soc_enum nau8821_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8821_R2B_ADC_RATE, NAU8821_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8821_adc_decimation), nau8821_adc_decimation); + +static const char * const nau8821_dac_oversampl[] = { + "64", "256", "128", "", "32" }; + +static const struct soc_enum nau8821_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8821_dac_oversampl), nau8821_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -6600, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -900, 0); +static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -6600, 50, 1); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); +static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -7000, 2400); + +static const struct snd_kcontrol_new nau8821_controls[] = { + SOC_DOUBLE_TLV("Mic Volume", NAU8821_R35_ADC_DGAIN_CTRL1, + NAU8821_ADCL_CH_VOL_SFT, NAU8821_ADCR_CH_VOL_SFT, + 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8821_R30_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("Headphone Volume", NAU8821_R32_HSVOL_CTRL, + NAU8821_HPL_VOL_SFT, NAU8821_HPR_VOL_SFT, 0x3, 1, hp_vol_tlv), + SOC_DOUBLE_TLV("Digital Playback Volume", NAU8821_R34_DACR_CTRL, + NAU8821_DACL_CH_VOL_SFT, NAU8821_DACR_CH_VOL_SFT, + 0xcf, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Frontend PGA Volume", NAU8821_R7E_PGA_GAIN, + NAU8821_PGA_GAIN_L_SFT, NAU8821_PGA_GAIN_R_SFT, + 37, 0, fepga_gain_tlv), + SOC_DOUBLE_TLV("Headphone Crosstalk Volume", + NAU8821_R2F_DAC_DGAIN_CTRL, + 0, 8, 0xff, 0, crosstalk_vol_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8821_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8821_dac_oversampl_enum), + SND_SOC_BYTES_EXT("BIQ Coefficients", 20, + nau8821_biq_coeff_get, nau8821_biq_coeff_put), +}; + +static const struct snd_kcontrol_new nau8821_adc_ch0_dmic = + SOC_DAPM_SINGLE("Switch", NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_EN_SFT, 1, 0); + +static int dmic_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + int i, speed_selection, clk_adc_src, clk_adc; + unsigned int clk_divider_r03; + + /* The DMIC clock is gotten from adc clock divided by + * CLK_DMIC_SRC (1, 2, 4, 8). The clock has to be equal or + * less than nau8821->dmic_clk_threshold. + */ + regmap_read(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + &clk_divider_r03); + clk_adc_src = (clk_divider_r03 & NAU8821_CLK_ADC_SRC_MASK) + >> NAU8821_CLK_ADC_SRC_SFT; + clk_adc = (nau8821->fs * 256) >> clk_adc_src; + + for (i = 0 ; i < 4 ; i++) { + if ((clk_adc >> dmic_speed_sel[i].param) <= + nau8821->dmic_clk_threshold) { + speed_selection = dmic_speed_sel[i].val; + break; + } + } + + dev_dbg(nau8821->dev, + "clk_adc=%d, dmic_clk_threshold = %d, param=%d, val = %d\n", + clk_adc, nau8821->dmic_clk_threshold, + dmic_speed_sel[i].param, dmic_speed_sel[i].val); + regmap_update_bits(nau8821->regmap, NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_SRC_MASK, + (speed_selection << NAU8821_DMIC_SRC_SFT)); + + return 0; +} + +static int nau8821_left_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(125); + regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCL, NAU8821_EN_ADCL); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, + NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCL, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_right_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(125); + regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCR, NAU8821_EN_ADCR); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, + NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(20); + regmap_update_bits(nau8821->regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_JAMNODCLOW, NAU8821_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8821->regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_output_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8821->regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_TESTDAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_TESTDAC_EN, NAU8821_BIAS_TESTDAC_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget nau8821_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_POWERUP_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA L", NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_PGA_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA R", NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_PGA_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCL Power", NAU8821_R72_ANALOG_ADC_2, + NAU8821_POWERUP_ADCL_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCR Power", NAU8821_R72_ANALOG_ADC_2, + NAU8821_POWERUP_ADCR_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Clock", SND_SOC_NOPM, 0, 0, + dmic_clock_control, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCL", NULL, SND_SOC_NOPM, 0, 0, + nau8821_left_adc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADCR", NULL, SND_SOC_NOPM, 0, 0, + nau8821_right_adc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("DMIC Enable", SND_SOC_NOPM, + 0, 0, &nau8821_adc_ch0_dmic), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_TRISTATE_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8821_R73_RDAC, + NAU8821_DACL_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8821_R73_RDAC, + NAU8821_DACR_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL Clock", 3, NAU8821_R73_RDAC, + NAU8821_DACL_CLK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR Clock", 3, NAU8821_R73_RDAC, + NAU8821_DACR_CLK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8821_R01_ENA_CTRL, + NAU8821_EN_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8821_R01_ENA_CTRL, + NAU8821_EN_DACL_SFT, 0), + SND_SOC_DAPM_PGA_S("HP amp L", 0, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_LDAC_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP amp R", 0, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_RDAC_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8821_R80_CHARGE_PUMP, + NAU8821_CHANRGE_PUMP_EN_SFT, 0, nau8821_pump_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA_S("Output Driver R Stage 1", 4, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_INTEG_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 1", 4, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_INTEG_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 2", 5, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_DRV_INSTG_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 2", 5, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_DRV_INSTG_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 6, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_MAIN_DRV_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 6, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_MAIN_DRV_L_SFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 7, + NAU8821_R80_CHARGE_PUMP, NAU8821_POWER_DOWN_DACL_SFT, + 0, nau8821_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 7, + NAU8821_R80_CHARGE_PUMP, NAU8821_POWER_DOWN_DACR_SFT, + 0, nau8821_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* HPOL/R are ungrounded by disabling 16 Ohm pull-downs on playback */ + SND_SOC_DAPM_PGA_S("HPOL Pulldown", 8, + NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HPOR Pulldown", 8, + NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R_SFT, 0, NULL, 0), + + /* High current HPOL/R boost driver */ + SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, + NAU8821_R76_BOOST, NAU8821_HP_BOOST_DIS_SFT, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Class G", NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8821_dapm_routes[] = { + {"Frontend PGA L", NULL, "MICL"}, + {"Frontend PGA R", NULL, "MICR"}, + + {"DMIC Enable", "Switch", "DMIC"}, + {"DMIC Enable", NULL, "DMIC Clock"}, + {"ADCL", NULL, "DMIC Enable"}, + {"ADCR", NULL, "DMIC Enable"}, + + {"Frontend PGA L", NULL, "ADCL Power"}, + {"Frontend PGA R", NULL, "ADCR Power"}, + {"Frontend PGA L", NULL, "MICBIAS"}, + {"Frontend PGA R", NULL, "MICBIAS"}, + + {"ADCL", NULL, "Frontend PGA L"}, + {"ADCR", NULL, "Frontend PGA R"}, + + {"AIFTX", NULL, "ADCL"}, + {"AIFTX", NULL, "ADCR"}, + {"AIFTX", NULL, "DMIC Enable"}, + + {"DDACL", NULL, "AIFRX"}, + {"DDACR", NULL, "AIFRX"}, + + {"HP amp L", NULL, "DDACL"}, + {"HP amp R", NULL, "DDACR"}, + {"Charge Pump", NULL, "HP amp L"}, + {"Charge Pump", NULL, "HP amp R"}, + {"ADACL", NULL, "Charge Pump"}, + {"ADACR", NULL, "Charge Pump"}, + {"ADACL Clock", NULL, "ADACL"}, + {"ADACR Clock", NULL, "ADACR"}, + {"Output Driver L Stage 1", NULL, "ADACL Clock"}, + {"Output Driver R Stage 1", NULL, "ADACR Clock"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL Pulldown", NULL, "Output DACL"}, + {"HPOR Pulldown", NULL, "Output DACR"}, + {"HP Boost Driver", NULL, "HPOL Pulldown"}, + {"HP Boost Driver", NULL, "HPOR Pulldown"}, + {"Class G", NULL, "HP Boost Driver"}, + {"HPOL", NULL, "Class G"}, + {"HPOR", NULL, "Class G"}, +}; + +static int nau8821_clock_check(struct nau8821 *nau8821, + int stream, int rate, int osr) +{ + int osrate = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (osr >= ARRAY_SIZE(osr_dac_sel)) + return -EINVAL; + osrate = osr_dac_sel[osr].osr; + } else { + if (osr >= ARRAY_SIZE(osr_adc_sel)) + return -EINVAL; + osrate = osr_adc_sel[osr].osr; + } + + if (!osrate || rate * osrate > CLK_DA_AD_MAX) { + dev_err(nau8821->dev, + "exceed the maximum frequency of CLK_ADC or CLK_DAC\n"); + return -EINVAL; + } + + return 0; +} + +static int nau8821_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, clk_div; + + nau8821_sema_acquire(nau8821, HZ); + nau8821->fs = params_rate(params); + /* CLK_DAC or CLK_ADC = OSR * FS + * DAC or ADC clock frequency is defined as Over Sampling Rate (OSR) + * multiplied by the audio sample rate (Fs). Note that the OSR and Fs + * values must be selected such that the maximum frequency is less + * than 6.144 MHz. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8821->regmap, NAU8821_R2C_DAC_CTRL1, &osr); + osr &= NAU8821_DAC_OVERSAMPLE_MASK; + if (nau8821_clock_check(nau8821, substream->stream, + nau8821->fs, osr)) { + nau8821_sema_release(nau8821); + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_DAC_SRC_MASK, + osr_dac_sel[osr].clk_src << NAU8821_CLK_DAC_SRC_SFT); + } else { + regmap_read(nau8821->regmap, NAU8821_R2B_ADC_RATE, &osr); + osr &= NAU8821_ADC_SYNC_DOWN_MASK; + if (nau8821_clock_check(nau8821, substream->stream, + nau8821->fs, osr)) { + nau8821_sema_release(nau8821); + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_ADC_SRC_MASK, + osr_adc_sel[osr].clk_src << NAU8821_CLK_ADC_SRC_SFT); + } + + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, &ctrl_val); + if (ctrl_val & NAU8821_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / nau8821->fs; + if (bclk_fs <= 32) + clk_div = 3; + else if (bclk_fs <= 64) + clk_div = 2; + else if (bclk_fs <= 128) + clk_div = 1; + else { + nau8821_sema_release(nau8821); + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_LRC_DIV_MASK | NAU8821_I2S_BLK_DIV_MASK, + (clk_div << NAU8821_I2S_LRC_DIV_SFT) | clk_div); + } + + switch (params_width(params)) { + case 16: + val_len |= NAU8821_I2S_DL_16; + break; + case 20: + val_len |= NAU8821_I2S_DL_20; + break; + case 24: + val_len |= NAU8821_I2S_DL_24; + break; + case 32: + val_len |= NAU8821_I2S_DL_32; + break; + default: + nau8821_sema_release(nau8821); + return -EINVAL; + } + + regmap_update_bits(nau8821->regmap, NAU8821_R1C_I2S_PCM_CTRL1, + NAU8821_I2S_DL_MASK, val_len); + + nau8821_sema_release(nau8821); + + return 0; +} + +static int nau8821_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8821_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8821_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8821_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8821_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8821_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8821_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8821_I2S_DF_PCM_AB; + ctrl1_val |= NAU8821_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + nau8821_sema_acquire(nau8821, HZ); + + regmap_update_bits(nau8821->regmap, NAU8821_R1C_I2S_PCM_CTRL1, + NAU8821_I2S_DL_MASK | NAU8821_I2S_DF_MASK | + NAU8821_I2S_BP_MASK | NAU8821_I2S_PCMB_MASK, ctrl1_val); + regmap_update_bits(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, ctrl2_val); + + nau8821_sema_release(nau8821); + + return 0; +} + +static int nau8821_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int val = 0; + + if (mute) + val = NAU8821_DAC_SOFT_MUTE; + + return regmap_update_bits(nau8821->regmap, + NAU8821_R31_MUTE_CTRL, NAU8821_DAC_SOFT_MUTE, val); +} + +static const struct snd_soc_dai_ops nau8821_dai_ops = { + .hw_params = nau8821_hw_params, + .set_fmt = nau8821_set_dai_fmt, + .mute_stream = nau8821_digital_mute, +}; + +#define NAU8821_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8821_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8821_dai = { + .name = NUVOTON_CODEC_DAI, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8821_RATES, + .formats = NAU8821_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8821_RATES, + .formats = NAU8821_FORMATS, + }, + .ops = &nau8821_dai_ops, +}; + + +static bool nau8821_is_jack_inserted(struct regmap *regmap) +{ + bool active_high, is_high; + int status, jkdet; + + regmap_read(regmap, NAU8821_R0D_JACK_DET_CTRL, &jkdet); + active_high = jkdet & NAU8821_JACK_POLARITY; + regmap_read(regmap, NAU8821_R82_GENERAL_STATUS, &status); + is_high = status & NAU8821_GPIO2_IN; + /* return jack connection status according to jack insertion logic + * active high or active low. + */ + return active_high == is_high; +} + +static void nau8821_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8821_R10_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8821_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8821_R11_INT_CLR_KEY_STATUS, clear_irq); + } +} + +static void nau8821_eject_jack(struct nau8821 *nau8821) +{ + struct snd_soc_dapm_context *dapm = nau8821->dapm; + struct regmap *regmap = nau8821->regmap; + + /* Reset semaphore */ + nau8821_sema_reset(nau8821); + + /* Detach 2kOhm Resistors from MICBIAS to MICGND */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, 0); + /* HPL/HPR short to ground */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R | NAU8821_SPKR_DWN1L, 0); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8821_int_status_clear_all(regmap); + + /* Enable the insertion interruption, disable the ejection inter- + * ruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_EJECT_DIS | NAU8821_IRQ_INSERT_DIS, + NAU8821_IRQ_EJECT_DIS); + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_EJECT_EN); + + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, NAU8821_JACK_DET_DB_BYPASS); + + /* Close clock for jack type detection at manual mode */ + if (dapm->bias_level < SND_SOC_BIAS_PREPARE) + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + + /* Recover to normal channel input */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_R_SRC_EN, 0); +} + +static void nau8821_jdet_work(struct work_struct *work) +{ + struct nau8821 *nau8821 = + container_of(work, struct nau8821, jdet_work); + struct snd_soc_dapm_context *dapm = nau8821->dapm; + struct regmap *regmap = nau8821->regmap; + int jack_status_reg, mic_detected, event = 0, event_mask = 0; + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + msleep(20); + + regmap_read(regmap, NAU8821_R58_I2C_DEVICE_ID, &jack_status_reg); + mic_detected = !(jack_status_reg & NAU8821_KEYDET); + + if (mic_detected) { + dev_dbg(nau8821->dev, "Headset connected\n"); + event |= SND_JACK_HEADSET; + + /* 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, NAU8821_MICBIAS_JKR2); + /* Latch Right Channel Analog data + * input into the Right Channel Filter + */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_R_SRC_EN, NAU8821_ADC_R_SRC_EN); + } else { + dev_dbg(nau8821->dev, "Headphone connected\n"); + event |= SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + } + event_mask |= SND_JACK_HEADSET; + snd_soc_jack_report(nau8821->jack, event, event_mask); + nau8821_sema_release(nau8821); +} + +/* Enable interruptions with internal clock. */ +static void nau8821_setup_inserted_irq(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Enable internal VCO needed for interruptions */ + if (nau8821->dapm->bias_level < SND_SOC_BIAS_PREPARE) + nau8821_configure_sysclk(nau8821, NAU8821_CLK_INTERNAL, 0); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, NAU8821_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, NAU8821_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, 0); + + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN, 0); + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_EJECT_DIS, 0); +} + +static irqreturn_t nau8821_interrupt(int irq, void *data) +{ + struct nau8821 *nau8821 = (struct nau8821 *)data; + struct regmap *regmap = nau8821->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + if (regmap_read(regmap, NAU8821_R10_IRQ_STATUS, &active_irq)) { + dev_err(nau8821->dev, "failed to read irq status\n"); + return IRQ_NONE; + } + + dev_dbg(nau8821->dev, "IRQ %d\n", active_irq); + + if ((active_irq & NAU8821_JACK_EJECT_IRQ_MASK) == + NAU8821_JACK_EJECT_DETECTED) { + regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, + NAU8821_MICDET_MASK, NAU8821_MICDET_DIS); + nau8821_eject_jack(nau8821); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8821_JACK_EJECT_IRQ_MASK; + } else if ((active_irq & NAU8821_JACK_INSERT_IRQ_MASK) == + NAU8821_JACK_INSERT_DETECTED) { + regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, + NAU8821_MICDET_MASK, NAU8821_MICDET_EN); + if (nau8821_is_jack_inserted(regmap)) { + /* detect microphone and jack type */ + cancel_work_sync(&nau8821->jdet_work); + schedule_work(&nau8821->jdet_work); + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_INSERT_DIS, + NAU8821_IRQ_INSERT_DIS); + regmap_update_bits(regmap, + NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_INSERT_EN); + nau8821_setup_inserted_irq(nau8821); + } else { + dev_warn(nau8821->dev, + "Inserted IRQ fired but not connected\n"); + nau8821_eject_jack(nau8821); + } + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8821_R11_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8821->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static const struct regmap_config nau8821_regmap_config = { + .val_bits = NAU8821_REG_DATA_LEN, + .reg_bits = NAU8821_REG_ADDR_LEN, + + .max_register = NAU8821_REG_MAX, + .readable_reg = nau8821_readable_reg, + .writeable_reg = nau8821_writeable_reg, + .volatile_reg = nau8821_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8821_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8821_reg_defaults), +}; + +static int nau8821_component_probe(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + nau8821->dapm = dapm; + + return 0; +} + +static void nau8821_component_remove(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (nau8821->irq) + /* Reset semaphore */ + nau8821_sema_reset(nau8821); +} + +/** + * nau8821_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau8821_calc_fll_param(unsigned int fll_in, + unsigned int fs, struct nau8821_fll *fll_param) +{ + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by + * dividing freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8821_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in >> fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } + } + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; + + /* Calculate the FLL 10-bit integer input and the FLL 24-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco_max << 24, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 24) & 0x3ff; + fll_param->fll_frac = fvco & 0xffffff; + + return 0; +} + +static void nau8821_fll_apply(struct nau8821 *nau8821, + struct nau8821_fll *fll_param) +{ + struct regmap *regmap = nau8821->regmap; + + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK | NAU8821_CLK_MCLK_SRC_MASK, + NAU8821_CLK_SRC_MCLK | fll_param->mclk_src); + /* Make DSP operate at high speed for better performance. */ + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_FLL_RATIO_MASK | NAU8821_ICTRL_LATCH_MASK, + fll_param->ratio | (0x6 << NAU8821_ICTRL_LATCH_SFT)); + /* FLL 24-bit fractional input */ + regmap_write(regmap, NAU8821_R0A_FLL7, + (fll_param->fll_frac >> 16) & 0xff); + regmap_write(regmap, NAU8821_R0B_FLL8, fll_param->fll_frac & 0xffff); + /* FLL 10-bit integer input */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(regmap, NAU8821_R07_FLL4, + NAU8821_HIGHBW_EN | NAU8821_FLL_REF_DIV_MASK, + NAU8821_HIGHBW_EN | + (fll_param->clk_ref_div << NAU8821_FLL_REF_DIV_SFT)); + /* select divided VCO input */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_CLK_SW_MASK, NAU8821_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(regmap, + NAU8821_R09_FLL6, NAU8821_DCO_EN, 0); + if (fll_param->fll_frac) { + /* set FLL loop filter enable and cutoff frequency at 500Khz */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_MASK, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_FILTER); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN | NAU8821_CUTOFF500, + NAU8821_SDM_EN | NAU8821_CUTOFF500); + } else { + /* disable FLL loop filter and cutoff frequency */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_MASK, NAU8821_FLL_FTR_SW_ACCU); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN | NAU8821_CUTOFF500, 0); + } +} + +/** + * nau8821_set_fll - FLL configuration of nau8821 + * @codec: codec component + * @freq_in: frequency of input clock source + * @freq_out: must be 256*Fs in order to achieve the best performance + * + * The FLL function can select BCLK or MCLK as the input clock source. + * + * Returns 0 if the parameters have been applied successfully + * or negative error code. + */ +static int nau8821_set_fll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, unsigned int freq_out) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct nau8821_fll fll_set_param, *fll_param = &fll_set_param; + int ret, fs; + + fs = freq_out >> 8; + ret = nau8821_calc_fll_param(freq_in, fs, fll_param); + if (ret) { + dev_err(nau8821->dev, + "Unsupported input clock %d to output clock %d\n", + freq_in, freq_out); + return ret; + } + dev_dbg(nau8821->dev, + "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param->mclk_src, fll_param->ratio, fll_param->fll_frac, + fll_param->fll_int, fll_param->clk_ref_div); + + nau8821_fll_apply(nau8821, fll_param); + mdelay(2); + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_VCO); + + return 0; +} + +static void nau8821_configure_mclk_as_sysclk(struct regmap *regmap) +{ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_DCO_EN, 0); + /* Make DSP operate as default setting for power saving. */ + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_ICTRL_LATCH_MASK, 0); +} + +static int nau8821_configure_sysclk(struct nau8821 *nau8821, + int clk_id, unsigned int freq) +{ + struct regmap *regmap = nau8821->regmap; + + switch (clk_id) { + case NAU8821_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + nau8821_configure_mclk_as_sysclk(regmap); + break; + case NAU8821_CLK_MCLK: + nau8821_sema_acquire(nau8821, HZ); + nau8821_configure_mclk_as_sysclk(regmap); + /* MCLK not changed by clock tree */ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_MCLK_SRC_MASK, 0); + nau8821_sema_release(nau8821); + break; + case NAU8821_CLK_INTERNAL: + if (nau8821_is_jack_inserted(regmap)) { + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_DCO_EN, NAU8821_DCO_EN); + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_VCO); + /* Decrease the VCO frequency and make DSP operate + * as default setting for power saving. + */ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_ICTRL_LATCH_MASK | + NAU8821_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN, NAU8821_SDM_EN); + } + break; + case NAU8821_CLK_FLL_MCLK: + nau8821_sema_acquire(nau8821, HZ); + /* Higher FLL reference input frequency can only set lower + * gain error, such as 0000 for input reference from MCLK + * 12.288Mhz. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_MCLK | 0); + nau8821_sema_release(nau8821); + break; + case NAU8821_CLK_FLL_BLK: + nau8821_sema_acquire(nau8821, HZ); + /* If FLL reference input is from low frequency source, + * higher error gain can apply such as 0xf which has + * the most sensitive gain error correction threshold, + * Therefore, FLL has the most accurate DCO to + * target frequency. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_BLK | + (0xf << NAU8821_GAIN_ERR_SFT)); + nau8821_sema_release(nau8821); + break; + case NAU8821_CLK_FLL_FS: + nau8821_sema_acquire(nau8821, HZ); + /* If FLL reference input is from low frequency source, + * higher error gain can apply such as 0xf which has + * the most sensitive gain error correction threshold, + * Therefore, FLL has the most accurate DCO to + * target frequency. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_FS | + (0xf << NAU8821_GAIN_ERR_SFT)); + nau8821_sema_release(nau8821); + break; + default: + dev_err(nau8821->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + nau8821->clk_id = clk_id; + dev_dbg(nau8821->dev, "Sysclk is %dHz and clock id is %d\n", freq, + nau8821->clk_id); + + return 0; +} + +static int nau8821_set_sysclk(struct snd_soc_component *component, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + return nau8821_configure_sysclk(nau8821, clk_id, freq); +} + +static int nau8821_resume_setup(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + if (nau8821->irq) { + /* Clear all interruption status */ + nau8821_int_status_clear_all(regmap); + + /* Enable both insertion and ejection interruptions, and then + * bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, 0); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, + NAU8821_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_INSERT_DIS | NAU8821_IRQ_EJECT_DIS, 0); + } + + return 0; +} + +static int nau8821_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct regmap *regmap = nau8821->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + /* Setup codec configuration after resume */ + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) + nau8821_resume_setup(nau8821); + break; + + case SND_SOC_BIAS_OFF: + /* HPL/HPR short to ground */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R | NAU8821_SPKR_DWN1L, 0); + if (nau8821->irq) { + /* Reset semaphore */ + nau8821_sema_reset(nau8821); + /* Reset the configuration of jack type for detection. + * Detach 2kOhm Resistors from MICBIAS to MICGND1/2. + */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, 0); + /* Turn off all interruptions before system shutdown. + * Keep theinterruption quiet before resume + * setup completes. + */ + regmap_write(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, 0xffff); + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN); + } + break; + } + + return 0; +} + +static int __maybe_unused nau8821_suspend(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (nau8821->irq) + disable_irq(nau8821->irq); + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8821->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8821->dapm); + regcache_cache_only(nau8821->regmap, true); + regcache_mark_dirty(nau8821->regmap); + + return 0; +} + +static int __maybe_unused nau8821_resume(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(nau8821->regmap, false); + regcache_sync(nau8821->regmap); + if (nau8821->irq) { + /* Hold semaphore to postpone playback happening + * until jack detection done. + */ + nau8821_sema_acquire(nau8821, 0); + enable_irq(nau8821->irq); + } + + return 0; +} + +static const struct snd_soc_component_driver nau8821_component_driver = { + .probe = nau8821_component_probe, + .remove = nau8821_component_remove, + .set_sysclk = nau8821_set_sysclk, + .set_pll = nau8821_set_fll, + .set_bias_level = nau8821_set_bias_level, + .suspend = nau8821_suspend, + .resume = nau8821_resume, + .controls = nau8821_controls, + .num_controls = ARRAY_SIZE(nau8821_controls), + .dapm_widgets = nau8821_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8821_dapm_widgets), + .dapm_routes = nau8821_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8821_dapm_routes), + .suspend_bias_off = 1, + .non_legacy_dai_naming = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, +}; + +/** + * nau8821_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8821_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + int ret; + + nau8821->jack = jack; + /* Initiate jack detection work queue */ + INIT_WORK(&nau8821->jdet_work, nau8821_jdet_work); + ret = devm_request_threaded_irq(nau8821->dev, nau8821->irq, NULL, + nau8821_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8821", nau8821); + if (ret) { + dev_err(nau8821->dev, "Cannot request irq %d (%d)\n", + nau8821->irq, ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(nau8821_enable_jack_detect); + +static void nau8821_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8821_R00_RESET, 0xffff); + regmap_write(regmap, NAU8821_R00_RESET, 0xffff); +} + +static void nau8821_print_device_properties(struct nau8821 *nau8821) +{ + struct device *dev = nau8821->dev; + + dev_dbg(dev, "jkdet-enable: %d\n", nau8821->jkdet_enable); + dev_dbg(dev, "jkdet-pull-enable: %d\n", nau8821->jkdet_pull_enable); + dev_dbg(dev, "jkdet-pull-up: %d\n", nau8821->jkdet_pull_up); + dev_dbg(dev, "jkdet-polarity: %d\n", nau8821->jkdet_polarity); + dev_dbg(dev, "micbias-voltage: %d\n", nau8821->micbias_voltage); + dev_dbg(dev, "vref-impedance: %d\n", nau8821->vref_impedance); + dev_dbg(dev, "jack-insert-debounce: %d\n", + nau8821->jack_insert_debounce); + dev_dbg(dev, "jack-eject-debounce: %d\n", + nau8821->jack_eject_debounce); + dev_dbg(dev, "dmic-clk-threshold: %d\n", + nau8821->dmic_clk_threshold); +} + +static int nau8821_read_device_properties(struct device *dev, + struct nau8821 *nau8821) +{ + int ret; + + nau8821->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8821->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8821->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8821->jkdet_polarity); + if (ret) + nau8821->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8821->micbias_voltage); + if (ret) + nau8821->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8821->vref_impedance); + if (ret) + nau8821->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8821->jack_insert_debounce); + if (ret) + nau8821->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8821->jack_eject_debounce); + if (ret) + nau8821->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,dmic-clk-threshold", + &nau8821->dmic_clk_threshold); + if (ret) + nau8821->dmic_clk_threshold = 3072000; + + return 0; +} + +static void nau8821_init_regs(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Enable Bias/Vmid */ + regmap_update_bits(regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_VMID, NAU8821_BIAS_VMID); + regmap_update_bits(regmap, NAU8821_R76_BOOST, + NAU8821_GLOBAL_BIAS_EN, NAU8821_GLOBAL_BIAS_EN); + /* VMID Tieoff setting and enable TESTDAC. + * This sets the analog DAC inputs to a '0' input signal to avoid + * any glitches due to power up transients in both the analog and + * digital DAC circuit. + */ + regmap_update_bits(regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_VMID_SEL_MASK | NAU8821_BIAS_TESTDAC_EN, + (nau8821->vref_impedance << NAU8821_BIAS_VMID_SEL_SFT) | + NAU8821_BIAS_TESTDAC_EN); + /* Disable short Frame Sync detection logic */ + regmap_update_bits(regmap, NAU8821_R1E_LEFT_TIME_SLOT, + NAU8821_DIS_FS_SHORT_DET, NAU8821_DIS_FS_SHORT_DET); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8821_R76_BOOST, + NAU8821_PRECHARGE_DIS | NAU8821_HP_BOOST_DIS | + NAU8821_HP_BOOST_G_DIS | NAU8821_SHORT_SHUTDOWN_EN, + NAU8821_PRECHARGE_DIS | NAU8821_HP_BOOST_DIS | + NAU8821_HP_BOOST_G_DIS | NAU8821_SHORT_SHUTDOWN_EN); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_TIMER_MASK, + 0x20 << NAU8821_CLASSG_TIMER_SFT); + /* Class AB bias current to 2x, DAC Capacitor enable MSB/LSB */ + regmap_update_bits(regmap, NAU8821_R6A_ANALOG_CONTROL_2, + NAU8821_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8821_DAC_CAPACITOR_MSB | NAU8821_DAC_CAPACITOR_LSB, + NAU8821_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8821_DAC_CAPACITOR_MSB | NAU8821_DAC_CAPACITOR_LSB); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_POWER_DOWN_DACR | NAU8821_POWER_DOWN_DACL, 0); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8821_R73_RDAC, + NAU8821_DAC_CLK_DELAY_MASK | NAU8821_DAC_VREF_MASK, + (0x2 << NAU8821_DAC_CLK_DELAY_SFT) | + (0x3 << NAU8821_DAC_VREF_SFT)); + + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_VOLTAGE_MASK, nau8821->micbias_voltage); + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_SYNC_DOWN_MASK, NAU8821_ADC_SYNC_DOWN_64); + regmap_update_bits(regmap, NAU8821_R2C_DAC_CTRL1, + NAU8821_DAC_OVERSAMPLE_MASK, NAU8821_DAC_OVERSAMPLE_64); +} + +static int nau8821_setup_irq(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + sema_init(&nau8821->jd_sem, 1); + + /* Jack detection */ + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_OUTPUT_EN, + nau8821->jkdet_enable ? 0 : NAU8821_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_PULL_EN, + nau8821->jkdet_pull_enable ? 0 : NAU8821_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_PULL_UP, + nau8821->jkdet_pull_up ? NAU8821_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8821->jkdet_polarity ? 0 : NAU8821_JACK_POLARITY); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_INSERT_DEBOUNCE_MASK, + nau8821->jack_insert_debounce << + NAU8821_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_EJECT_DEBOUNCE_MASK, + nau8821->jack_eject_debounce << + NAU8821_JACK_EJECT_DEBOUNCE_SFT); + /* Pull up IRQ pin */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_PIN_PULL_UP | NAU8821_IRQ_PIN_PULL_EN | + NAU8821_IRQ_OUTPUT_EN, NAU8821_IRQ_PIN_PULL_UP | + NAU8821_IRQ_PIN_PULL_EN | NAU8821_IRQ_OUTPUT_EN); + /* Disable interruption before codec initiation done */ + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, 0x3f5, 0x3f5); + + return 0; +} + +static int nau8821_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8821 *nau8821 = dev_get_platdata(&i2c->dev); + int ret, value; + + if (!nau8821) { + nau8821 = devm_kzalloc(dev, sizeof(*nau8821), GFP_KERNEL); + if (!nau8821) + return -ENOMEM; + nau8821_read_device_properties(dev, nau8821); + } + i2c_set_clientdata(i2c, nau8821); + + nau8821->regmap = devm_regmap_init_i2c(i2c, &nau8821_regmap_config); + if (IS_ERR(nau8821->regmap)) + return PTR_ERR(nau8821->regmap); + + nau8821->dev = dev; + nau8821->irq = i2c->irq; + nau8821_print_device_properties(nau8821); + + nau8821_reset_chip(nau8821->regmap); + ret = regmap_read(nau8821->regmap, NAU8821_R58_I2C_DEVICE_ID, &value); + if (ret) { + dev_err(dev, "Failed to read device id (%d)\n", ret); + return ret; + } + nau8821_init_regs(nau8821); + + if (i2c->irq) + nau8821_setup_irq(nau8821); + + ret = devm_snd_soc_register_component(&i2c->dev, + &nau8821_component_driver, &nau8821_dai, 1); + + return ret; +} + +static const struct i2c_device_id nau8821_i2c_ids[] = { + { "nau8821", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8821_i2c_ids); + +#ifdef CONFIG_OF +static const struct of_device_id nau8821_of_ids[] = { + { .compatible = "nuvoton,nau8821", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8821_of_ids); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8821_acpi_match[] = { + { "NVTN2020", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8821_acpi_match); +#endif + +static struct i2c_driver nau8821_driver = { + .driver = { + .name = "nau8821", + .of_match_table = of_match_ptr(nau8821_of_ids), + .acpi_match_table = ACPI_PTR(nau8821_acpi_match), + }, + .probe = nau8821_i2c_probe, + .id_table = nau8821_i2c_ids, +}; +module_i2c_driver(nau8821_driver); + +MODULE_DESCRIPTION("ASoC nau8821 driver"); +MODULE_AUTHOR("John Hsu <kchsu0(a)nuvoton.com>"); +MODULE_AUTHOR("Seven Lee <wtli(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h new file mode 100644 index 000000000000..3541decc34f0 --- /dev/null +++ b/sound/soc/codecs/nau8821.h @@ -0,0 +1,533 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * NAU88L21 ALSA SoC audio driver + * + * Copyright 2021 Nuvoton Technology Corp. + * Author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Lee <wtli(a)nuvoton.com> + */ + +#ifndef __NAU8821_H__ +#define __NAU8821_H__ + +#define NAU8821_R00_RESET 0x00 +#define NAU8821_R01_ENA_CTRL 0x01 +#define NAU8821_R03_CLK_DIVIDER 0x03 +#define NAU8821_R04_FLL1 0x04 +#define NAU8821_R05_FLL2 0x05 +#define NAU8821_R06_FLL3 0x06 +#define NAU8821_R07_FLL4 0x07 +#define NAU8821_R08_FLL5 0x08 +#define NAU8821_R09_FLL6 0x09 +#define NAU8821_R0A_FLL7 0x0a +#define NAU8821_R0B_FLL8 0x0b +#define NAU8821_R0D_JACK_DET_CTRL 0x0d +#define NAU8821_R0F_INTERRUPT_MASK 0x0f +#define NAU8821_R10_IRQ_STATUS 0x10 +#define NAU8821_R11_INT_CLR_KEY_STATUS 0x11 +#define NAU8821_R12_INTERRUPT_DIS_CTRL 0x12 +#define NAU8821_R13_DMIC_CTRL 0x13 +#define NAU8821_R1A_GPIO12_CTRL 0x1a +#define NAU8821_R1B_TDM_CTRL 0x1b +#define NAU8821_R1C_I2S_PCM_CTRL1 0x1c +#define NAU8821_R1D_I2S_PCM_CTRL2 0x1d +#define NAU8821_R1E_LEFT_TIME_SLOT 0x1e +#define NAU8821_R1F_RIGHT_TIME_SLOT 0x1f +#define NAU8821_R21_BIQ0_COF1 0x21 +#define NAU8821_R22_BIQ0_COF2 0x22 +#define NAU8821_R23_BIQ0_COF3 0x23 +#define NAU8821_R24_BIQ0_COF4 0x24 +#define NAU8821_R25_BIQ0_COF5 0x25 +#define NAU8821_R26_BIQ0_COF6 0x26 +#define NAU8821_R27_BIQ0_COF7 0x27 +#define NAU8821_R28_BIQ0_COF8 0x28 +#define NAU8821_R29_BIQ0_COF9 0x29 +#define NAU8821_R2A_BIQ0_COF10 0x2a +#define NAU8821_R2B_ADC_RATE 0x2b +#define NAU8821_R2C_DAC_CTRL1 0x2c +#define NAU8821_R2D_DAC_CTRL2 0x2d +#define NAU8821_R2F_DAC_DGAIN_CTRL 0x2f +#define NAU8821_R30_ADC_DGAIN_CTRL 0x30 +#define NAU8821_R31_MUTE_CTRL 0x31 +#define NAU8821_R32_HSVOL_CTRL 0x32 +#define NAU8821_R34_DACR_CTRL 0x34 +#define NAU8821_R35_ADC_DGAIN_CTRL1 0x35 +#define NAU8821_R36_ADC_DRC_KNEE_IP12 0x36 +#define NAU8821_R37_ADC_DRC_KNEE_IP34 0x37 +#define NAU8821_R38_ADC_DRC_SLOPES 0x38 +#define NAU8821_R39_ADC_DRC_ATKDCY 0x39 +#define NAU8821_R3A_DAC_DRC_KNEE_IP12 0x3a +#define NAU8821_R3B_DAC_DRC_KNEE_IP34 0x3b +#define NAU8821_R3C_DAC_DRC_SLOPES 0x3c +#define NAU8821_R3D_DAC_DRC_ATKDCY 0x3d +#define NAU8821_R41_BIQ1_COF1 0x41 +#define NAU8821_R42_BIQ1_COF2 0x42 +#define NAU8821_R43_BIQ1_COF3 0x43 +#define NAU8821_R44_BIQ1_COF4 0x44 +#define NAU8821_R45_BIQ1_COF5 0x45 +#define NAU8821_R46_BIQ1_COF6 0x46 +#define NAU8821_R47_BIQ1_COF7 0x47 +#define NAU8821_R48_BIQ1_COF8 0x48 +#define NAU8821_R49_BIQ1_COF9 0x49 +#define NAU8821_R4A_BIQ1_COF10 0x4a +#define NAU8821_R4B_CLASSG_CTRL 0x4b +#define NAU8821_R4C_IMM_MODE_CTRL 0x4c +#define NAU8821_R4D_IMM_RMS_L 0x4d +#define NAU8821_R4E_FUSE_CTRL2 0x4e +#define NAU8821_R4F_FUSE_CTRL3 0x4f +#define NAU8821_R51_FUSE_CTRL1 0x51 +#define NAU8821_R53_OTPDOUT_1 0x53 +#define NAU8821_R54_OTPDOUT_2 0x54 +#define NAU8821_R55_MISC_CTRL 0x55 +#define NAU8821_R58_I2C_DEVICE_ID 0x58 +#define NAU8821_R59_SARDOUT_RAM_STATUS 0x59 +#define NAU8821_R5A_SOFTWARE_RST 0x5a +#define NAU8821_R66_BIAS_ADJ 0x66 +#define NAU8821_R68_TRIM_SETTINGS 0x68 +#define NAU8821_R69_ANALOG_CONTROL_1 0x69 +#define NAU8821_R6A_ANALOG_CONTROL_2 0x6a +#define NAU8821_R6B_PGA_MUTE 0x6b +#define NAU8821_R71_ANALOG_ADC_1 0x71 +#define NAU8821_R72_ANALOG_ADC_2 0x72 +#define NAU8821_R73_RDAC 0x73 +#define NAU8821_R74_MIC_BIAS 0x74 +#define NAU8821_R76_BOOST 0x76 +#define NAU8821_R77_FEPGA 0x77 +#define NAU8821_R7E_PGA_GAIN 0x7e +#define NAU8821_R7F_POWER_UP_CONTROL 0x7f +#define NAU8821_R80_CHARGE_PUMP 0x80 +#define NAU8821_R81_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8821_R82_GENERAL_STATUS 0x82 +#define NAU8821_REG_MAX NAU8821_R82_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8821_REG_ADDR_LEN 16 +#define NAU8821_REG_DATA_LEN 16 + +/* ENA_CTRL (0x01) */ +#define NAU8821_CLK_DAC_INV_SFT 14 +#define NAU8821_CLK_DAC_INV (0x1 << NAU8821_CLK_DAC_INV) +#define NAU8821_EN_DACR_SFT 11 +#define NAU8821_EN_DACR (0x1 << NAU8821_EN_DACR_SFT) +#define NAU8821_EN_DACL_SFT 10 +#define NAU8821_EN_DACL (0x1 << NAU8821_EN_DACL_SFT) +#define NAU8821_EN_ADCR_SFT 9 +#define NAU8821_EN_ADCR (0x1 << NAU8821_EN_ADCR_SFT) +#define NAU8821_EN_ADCL_SFT 8 +#define NAU8821_EN_ADCL (0x1 << NAU8821_EN_ADCL_SFT) +#define NAU8821_EN_ADC_CLK_SFT 7 +#define NAU8821_EN_ADC_CLK (0x1 << NAU8821_EN_ADC_CLK_SFT) +#define NAU8821_EN_DAC_CLK_SFT 6 +#define NAU8821_EN_DAC_CLK (0x1 << NAU8821_EN_DAC_CLK_SFT) +#define NAU8821_EN_I2S_CLK_SFT 4 +#define NAU8821_EN_I2S_CLK (0x1 << NAU8821_EN_I2S_CLK_SFT) +#define NAU8821_EN_DRC_CLK_SFT 0 +#define NAU8821_EN_DRC_CLK (0x1 << NAU8821_EN_DRC_CLK_SFT) + +/* CLK_DIVIDER (0x03) */ +#define NAU8821_CLK_SRC_SFT 15 +#define NAU8821_CLK_SRC_MASK (0x1 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_SRC_VCO (0x1 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_SRC_MCLK (0x0 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_SFT 13 +#define NAU8821_CLK_CODEC_SRC_MASK (0x1 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_VCO (0x1 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_MCLK (0x0 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_ADC_SRC_SFT 6 +#define NAU8821_CLK_ADC_SRC_MASK (0x3 << NAU8821_CLK_ADC_SRC_SFT) +#define NAU8821_CLK_DAC_SRC_SFT 4 +#define NAU8821_CLK_DAC_SRC_MASK (0x3 << NAU8821_CLK_DAC_SRC_SFT) +#define NAU8821_CLK_MCLK_SRC_MASK 0xf + +/* FLL1 (0x04) */ +#define NAU8821_ICTRL_LATCH_SFT 10 +#define NAU8821_ICTRL_LATCH_MASK (0x7 << NAU8821_ICTRL_LATCH_SFT) +#define NAU8821_FLL_RATIO_MASK 0x7f + +/* FLL3 (0x06) */ +#define NAU8821_GAIN_ERR_SFT 12 +#define NAU8821_GAIN_ERR_MASK (0xf << NAU8821_GAIN_ERR_SFT) +#define NAU8821_FLL_CLK_SRC_SFT 10 +#define NAU8821_FLL_CLK_SRC_MASK (0x3 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_FS (0x3 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_BLK (0x2 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_MCLK (0x0 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_INTEGER_MASK 0x3ff + +/* FLL4 (0x07) */ +#define NAU8821_HIGHBW_EN_SFT 15 +#define NAU8821_HIGHBW_EN (0x1 << NAU8821_HIGHBW_EN_SFT) +#define NAU8821_FLL_REF_DIV_SFT 10 +#define NAU8821_FLL_REF_DIV_MASK (0x3 << NAU8821_FLL_REF_DIV_SFT) + +/* FLL5 (0x08) */ +#define NAU8821_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8821_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8821_FLL_CLK_SW_SFT 13 +#define NAU8821_FLL_CLK_SW_MASK (0x1 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_CLK_SW_N2 (0x1 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_CLK_SW_REF (0x0 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_FTR_SW_SFT 12 +#define NAU8821_FLL_FTR_SW_MASK (0x1 << NAU8821_FLL_FTR_SW_SFT) +#define NAU8821_FLL_FTR_SW_ACCU (0x1 << NAU8821_FLL_FTR_SW_SFT) +#define NAU8821_FLL_FTR_SW_FILTER (0x0 << NAU8821_FLL_FTR_SW_SFT) + +/* FLL6 (0x09) */ +#define NAU8821_DCO_EN (0x1 << 15) +#define NAU8821_SDM_EN (0x1 << 14) +#define NAU8821_CUTOFF500 (0x1 << 13) + +/* FLL7 (0x0a) */ +#define NAU8821_FLL_FRACH_MASK 0xff + +/* FLL8 (0x0b) */ +#define NAU8821_FLL_FRACL_MASK 0xffff + +/* JACK_DET_CTRL (0x0d) */ +/* 0 - open, 1 - short to GND */ +#define NAU8821_SPKR_DWN1R_SFT 15 +#define NAU8821_SPKR_DWN1R (0x1 << NAU8821_SPKR_DWN1R_SFT) +#define NAU8821_SPKR_DWN1L_SFT 14 +#define NAU8821_SPKR_DWN1L (0x1 << NAU8821_SPKR_DWN1L_SFT) +#define NAU8821_JACK_DET_RESTART (0x1 << 9) +#define NAU8821_JACK_DET_DB_BYPASS (0x1 << 8) +#define NAU8821_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8821_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8821_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8821_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8821_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8821_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8821_JACK_POLARITY (0x1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0x0f) */ +#define NAU8821_IRQ_PIN_PULL_UP (0x1 << 14) +#define NAU8821_IRQ_PIN_PULL_EN (0x1 << 13) +#define NAU8821_IRQ_OUTPUT_EN (0x1 << 11) +#define NAU8821_IRQ_RMS_EN (0x1 << 8) +#define NAU8821_IRQ_KEY_RELEASE_EN (0x1 << 7) +#define NAU8821_IRQ_KEY_PRESS_EN (0x1 << 6) +#define NAU8821_IRQ_MIC_DET_EN (0x1 << 4) +#define NAU8821_IRQ_EJECT_EN (0x1 << 2) +#define NAU8821_IRQ_INSERT_EN 0x1 + +/* IRQ_STATUS (0x10) */ +#define NAU8821_SHORT_CIRCUIT_IRQ (0x1 << 9) +#define NAU8821_IMPEDANCE_MEAS_IRQ (0x1 << 8) +#define NAU8821_KEY_IRQ_SFT 6 +#define NAU8821_KEY_IRQ_MASK (0x3 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_KEY_RELEASE_IRQ (0x2 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_KEY_SHORT_PRESS_IRQ (0x1 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_MIC_DETECT_IRQ (0x1 << 4) +#define NAU8821_JACK_EJECT_IRQ_MASK (0x3 << 2) +#define NAU8821_JACK_EJECT_DETECTED (0x1 << 2) +#define NAU8821_JACK_INSERT_IRQ_MASK 0x3 +#define NAU8821_JACK_INSERT_DETECTED 0x1 + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8821_IRQ_KEY_RELEASE_DIS (0x1 << 7) +#define NAU8821_IRQ_KEY_PRESS_DIS (0x1 << 6) +#define NAU8821_IRQ_MIC_DIS (0x1 << 4) +#define NAU8821_IRQ_EJECT_DIS (0x1 << 2) +#define NAU8821_IRQ_INSERT_DIS 0x1 + +/* DMIC_CTRL (0x13) */ +#define NAU8821_DMIC_DS_SFT 7 +#define NAU8821_DMIC_DS_MASK (0x1 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_DS_HIGH (0x1 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_DS_LOW (0x0 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_SRC_SFT 1 +#define NAU8821_DMIC_SRC_MASK (0x3 << NAU8821_DMIC_SRC_SFT) +#define NAU8821_CLK_DMIC_SRC (0x2 << NAU8821_DMIC_SRC_SFT) +#define NAU8821_DMIC_EN_SFT 0 + +/* GPIO12_CTRL (0x1a) */ +#define NAU8821_JKDET_PULL_UP (0x1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8821_JKDET_PULL_EN (0x1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8821_JKDET_OUTPUT_EN (0x1 << 8) /* 0 - enable input, 1 - enable output */ + +/* TDM_CTRL (0x1b) */ +#define NAU8821_TDM_EN_SFT 15 +#define NAU8821_TDM_EN (0x1 << NAU8821_TDM_EN_SFT) +#define NAU8821_DACL_CH_SFT 7 +#define NAU8821_DACL_CH_MASK (0x7 << NAU8821_DACL_CH_SFT) +#define NAU8821_DACR_CH_SFT 4 +#define NAU8821_DACR_CH_MASK (0x7 << NAU8821_DACR_CH_SFT) +#define NAU8821_ADCL_CH_SFT 2 +#define NAU8821_ADCL_CH_MASK (0x3 << NAU8821_ADCL_CH_SFT) +#define NAU8821_ADCR_CH_SFT 0 +#define NAU8821_ADCR_CH_MASK 0x3 + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8821_I2S_BP_SFT 7 +#define NAU8821_I2S_BP_MASK (0x1 << NAU8821_I2S_BP_SFT) +#define NAU8821_I2S_BP_INV (0x1 << NAU8821_I2S_BP_SFT) +#define NAU8821_I2S_PCMB_SFT 6 +#define NAU8821_I2S_PCMB_MASK (0x1 << NAU8821_I2S_PCMB_SFT) +#define NAU8821_I2S_PCMB_EN (0x1 << NAU8821_I2S_PCMB_SFT) +#define NAU8821_I2S_DL_SFT 2 +#define NAU8821_I2S_DL_MASK (0x3 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_32 (0x3 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_24 (0x2 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_20 (0x1 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_16 (0x0 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DF_MASK 0x3 +#define NAU8821_I2S_DF_PCM_AB 0x3 +#define NAU8821_I2S_DF_I2S 0x2 +#define NAU8821_I2S_DF_LEFT 0x1 +#define NAU8821_I2S_DF_RIGTH 0x0 + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8821_I2S_TRISTATE_SFT 15 +#define NAU8821_I2S_TRISTATE (0x1 << NAU8821_I2S_TRISTATE_SFT) +#define NAU8821_I2S_LRC_DIV_SFT 12 +#define NAU8821_I2S_LRC_DIV_MASK (0x3 << NAU8821_I2S_LRC_DIV_SFT) +#define NAU8821_I2S_MS_SFT 3 +#define NAU8821_I2S_MS_MASK (0x1 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_MS_MASTER (0x1 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_MS_SLAVE (0x0 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_BLK_DIV_MASK 0x7 + +/* LEFT_TIME_SLOT (0x1e) */ +#define NAU8821_TSLOT_L_OFFSET_MASK 0x3ff +#define NAU8821_DIS_FS_SHORT_DET (0x1 << 13) + +/* RIGHT_TIME_SLOT (0x1f) */ +#define NAU8821_TSLOT_R_OFFSET_MASK 0x3ff + +/* BIQ0_COF10 (0x2a) */ +#define NAU8821_BIQ0_ADC_EN_SFT 3 +#define NAU8821_BIQ0_ADC_EN_EN (0x1 << NAU8821_BIQ0_ADC_EN_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8821_ADC_SYNC_DOWN_SFT 0 +#define NAU8821_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8821_ADC_SYNC_DOWN_256 0x3 +#define NAU8821_ADC_SYNC_DOWN_128 0x2 +#define NAU8821_ADC_SYNC_DOWN_64 0x1 +#define NAU8821_ADC_SYNC_DOWN_32 0x0 +#define NAU8821_ADC_L_SRC_SFT 15 +#define NAU8821_ADC_L_SRC_EN (0x1 << NAU8821_ADC_L_SRC_SFT) +#define NAU8821_ADC_R_SRC_SFT 14 +#define NAU8821_ADC_R_SRC_EN (0x1 << NAU8821_ADC_R_SRC_SFT) + +/* DAC_CTRL1 (0x2c) */ +#define NAU8821_DAC_OVERSAMPLE_SFT 0 +#define NAU8821_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8821_DAC_OVERSAMPLE_32 0x4 +#define NAU8821_DAC_OVERSAMPLE_128 0x2 +#define NAU8821_DAC_OVERSAMPLE_256 0x1 +#define NAU8821_DAC_OVERSAMPLE_64 0x0 + +/* DAC_DGAIN_CTRL (0x2f) */ +#define NAU8821_DAC1_TO_DAC0_ST_SFT 8 +#define NAU8821_DAC1_TO_DAC0_ST_MASK (0xff << NAU8821_DAC1_TO_DAC0_ST_SFT) +#define NAU8821_DAC0_TO_DAC1_ST_SFT 0 +#define NAU8821_DAC0_TO_DAC1_ST_MASK 0xff + +/* MUTE_CTRL (0x31) */ +#define NAU8821_DAC_ZC_EN (0x1 << 12) +#define NAU8821_DAC_SOFT_MUTE (0x1 << 9) +#define NAU8821_ADC_ZC_EN (0x1 << 2) +#define NAU8821_ADC_SOFT_MUTE (0x1 << 1) + +/* HSVOL_CTRL (0x32) */ +#define NAU8821_HP_MUTE (0x1 << 15) +#define NAU8821_HP_MUTE_AUTO (0x1 << 14) +#define NAU8821_HPL_MUTE (0x1 << 13) +#define NAU8821_HPR_MUTE (0x1 << 12) +#define NAU8821_HPL_VOL_SFT 4 +#define NAU8821_HPL_VOL_MASK (0x3 << NAU8821_HPL_VOL_SFT) +#define NAU8821_HPR_VOL_SFT 0 +#define NAU8821_HPR_VOL_MASK (0x3 << NAU8821_HPR_VOL_SFT) + +/* DACR_CTRL (0x34) */ +#define NAU8821_DACR_CH_VOL_SFT 8 +#define NAU8821_DACR_CH_VOL_MASK (0xff << NAU8821_DACR_CH_VOL_SFT) +#define NAU8821_DACL_CH_VOL_SFT 0 +#define NAU8821_DACL_CH_VOL_MASK 0xff + +/* ADC_DGAIN_CTRL1 (0x35) */ +#define NAU8821_ADCR_CH_VOL_SFT 8 +#define NAU8821_ADCR_CH_VOL_MASK (0xff << NAU8821_ADCR_CH_VOL_SFT) +#define NAU8821_ADCL_CH_VOL_SFT 0 +#define NAU8821_ADCL_CH_VOL_MASK 0xff + +/* BIQ1_COF10 (0x4a) */ +#define NAU8821_BIQ1_DAC_EN_SFT 3 +#define NAU8821_BIQ1_DAC_EN_EN (0x1 << NAU8821_BIQ1_DAC_EN_SFT) + +/* CLASSG_CTRL (0x4b) */ +#define NAU8821_CLASSG_TIMER_SFT 8 +#define NAU8821_CLASSG_TIMER_MASK (0x3f << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_64MS (0x20 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_32MS (0x10 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_16MS (0x8 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_8MS (0x4 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_2MS (0x2 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_1MS (0x1 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_RDAC_EN_SFT 2 +#define NAU8821_CLASSG_RDAC_EN (0x1 << NAU8821_CLASSG_RDAC_EN_SFT) +#define NAU8821_CLASSG_LDAC_EN_SFT 1 +#define NAU8821_CLASSG_LDAC_EN (0x1 << NAU8821_CLASSG_LDAC_EN_SFT) +#define NAU8821_CLASSG_EN_SFT 0 +#define NAU8821_CLASSG_EN 0x1 + +/* IMM_MODE_CTRL (0x4C) */ +#define NAU8821_IMM_THD_SFT 8 +#define NAU8821_IMM_THD_MASK (0x3f << NAU8821_IMM_THD_SFT) +#define NAU8821_IMM_GEN_VOL_SFT 6 +#define NAU8821_IMM_GEN_VOL_MASK (0x3 << NAU8821_IMM_GEN_VOL_SFT) +#define NAU8821_IMM_CYC_SFT 4 +#define NAU8821_IMM_CYC_MASK (0x3 << NAU8821_IMM_CYC_SFT) +#define NAU8821_IMM_EN (0x1 << 3) +#define NAU8821_IMM_DAC_SRC_MASK 0x3 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8821_KEYDET (0x1 << 7) +#define NAU8821_MICDET (0x1 << 6) +#define NAU8821_SOFTWARE_ID_MASK 0x3 + +/* BIAS_ADJ (0x66) */ +#define NAU8821_BIAS_HP_IMP (0x1 << 15) +#define NAU8821_BIAS_TESTDAC_SFT 8 +#define NAU8821_BIAS_TESTDAC_EN (0x3 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_TESTDACR_EN (0x2 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_TESTDACL_EN (0x1 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_VMID (0x1 << 6) +#define NAU8821_BIAS_VMID_SEL_SFT 4 +#define NAU8821_BIAS_VMID_SEL_MASK (0x3 << NAU8821_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_1 (0x69) */ +#define NAU8821_JD_POL_SFT 2 +#define NAU8821_JD_POL_MASK (0x1 << NAU8821_JD_POL_SFT) +#define NAU8821_JD_POL_INV (0x1 << NAU8821_JD_POL_SFT) +#define NAU8821_JD_OUT_POL_SFT 1 +#define NAU8821_JD_OUT_POL_MASK (0x1 << NAU8821_JD_OUT_POL_SFT) +#define NAU8821_JD_OUT_POL_INV (0x1 << NAU8821_JD_OUT_POL_SFT) +#define NAU8821_JD_EN_SFT 0 +#define NAU8821_JD_EN 0x1 + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8821_HP_NON_CLASSG_CURRENT_2xADJ (0x1 << 12) +#define NAU8821_DAC_CAPACITOR_MSB (0x1 << 1) +#define NAU8821_DAC_CAPACITOR_LSB 0x1 + +/* ANALOG_ADC_1 (0x71) */ +#define NAU8821_MICDET_EN_SFT 0 +#define NAU8821_MICDET_MASK 0x1 +#define NAU8821_MICDET_DIS 0x1 +#define NAU8821_MICDET_EN 0x0 + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8821_ADC_VREFSEL_SFT 8 +#define NAU8821_ADC_VREFSEL_MASK (0x3 << NAU8821_ADC_VREFSEL_SFT) +#define NAU8821_POWERUP_ADCL_SFT 6 +#define NAU8821_POWERUP_ADCL (0x1 << NAU8821_POWERUP_ADCL_SFT) +#define NAU8821_POWERUP_ADCR_SFT 4 +#define NAU8821_POWERUP_ADCR (0x1 << NAU8821_POWERUP_ADCR_SFT) + +/* RDAC (0x73) */ +#define NAU8821_DACR_EN_SFT 13 +#define NAU8821_DACR_EN (0x3 << NAU8821_DACR_EN_SFT) +#define NAU8821_DACL_EN_SFT 12 +#define NAU8821_DACL_EN (0x3 << NAU8821_DACL_EN_SFT) +#define NAU8821_DACR_CLK_EN_SFT 9 +#define NAU8821_DACR_CLK_EN (0x3 << NAU8821_DACR_CLK_EN_SFT) +#define NAU8821_DACL_CLK_EN_SFT 8 +#define NAU8821_DACL_CLK_EN (0x3 << NAU8821_DACL_CLK_EN_SFT) +#define NAU8821_DAC_CLK_DELAY_SFT 4 +#define NAU8821_DAC_CLK_DELAY_MASK (0x7 << NAU8821_DAC_CLK_DELAY_SFT) +#define NAU8821_DAC_VREF_SFT 2 +#define NAU8821_DAC_VREF_MASK (0x3 << NAU8821_DAC_VREF_SFT) + +/* MIC_BIAS (0x74) */ +#define NAU8821_MICBIAS_JKR2 (0x1 << 12) +#define NAU8821_MICBIAS_POWERUP_SFT 8 +#define NAU8821_MICBIAS_VOLTAGE_SFT 0 +#define NAU8821_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8821_PRECHARGE_DIS (0x1 << 13) +#define NAU8821_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8821_HP_BOOST_DIS_SFT 9 +#define NAU8821_HP_BOOST_DIS (0x1 << NAU8821_HP_BOOST_DIS_SFT) +#define NAU8821_HP_BOOST_G_DIS (0x1 << 8) +#define NAU8821_SHORT_SHUTDOWN_EN (0x1 << 6) + +/* FEPGA (0x77) */ +#define NAU8821_FEPGA_MODEL_SFT 4 +#define NAU8821_FEPGA_MODEL_MASK (0xf << NAU8821_FEPGA_MODEL_SFT) +#define NAU8821_FEPGA_MODER_SFT 0 +#define NAU8821_FEPGA_MODER_MASK 0xf + +/* PGA_GAIN (0x7e) */ +#define NAU8821_PGA_GAIN_L_SFT 8 +#define NAU8821_PGA_GAIN_L_MASK (0x3f << NAU8821_PGA_GAIN_L_SFT) +#define NAU8821_PGA_GAIN_R_SFT 0 +#define NAU8821_PGA_GAIN_R_MASK 0x3f + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8821_PUP_PGA_L_SFT 15 +#define NAU8821_PUP_PGA_L (0x1 << NAU8821_PUP_PGA_L_SFT) +#define NAU8821_PUP_PGA_R_SFT 14 +#define NAU8821_PUP_PGA_R (0x1 << NAU8821_PUP_PGA_R_SFT) +#define NAU8821_PUP_INTEG_R_SFT 5 +#define NAU8821_PUP_INTEG_R (0x1 << NAU8821_PUP_INTEG_R_SFT) +#define NAU8821_PUP_INTEG_L_SFT 4 +#define NAU8821_PUP_INTEG_L (0x1 << NAU8821_PUP_INTEG_L_SFT) +#define NAU8821_PUP_DRV_INSTG_R_SFT 3 +#define NAU8821_PUP_DRV_INSTG_R (0x1 << NAU8821_PUP_DRV_INSTG_R_SFT) +#define NAU8821_PUP_DRV_INSTG_L_SFT 2 +#define NAU8821_PUP_DRV_INSTG_L (0x1 << NAU8821_PUP_DRV_INSTG_L_SFT) +#define NAU8821_PUP_MAIN_DRV_R_SFT 1 +#define NAU8821_PUP_MAIN_DRV_R (0x1 << NAU8821_PUP_MAIN_DRV_R_SFT) +#define NAU8821_PUP_MAIN_DRV_L_SFT 0 +#define NAU8821_PUP_MAIN_DRV_L 0x1 + +/* CHARGE_PUMP (0x80) */ +#define NAU8821_JAMNODCLOW (0x1 << 10) +#define NAU8821_POWER_DOWN_DACR_SFT 9 +#define NAU8821_POWER_DOWN_DACR (0x1 << NAU8821_POWER_DOWN_DACR_SFT) +#define NAU8821_POWER_DOWN_DACL_SFT 8 +#define NAU8821_POWER_DOWN_DACL (0x1 << NAU8821_POWER_DOWN_DACL_SFT) +#define NAU8821_CHANRGE_PUMP_EN_SFT 5 +#define NAU8821_CHANRGE_PUMP_EN (0x1 << NAU8821_CHANRGE_PUMP_EN_SFT) + +/* GENERAL_STATUS (0x82) */ +#define NAU8821_GPIO2_IN_SFT 1 +#define NAU8821_GPIO2_IN (0x1 << NAU8821_GPIO2_IN_SFT) + +#define NUVOTON_CODEC_DAI "nau8821-hifi" + +/* System Clock Source */ +enum { + NAU8821_CLK_DIS, + NAU8821_CLK_MCLK, + NAU8821_CLK_INTERNAL, + NAU8821_CLK_FLL_MCLK, + NAU8821_CLK_FLL_BLK, + NAU8821_CLK_FLL_FS, +}; + +struct nau8821 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct semaphore jd_sem; + struct work_struct jdet_work; + int irq; + int clk_id; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int jack_insert_debounce; + int jack_eject_debounce; + int fs; + int dmic_clk_threshold; +}; + +int nau8821_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif /* __NAU8821_H__ */ -- 2.25.1

6 9

[PATCH v2 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 15 Sep '21

15 Sep '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. These patches depends on the dt-bindings header patch -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Chnages Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 169 ++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 343 +++++++++++++++++++++ 4 files changed, 526 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 8

[PATCH] ASoC: atmel: Convert to new style DAI format definitions
by Mark Brown 15 Sep '21

15 Sep '21

Convert the Atmel drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 4 ++-- sound/soc/atmel/atmel_ssc_dai.c | 10 +++++----- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++++---- sound/soc/atmel/mikroe-proto.c | 4 ++-- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- 7 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6b3d9c05eaf2..e119abbca099 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -343,7 +343,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* codec is slave, so cpu is master */ mr |= ATMEL_I2SC_MR_MODE_MASTER; ret = atmel_i2s_get_gck_param(dev, params_rate(params)); @@ -351,7 +351,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return ret; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* codec is master, so cpu is slave */ mr |= ATMEL_I2SC_MR_MODE_SLAVE; dev->gck_param = NULL; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6a63e8797a0b..4846a8958337 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -210,7 +210,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, return frame_size; switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) && ssc->clk_from_rk_pin) /* Receiver Frame Synchro (i.e. capture) @@ -220,7 +220,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, mck_div = 3; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) && !ssc->clk_from_rk_pin) /* Transmit Frame Synchro (i.e. playback) @@ -233,7 +233,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: r.num = ssc_p->mck_rate / mck_div / frame_size; ret = snd_interval_ratnum(i, 1, &r, &num, &den); @@ -243,8 +243,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBP_CFP: t.min = 8000; t.max = ssc_p->mck_rate / mck_div / frame_size; t.openmin = t.openmax = 0; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 9e237580afa9..9c974c4e187d 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -66,7 +66,7 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { .stream_name = "WM8904 PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &atmel_asoc_wm8904_ops, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 8988f024a732..67550e3d9ee3 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -350,7 +350,7 @@ static int mchp_i2s_mcc_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; /* We can't generate only FSYNC */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFS) + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBP_CFC) return -EINVAL; /* We can only reconfigure the IP when it's stopped */ @@ -547,19 +547,19 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* cpu is BCLK and LRC master */ mra |= MCHP_I2SMCC_MRA_MODE_MASTER; if (dev->sysclk) mra |= MCHP_I2SMCC_MRA_IMCKMODE_GEN; set_divs = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: /* cpu is BCLK master */ mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; set_divs = 1; fallthrough; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* cpu is slave */ mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; if (dev->sysclk) diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index 0be7b4221c14..f9331f7e80fe 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -129,9 +129,9 @@ static int snd_proto_probe(struct platform_device *pdev) } if (bitclkmaster) { if (codec_np == bitclkmaster) - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; else - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; } else { dai_fmt |= snd_soc_daifmt_parse_clock_provider_as_flag(np, NULL); } diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index ed1f69b57024..915da92e1ec8 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -126,7 +126,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .stream_name = "WM8731 PCM", .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 50c3dc6936f9..1b3a31296c9b 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link tse850_dailink = { .stream_name = "TSE-850-PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFS, + | SND_SOC_DAIFMT_CBP_CFC, SND_SOC_DAILINK_REG(pcm), }; -- 2.20.1

1 1

[PATCH] ASoC: amd: Convert to new style DAI format definitions
by Mark Brown 15 Sep '21

15 Sep '21

Convert the AMD machine drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 20 ++++++++++---------- sound/soc/amd/acp-rt5645.c | 4 ++-- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +++--- 3 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b2065f3fe42c..3bf86c2424ae 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -522,7 +522,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_da7219_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -533,7 +533,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, @@ -543,7 +543,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_max_play_ops, @@ -554,7 +554,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, @@ -565,7 +565,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, @@ -578,7 +578,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_rt5682_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -589,7 +589,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, @@ -599,7 +599,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, @@ -610,7 +610,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, @@ -621,7 +621,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index d6ba94677ac2..58b75cf399fc 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -111,7 +111,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-play", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_init, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware1, codec, platform), @@ -120,7 +120,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-cap", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware2, codec, platform), }, diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e561464f7d60..7af2bfdbf6d7 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -51,7 +51,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) /* set rt5682 dai fmt */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); + | SND_SOC_DAIFMT_CBP_CFP); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set rt5682 dai fmt: %d\n", ret); @@ -302,7 +302,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = acp3x_5682_init, .dpcm_playback = 1, .dpcm_capture = 1, @@ -313,7 +313,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .dpcm_playback = 1, .ops = &acp3x_max_play_ops, .cpus = acp3x_bt, -- 2.20.1

1 1

[PATCH 00/12] ASoC: SOF: Abstract away the 'bar' usage (mostly)
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Hi, The direct use of 'bar' as address regions in SOF can limit portability and makes assumption that all platforms are using memory mapped regions or BARs to communicate with the DSP. The core API should not use 'bar' in it's parameter lists or make assumption that the platform needs to comply with this system integration. Use the region type as an ID in core and translate that in platform dependent code to a way to access registers, mailbox or DSP memory. Regards, Peter --- Peter Ujfalusi (12): ASoC: SOF: Intel: bdw: Set the mailbox offset directly in bdw_probe ASoC: SOF: ipc: Remove snd_sof_dsp_mailbox_init() ASoC: SOF: imx: Do not initialize the snd_sof_dsp_ops.read64 ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types ASoC: SOF: loader: No need to export snd_sof_fw_parse_ext_data() ASoC: SOF: core: Do not use 'bar' as parameter for block_read/write ASoC: SOF: debug: Add generic API and ops for DSP regions ASoC: SOF: imx: Provide debugfs_add_region_item ops for core ASoC: SOF: Intel: Provide debugfs_add_region_item ops for core ASoC: SOF: loader: Use the generic ops for region debugfs handling ASoC: SOF: debug: No need to export the snd_sof_debugfs_io_item() sound/soc/sof/debug.c | 24 ++++-- sound/soc/sof/imx/imx8.c | 17 +++-- sound/soc/sof/imx/imx8m.c | 13 +++- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/bdw.c | 5 +- sound/soc/sof/intel/byt.c | 2 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/pci-tng.c | 1 + sound/soc/sof/intel/tgl.c | 1 + sound/soc/sof/ipc.c | 34 +++------ sound/soc/sof/loader.c | 137 +++++++++++++++------------------- sound/soc/sof/ops.h | 25 +++++-- sound/soc/sof/sof-priv.h | 55 +++++++------- sound/soc/sof/utils.c | 28 +++++-- 15 files changed, 187 insertions(+), 158 deletions(-) -- 2.33.0

2 15

[PATCH] ASoC: mediatek: mt6359: Fix unexpected error in bind/unbind flow
by Trevor Wu 15 Sep '21

15 Sep '21

mt6359-sound is a MFD driver. Because its regmap is retrieved from its parent, it shouldn't be freed in mt6359-sound driver. snd_soc_component_exit_regmap() will do regmap_exit(), this results in unexpected results if sound card unregister flow is invoked when users try to bind/unbind audio codec. Remove the usage of snd_soc_component_exit_regmap(). Instead, set component->regmap = NULL in the component remove function. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/codecs/mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 2d6a4a29b850..f8532aa7e4aa 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2697,7 +2697,7 @@ static int mt6359_codec_probe(struct snd_soc_component *cmpnt) static void mt6359_codec_remove(struct snd_soc_component *cmpnt) { - snd_soc_component_exit_regmap(cmpnt); + cmpnt->regmap = NULL; } static const DECLARE_TLV_DB_SCALE(hp_playback_tlv, -2200, 100, 0); -- 2.18.0

2 1

[PATCH 0/4] Support ALC5682I-VS codec
by Brent Lu 15 Sep '21

15 Sep '21

Support the ALC5682I-VS codec in Intel's rt5682 machine driver with three board configurations. Brent Lu (4): ASoC: Intel: sof_rt5682: support ALC5682I-VS codec ASoC: intel: sof_rt5682: support jsl_rt5682s_rt1015p board ASoC: intel: sof_rt5682: support jsl_rt5682s_rt1015 board ASoC: intel: sof_rt5682: support jsl_rt5682s_mx98360a board sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_rt5682.c | 105 +++++++++++++++--- .../intel/common/soc-acpi-intel-jsl-match.c | 24 ++++ 3 files changed, 112 insertions(+), 18 deletions(-) -- 2.25.1

3 6

[PATCH 0/3] ASoC: SOF: Intel: hda: Cleanups for local function uses
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Hi, Small cleanups regarding to HDA function locations and dropping of definitions of not implemented functions. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda-dsp: Declare locally used functions as static ASoC: SOF: Intel: hda: Remove boot_firmware skl and iccmax_icl declarations ASoC: SOF: Intel: hda: Relocate inline definitions from hda.h to hda.c for sdw sound/soc/sof/intel/hda-dsp.c | 61 +++++++++++++++++------------------ sound/soc/sof/intel/hda.c | 33 ++++++++++++++++++- sound/soc/sof/intel/hda.h | 41 ----------------------- 3 files changed, 62 insertions(+), 73 deletions(-) -- 2.33.0

2 4

[PATCH] ASoC: mediatek: mt8195: make array adda_dai_list static const
by Colin King 15 Sep '21

15 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array adda_dai_list on the stack but instead it static const. Also makes the object code smaller by 33 bytes: Before: text data bss dec hex filename 28271 11640 0 39911 9be7 mt8195/mt8195-dai-adda.o After: text data bss dec hex filename 28142 11736 0 39878 9bc6 mt8195/mt8195-dai-adda.o (gcc version 11.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c index 878dec0b69ed..f04bd1781356 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c @@ -788,9 +788,11 @@ static int init_adda_priv_data(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; struct mtk_dai_adda_priv *adda_priv; - int adda_dai_list[] = { MT8195_AFE_IO_DL_SRC, - MT8195_AFE_IO_UL_SRC1, - MT8195_AFE_IO_UL_SRC2}; + static const int adda_dai_list[] = { + MT8195_AFE_IO_DL_SRC, + MT8195_AFE_IO_UL_SRC1, + MT8195_AFE_IO_UL_SRC2 + }; int i; for (i = 0; i < ARRAY_SIZE(adda_dai_list); i++) { -- 2.32.0

2 1

[PATCH 1/6] ASoC: cs35l41: Fix use of an uninitialised variable
by Charles Keepax 15 Sep '21

15 Sep '21

The loop checking PDN_DONE doesn't check the return value from regmap_read, nor does it initialise val. This means if regmap_read fails val will be checked for the PDN_DONE bit whilst being uninitialised. Fix this up by switching to regmap_read_poll_timeout which tidies up the code and avoids the uninitialised variable. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index dbec54a28a9e3..d2a11cc336833 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -580,7 +580,6 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret = 0; bool pdn; - int i; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -599,19 +598,11 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, CS35L41_GLOBAL_EN_MASK, 0); pdn = false; - for (i = 0; i < 100; i++) { - regmap_read(cs35l41->regmap, - CS35L41_IRQ1_STATUS1, - &val); - if (val & CS35L41_PDN_DONE_MASK) { - pdn = true; - break; - } - usleep_range(1000, 1100); - } - - if (!pdn) - dev_warn(cs35l41->dev, "PDN failed\n"); + ret = regmap_read_poll_timeout(cs35l41->regmap, CS35L41_IRQ1_STATUS1, + val, val & CS35L41_PDN_DONE_MASK, + 1000, 100000); + if (ret) + dev_warn(cs35l41->dev, "PDN failed: %d\n", ret); regmap_write(cs35l41->regmap, CS35L41_IRQ1_STATUS1, CS35L41_PDN_DONE_MASK); -- 2.11.0

2 9

[PATCH 0/2] ASoC: SOF: Remove unused members from struct sof_dev_desc
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Hi, dma_engine, dma_size and resindex_dma_base is unused from sof_dev_desc, drop them. resindex_dma_base is initialized to -1 for Intel platforms, but it is not used. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: intel: Do no initialize resindex_dma_base ASoC: SOF: Drop resindex_dma_base, dma_engine, dma_size from sof_dev_desc include/sound/sof.h | 5 ----- sound/soc/sof/intel/pci-apl.c | 2 -- sound/soc/sof/intel/pci-cnl.c | 3 --- sound/soc/sof/intel/pci-icl.c | 2 -- sound/soc/sof/intel/pci-tgl.c | 5 ----- sound/soc/sof/intel/pci-tng.c | 1 - 6 files changed, 18 deletions(-) -- 2.33.0

2 3

[PATCH -next] ASoC: mediatek: mt8195: Add missing of_node_put()
by Bixuan Cui 15 Sep '21

15 Sep '21

The platform_node is returned by of_parse_phandle() should have of_node_put() before return. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Bixuan Cui <cuibixuan(a)huawei.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index c97ace7387b4..8cd8450409e8 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -1041,8 +1041,10 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) + if (!priv) { + of_node_put(platform_node); return -ENOMEM; + } snd_soc_card_set_drvdata(card, priv); @@ -1050,6 +1052,8 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(platform_node); return ret; } -- 2.17.1

3 3

[PATCH] ASoC: SOF: loader: load_firmware callback is mandatory, treat it like that
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Since the load_firmware callback in snd_sof_dsp_ops is mandatory and it is tested during probe. Move the snd_sof_load_firmware() wrapper to ops.h as inline and drop the check of sof_ops(sdev)->load_firmware Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/loader.c | 10 ---------- sound/soc/sof/ops.h | 8 ++++++++ sound/soc/sof/sof-priv.h | 1 - 3 files changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..5308c32e26fa 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -800,16 +800,6 @@ int snd_sof_load_firmware_memcpy(struct snd_sof_dev *sdev) } EXPORT_SYMBOL(snd_sof_load_firmware_memcpy); -int snd_sof_load_firmware(struct snd_sof_dev *sdev) -{ - dev_dbg(sdev->dev, "loading firmware\n"); - - if (sof_ops(sdev)->load_firmware) - return sof_ops(sdev)->load_firmware(sdev); - return 0; -} -EXPORT_SYMBOL(snd_sof_load_firmware); - int snd_sof_run_firmware(struct snd_sof_dev *sdev) { int ret; diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 4a5d6e497f05..731b8157aaea 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -400,6 +400,14 @@ snd_sof_pcm_platform_trigger(struct snd_sof_dev *sdev, return 0; } +/* Firmware loading */ +static inline int snd_sof_load_firmware(struct snd_sof_dev *sdev) +{ + dev_dbg(sdev->dev, "loading firmware\n"); + + return sof_ops(sdev)->load_firmware(sdev); +} + /* host DSP message data */ static inline void snd_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index fd8423172d8f..1adbcda0b364 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -485,7 +485,6 @@ int snd_sof_create_page_table(struct device *dev, /* * Firmware loading. */ -int snd_sof_load_firmware(struct snd_sof_dev *sdev); int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev); int snd_sof_load_firmware_memcpy(struct snd_sof_dev *sdev); int snd_sof_run_firmware(struct snd_sof_dev *sdev); -- 2.33.0

2 1

[PATCH] ASoC: SOF: Fix DSP oops stack dump output contents
by Peter Ujfalusi 15 Sep '21

15 Sep '21

From: Yong Zhi <yong.zhi(a)intel.com> Fix @buf arg given to hex_dump_to_buffer() and stack address used in dump error output. Fixes: e657c18a01c8 ('ASoC: SOF: Add xtensa support') Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/xtensa/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c index bbb9a2282ed9..f6e3411b33cf 100644 --- a/sound/soc/sof/xtensa/core.c +++ b/sound/soc/sof/xtensa/core.c @@ -122,9 +122,9 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63 */ for (i = 0; i < stack_words; i += 4) { - hex_dump_to_buffer(stack + i * 4, 16, 16, 4, + hex_dump_to_buffer(stack + i, 16, 16, 4, buf, sizeof(buf), false); - dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf); + dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i * 4, buf); } } -- 2.33.0

2 1

[PATCH v2 1/2] ASoC: rt5682s: Add driver for ALC5682I-VS codec
by derek.fang＠realtek.com 15 Sep '21

15 Sep '21

From: Derek Fang <derek.fang(a)realtek.com> This is an initial codec driver for Realtek ALC5682I-VS codec. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- include/sound/rt5682s.h | 48 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682s.c | 3188 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5682s.h | 1455 ++++++++++++++++ 5 files changed, 4698 insertions(+) create mode 100644 include/sound/rt5682s.h create mode 100644 sound/soc/codecs/rt5682s.c create mode 100644 sound/soc/codecs/rt5682s.h diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h new file mode 100644 index 000000000000..accfbc2dcdd2 --- /dev/null +++ b/include/sound/rt5682s.h @@ -0,0 +1,48 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt5682s.h -- Platform data for RT5682I-VS + * + * Copyright 2021 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT5682S_H +#define __LINUX_SND_RT5682S_H + +enum rt5682s_dmic1_data_pin { + RT5682S_DMIC1_DATA_NULL, + RT5682S_DMIC1_DATA_GPIO2, + RT5682S_DMIC1_DATA_GPIO5, +}; + +enum rt5682s_dmic1_clk_pin { + RT5682S_DMIC1_CLK_NULL, + RT5682S_DMIC1_CLK_GPIO1, + RT5682S_DMIC1_CLK_GPIO3, +}; + +enum rt5682s_jd_src { + RT5682S_JD_NULL, + RT5682S_JD1, +}; + +enum rt5682s_dai_clks { + RT5682S_DAI_WCLK_IDX, + RT5682S_DAI_BCLK_IDX, + RT5682S_DAI_NUM_CLKS, +}; + +struct rt5682s_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5682s_dmic1_data_pin dmic1_data_pin; + enum rt5682s_dmic1_clk_pin dmic1_clk_pin; + enum rt5682s_jd_src jd_src; + unsigned int dmic_clk_rate; + unsigned int dmic_delay; + bool dmic_clk_driving_high; + + const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..6b3c2cf153db 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,6 +180,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT5677 imply SND_SOC_RT5682_I2C imply SND_SOC_RT5682_SDW + imply SND_SOC_RT5682S imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW imply SND_SOC_RT711_SDCA_SDW @@ -1249,6 +1250,10 @@ config SND_SOC_RT5682_SDW select SND_SOC_RT5682 select REGMAP_SOUNDWIRE +config SND_SOC_RT5682S + tristate + depends on I2C + config SND_SOC_RT700 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..42d00aa4ee46 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -198,6 +198,7 @@ snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt5682-i2c-objs := rt5682-i2c.o +snd-soc-rt5682s-objs := rt5682s.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o @@ -526,6 +527,7 @@ obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_RT5682_I2C) += snd-soc-rt5682-i2c.o obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o +obj-$(CONFIG_SND_SOC_RT5682S) += snd-soc-rt5682s.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT711_SDCA_SDW) += snd-soc-rt711-sdca.o diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c new file mode 100644 index 000000000000..d878a20527f1 --- /dev/null +++ b/sound/soc/codecs/rt5682s.c @@ -0,0 +1,3188 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt5682s.c -- RT5682I-VS ALSA SoC audio component driver +// +// Copyright 2021 Realtek Semiconductor Corp. +// Author: Derek Fang <derek.fang(a)realtek.com> +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5682s.h> + +#include "rt5682s.h" + +#define DEVICE_ID 0x6749 + +static const struct rt5682s_platform_data i2s_default_platform_data = { + .dmic1_data_pin = RT5682S_DMIC1_DATA_GPIO2, + .dmic1_clk_pin = RT5682S_DMIC1_CLK_GPIO3, + .jd_src = RT5682S_JD1, + .dai_clk_names[RT5682S_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682S_DAI_BCLK_IDX] = "rt5682-dai-bclk", +}; + +const char *rt5682s_supply_names[RT5682S_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", +}; + +static const struct reg_sequence patch_list[] = { + {RT5682S_I2C_CTRL, 0x0007}, + {RT5682S_DIG_IN_CTRL_1, 0x0000}, + {RT5682S_CHOP_DAC_2, 0x2020}, + {RT5682S_VREF_REC_OP_FB_CAP_CTRL_2, 0x0101}, + {RT5682S_VREF_REC_OP_FB_CAP_CTRL_1, 0x80c0}, + {RT5682S_HP_CALIB_CTRL_9, 0x0002}, + {RT5682S_DEPOP_1, 0x0000}, + {RT5682S_HP_CHARGE_PUMP_2, 0x3c15}, + {RT5682S_DAC1_DIG_VOL, 0xfefe}, + {RT5682S_SAR_IL_CMD_2, 0xac00}, + {RT5682S_SAR_IL_CMD_3, 0x024c}, + {RT5682S_CBJ_CTRL_6, 0x0804}, +}; + +static void rt5682s_apply_patch_list(struct rt5682s_priv *rt5682s, + struct device *dev) +{ + int ret; + + ret = regmap_multi_reg_write(rt5682s->regmap, patch_list, ARRAY_SIZE(patch_list)); + if (ret) + dev_warn(dev, "Failed to apply regmap patch: %d\n", ret); +} + +const struct reg_default rt5682s_reg[] = { + {0x0000, 0x0001}, + {0x0002, 0x8080}, + {0x0003, 0x0001}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x8007}, + {0x000b, 0x0000}, + {0x000f, 0x4000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x0000}, + {0x0013, 0x1200}, + {0x0014, 0x200a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0017, 0x05a4}, + {0x0019, 0xffff}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x08c0}, + {0x0044, 0x1818}, + {0x004b, 0x00c0}, + {0x004c, 0x0000}, + {0x004d, 0x0000}, + {0x0061, 0x00c0}, + {0x0062, 0x008a}, + {0x0063, 0x0800}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x000c}, + {0x0068, 0x0000}, + {0x0069, 0x0000}, + {0x006a, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0810}, + {0x006f, 0xe4de}, + {0x0070, 0x3320}, + {0x0071, 0x0000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x007f, 0x0000}, + {0x0080, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008e, 0x0060}, + {0x008f, 0x4da1}, + {0x0091, 0x1c15}, + {0x0092, 0x0425}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0095, 0x008f}, + {0x0096, 0x0000}, + {0x0097, 0x0000}, + {0x0098, 0x0000}, + {0x0099, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x0000}, + {0x009f, 0x0009}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0000}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6749}, + {0x0100, 0xa000}, + {0x010b, 0x0066}, + {0x010c, 0x6666}, + {0x010d, 0x2202}, + {0x010e, 0x6666}, + {0x010f, 0xa800}, + {0x0110, 0x0006}, + {0x0111, 0x0460}, + {0x0112, 0x2000}, + {0x0113, 0x0200}, + {0x0117, 0x8000}, + {0x0118, 0x0303}, + {0x0125, 0x0020}, + {0x0132, 0x5026}, + {0x0136, 0x8000}, + {0x0139, 0x0005}, + {0x013a, 0x3030}, + {0x013b, 0xa000}, + {0x013c, 0x4110}, + {0x013f, 0x0000}, + {0x0145, 0x0022}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0156, 0x0022}, + {0x0157, 0x0303}, + {0x0158, 0x2222}, + {0x0159, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0001}, + {0x0190, 0x4131}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01ac, 0x0000}, + {0x01ad, 0x0000}, + {0x01ae, 0x0000}, + {0x01af, 0x2000}, + {0x01b0, 0x0000}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0017}, + {0x01b4, 0x004b}, + {0x01b5, 0x0000}, + {0x01b6, 0x03e8}, + {0x01b7, 0x0000}, + {0x01b8, 0x0000}, + {0x01b9, 0x0400}, + {0x01ba, 0xb5b6}, + {0x01bb, 0x9124}, + {0x01bc, 0x4924}, + {0x01bd, 0x0009}, + {0x01be, 0x0018}, + {0x01bf, 0x002a}, + {0x01c0, 0x004c}, + {0x01c1, 0x0097}, + {0x01c2, 0x01c3}, + {0x01c3, 0x03e9}, + {0x01c4, 0x1389}, + {0x01c5, 0xc351}, + {0x01c6, 0x02a0}, + {0x01c7, 0x0b0f}, + {0x01c8, 0x402f}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0000}, + {0x01d8, 0x0162}, + {0x01d9, 0x0007}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0020}, + {0x01e0, 0x04c1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e5, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa004}, + {0x0212, 0x0365}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x021d, 0x024c}, + {0x02fa, 0x0000}, + {0x02fb, 0x0000}, + {0x02fc, 0x0000}, + {0x03fe, 0x0000}, + {0x03ff, 0x0000}, + {0x0500, 0x0000}, + {0x0600, 0x0000}, + {0x0610, 0x6666}, + {0x0611, 0xa9aa}, + {0x0620, 0x6666}, + {0x0621, 0xa9aa}, + {0x0630, 0x6666}, + {0x0631, 0xa9aa}, + {0x0640, 0x6666}, + {0x0641, 0xa9aa}, + {0x07fa, 0x0000}, + {0x08fa, 0x0000}, + {0x08fb, 0x0000}, + {0x0d00, 0x0000}, + {0x1100, 0x0000}, + {0x1101, 0x0000}, + {0x1102, 0x0000}, + {0x1103, 0x0000}, + {0x1104, 0x0000}, + {0x1105, 0x0000}, + {0x1106, 0x0000}, + {0x1107, 0x0000}, + {0x1108, 0x0000}, + {0x1109, 0x0000}, + {0x110a, 0x0000}, + {0x110b, 0x0000}, + {0x110c, 0x0000}, + {0x1111, 0x0000}, + {0x1112, 0x0000}, + {0x1113, 0x0000}, + {0x1114, 0x0000}, + {0x1115, 0x0000}, + {0x1116, 0x0000}, + {0x1117, 0x0000}, + {0x1118, 0x0000}, + {0x1119, 0x0000}, + {0x111a, 0x0000}, + {0x111b, 0x0000}, + {0x111c, 0x0000}, + {0x1401, 0x0404}, + {0x1402, 0x0007}, + {0x1403, 0x0365}, + {0x1404, 0x0210}, + {0x1405, 0x0365}, + {0x1406, 0x0210}, + {0x1407, 0x0000}, + {0x1408, 0x0000}, + {0x1409, 0x0000}, + {0x140a, 0x0000}, + {0x140b, 0x0000}, + {0x140c, 0x0000}, + {0x140d, 0x0000}, + {0x140e, 0x0000}, + {0x140f, 0x0000}, + {0x1410, 0x0000}, + {0x1411, 0x0000}, + {0x1801, 0x0004}, + {0x1802, 0x0000}, + {0x1803, 0x0000}, + {0x1804, 0x0000}, + {0x1805, 0x00ff}, + {0x2c00, 0x0000}, + {0x3400, 0x0200}, + {0x3404, 0x0000}, + {0x3405, 0x0000}, + {0x3406, 0x0000}, + {0x3407, 0x0000}, + {0x3408, 0x0000}, + {0x3409, 0x0000}, + {0x340a, 0x0000}, + {0x340b, 0x0000}, + {0x340c, 0x0000}, + {0x340d, 0x0000}, + {0x340e, 0x0000}, + {0x340f, 0x0000}, + {0x3410, 0x0000}, + {0x3411, 0x0000}, + {0x3412, 0x0000}, + {0x3413, 0x0000}, + {0x3414, 0x0000}, + {0x3415, 0x0000}, + {0x3424, 0x0000}, + {0x3425, 0x0000}, + {0x3426, 0x0000}, + {0x3427, 0x0000}, + {0x3428, 0x0000}, + {0x3429, 0x0000}, + {0x342a, 0x0000}, + {0x342b, 0x0000}, + {0x342c, 0x0000}, + {0x342d, 0x0000}, + {0x342e, 0x0000}, + {0x342f, 0x0000}, + {0x3430, 0x0000}, + {0x3431, 0x0000}, + {0x3432, 0x0000}, + {0x3433, 0x0000}, + {0x3434, 0x0000}, + {0x3435, 0x0000}, + {0x3440, 0x6319}, + {0x3441, 0x3771}, + {0x3500, 0x0002}, + {0x3501, 0x5728}, + {0x3b00, 0x3010}, + {0x3b01, 0x3300}, + {0x3b02, 0x2200}, + {0x3b03, 0x0100}, +}; + +static bool rt5682s_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682S_RESET: + case RT5682S_CBJ_CTRL_2: + case RT5682S_I2S1_F_DIV_CTRL_2: + case RT5682S_I2S2_F_DIV_CTRL_2: + case RT5682S_INT_ST_1: + case RT5682S_GPIO_ST: + case RT5682S_IL_CMD_1: + case RT5682S_4BTN_IL_CMD_1: + case RT5682S_AJD1_CTRL: + case RT5682S_VERSION_ID...RT5682S_DEVICE_ID: + case RT5682S_STO_NG2_CTRL_1: + case RT5682S_STO_NG2_CTRL_5...RT5682S_STO_NG2_CTRL_7: + case RT5682S_STO1_DAC_SIL_DET: + case RT5682S_HP_IMP_SENS_CTRL_1...RT5682S_HP_IMP_SENS_CTRL_4: + case RT5682S_HP_IMP_SENS_CTRL_13: + case RT5682S_HP_IMP_SENS_CTRL_14: + case RT5682S_HP_IMP_SENS_CTRL_43...RT5682S_HP_IMP_SENS_CTRL_46: + case RT5682S_HP_CALIB_CTRL_1: + case RT5682S_HP_CALIB_CTRL_10: + case RT5682S_HP_CALIB_ST_1...RT5682S_HP_CALIB_ST_11: + case RT5682S_SAR_IL_CMD_2...RT5682S_SAR_IL_CMD_5: + case RT5682S_SAR_IL_CMD_10: + case RT5682S_SAR_IL_CMD_11: + case RT5682S_VERSION_ID_HIDE: + case RT5682S_VERSION_ID_CUS: + case RT5682S_I2C_TRANS_CTRL: + case RT5682S_DMIC_FLOAT_DET: + case RT5682S_HA_CMP_OP_1: + case RT5682S_NEW_CBJ_DET_CTL_10...RT5682S_NEW_CBJ_DET_CTL_16: + case RT5682S_CLK_SW_TEST_1: + case RT5682S_CLK_SW_TEST_2: + case RT5682S_EFUSE_READ_1...RT5682S_EFUSE_READ_18: + case RT5682S_PILOT_DIG_CTL_1: + return true; + default: + return false; + } +} + +static bool rt5682s_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682S_RESET: + case RT5682S_VERSION_ID: + case RT5682S_VENDOR_ID: + case RT5682S_DEVICE_ID: + case RT5682S_HP_CTRL_1: + case RT5682S_HP_CTRL_2: + case RT5682S_HPL_GAIN: + case RT5682S_HPR_GAIN: + case RT5682S_I2C_CTRL: + case RT5682S_CBJ_BST_CTRL: + case RT5682S_CBJ_DET_CTRL: + case RT5682S_CBJ_CTRL_1...RT5682S_CBJ_CTRL_8: + case RT5682S_DAC1_DIG_VOL: + case RT5682S_STO1_ADC_DIG_VOL: + case RT5682S_STO1_ADC_BOOST: + case RT5682S_HP_IMP_GAIN_1: + case RT5682S_HP_IMP_GAIN_2: + case RT5682S_SIDETONE_CTRL: + case RT5682S_STO1_ADC_MIXER: + case RT5682S_AD_DA_MIXER: + case RT5682S_STO1_DAC_MIXER: + case RT5682S_A_DAC1_MUX: + case RT5682S_DIG_INF2_DATA: + case RT5682S_REC_MIXER: + case RT5682S_CAL_REC: + case RT5682S_HP_ANA_OST_CTRL_1...RT5682S_HP_ANA_OST_CTRL_3: + case RT5682S_PWR_DIG_1...RT5682S_PWR_MIXER: + case RT5682S_MB_CTRL: + case RT5682S_CLK_GATE_TCON_1...RT5682S_CLK_GATE_TCON_3: + case RT5682S_CLK_DET...RT5682S_LPF_AD_DMIC: + case RT5682S_I2S1_SDP: + case RT5682S_I2S2_SDP: + case RT5682S_ADDA_CLK_1: + case RT5682S_ADDA_CLK_2: + case RT5682S_I2S1_F_DIV_CTRL_1: + case RT5682S_I2S1_F_DIV_CTRL_2: + case RT5682S_TDM_CTRL: + case RT5682S_TDM_ADDA_CTRL_1: + case RT5682S_TDM_ADDA_CTRL_2: + case RT5682S_DATA_SEL_CTRL_1: + case RT5682S_TDM_TCON_CTRL_1: + case RT5682S_TDM_TCON_CTRL_2: + case RT5682S_GLB_CLK: + case RT5682S_PLL_TRACK_1...RT5682S_PLL_TRACK_6: + case RT5682S_PLL_TRACK_11: + case RT5682S_DEPOP_1: + case RT5682S_HP_CHARGE_PUMP_1: + case RT5682S_HP_CHARGE_PUMP_2: + case RT5682S_HP_CHARGE_PUMP_3: + case RT5682S_MICBIAS_1...RT5682S_MICBIAS_3: + case RT5682S_PLL_TRACK_12...RT5682S_PLL_CTRL_7: + case RT5682S_RC_CLK_CTRL: + case RT5682S_I2S2_M_CLK_CTRL_1: + case RT5682S_I2S2_F_DIV_CTRL_1: + case RT5682S_I2S2_F_DIV_CTRL_2: + case RT5682S_IRQ_CTRL_1...RT5682S_IRQ_CTRL_4: + case RT5682S_INT_ST_1: + case RT5682S_GPIO_CTRL_1: + case RT5682S_GPIO_CTRL_2: + case RT5682S_GPIO_ST: + case RT5682S_HP_AMP_DET_CTRL_1: + case RT5682S_MID_HP_AMP_DET: + case RT5682S_LOW_HP_AMP_DET: + case RT5682S_DELAY_BUF_CTRL: + case RT5682S_SV_ZCD_1: + case RT5682S_SV_ZCD_2: + case RT5682S_IL_CMD_1...RT5682S_IL_CMD_6: + case RT5682S_4BTN_IL_CMD_1...RT5682S_4BTN_IL_CMD_7: + case RT5682S_ADC_STO1_HP_CTRL_1: + case RT5682S_ADC_STO1_HP_CTRL_2: + case RT5682S_AJD1_CTRL: + case RT5682S_JD_CTRL_1: + case RT5682S_DUMMY_1...RT5682S_DUMMY_3: + case RT5682S_DAC_ADC_DIG_VOL1: + case RT5682S_BIAS_CUR_CTRL_2...RT5682S_BIAS_CUR_CTRL_10: + case RT5682S_VREF_REC_OP_FB_CAP_CTRL_1: + case RT5682S_VREF_REC_OP_FB_CAP_CTRL_2: + case RT5682S_CHARGE_PUMP_1: + case RT5682S_DIG_IN_CTRL_1: + case RT5682S_PAD_DRIVING_CTRL: + case RT5682S_CHOP_DAC_1: + case RT5682S_CHOP_DAC_2: + case RT5682S_CHOP_ADC: + case RT5682S_CALIB_ADC_CTRL: + case RT5682S_VOL_TEST: + case RT5682S_SPKVDD_DET_ST: + case RT5682S_TEST_MODE_CTRL_1...RT5682S_TEST_MODE_CTRL_4: + case RT5682S_PLL_INTERNAL_1...RT5682S_PLL_INTERNAL_4: + case RT5682S_STO_NG2_CTRL_1...RT5682S_STO_NG2_CTRL_10: + case RT5682S_STO1_DAC_SIL_DET: + case RT5682S_SIL_PSV_CTRL1: + case RT5682S_SIL_PSV_CTRL2: + case RT5682S_SIL_PSV_CTRL3: + case RT5682S_SIL_PSV_CTRL4: + case RT5682S_SIL_PSV_CTRL5: + case RT5682S_HP_IMP_SENS_CTRL_1...RT5682S_HP_IMP_SENS_CTRL_46: + case RT5682S_HP_LOGIC_CTRL_1...RT5682S_HP_LOGIC_CTRL_3: + case RT5682S_HP_CALIB_CTRL_1...RT5682S_HP_CALIB_CTRL_11: + case RT5682S_HP_CALIB_ST_1...RT5682S_HP_CALIB_ST_11: + case RT5682S_SAR_IL_CMD_1...RT5682S_SAR_IL_CMD_14: + case RT5682S_DUMMY_4...RT5682S_DUMMY_6: + case RT5682S_VERSION_ID_HIDE: + case RT5682S_VERSION_ID_CUS: + case RT5682S_SCAN_CTL: + case RT5682S_HP_AMP_DET: + case RT5682S_BIAS_CUR_CTRL_11: + case RT5682S_BIAS_CUR_CTRL_12: + case RT5682S_BIAS_CUR_CTRL_13: + case RT5682S_BIAS_CUR_CTRL_14: + case RT5682S_BIAS_CUR_CTRL_15: + case RT5682S_BIAS_CUR_CTRL_16: + case RT5682S_BIAS_CUR_CTRL_17: + case RT5682S_BIAS_CUR_CTRL_18: + case RT5682S_I2C_TRANS_CTRL: + case RT5682S_DUMMY_7: + case RT5682S_DUMMY_8: + case RT5682S_DMIC_FLOAT_DET: + case RT5682S_HA_CMP_OP_1...RT5682S_HA_CMP_OP_13: + case RT5682S_HA_CMP_OP_14...RT5682S_HA_CMP_OP_25: + case RT5682S_NEW_CBJ_DET_CTL_1...RT5682S_NEW_CBJ_DET_CTL_16: + case RT5682S_DA_FILTER_1...RT5682S_DA_FILTER_5: + case RT5682S_CLK_SW_TEST_1: + case RT5682S_CLK_SW_TEST_2: + case RT5682S_CLK_SW_TEST_3...RT5682S_CLK_SW_TEST_14: + case RT5682S_EFUSE_MANU_WRITE_1...RT5682S_EFUSE_MANU_WRITE_6: + case RT5682S_EFUSE_READ_1...RT5682S_EFUSE_READ_18: + case RT5682S_EFUSE_TIMING_CTL_1: + case RT5682S_EFUSE_TIMING_CTL_2: + case RT5682S_PILOT_DIG_CTL_1: + case RT5682S_PILOT_DIG_CTL_2: + case RT5682S_HP_AMP_DET_CTL_1...RT5682S_HP_AMP_DET_CTL_4: + return true; + default: + return false; + } +} + +static void rt5682s_reset(struct rt5682s_priv *rt5682s) +{ + regmap_write(rt5682s->regmap, RT5682S_RESET, 0); +} + +static int rt5682s_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read(component, RT5682S_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682S_4BTN_IL_CMD_1, val); + dev_dbg(component->dev, "%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_2, + RT5682S_SAR_ADC_PSV_MASK, RT5682S_SAR_ADC_PSV_ENTRY); + + return btn_type; +} + +enum { + SAR_PWR_OFF, + SAR_PWR_NORMAL, + SAR_PWR_SAVING, +}; + +static void rt5682s_sar_power_mode(struct snd_soc_component *component, + int mode, int jd_step) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + mutex_lock(&rt5682s->sar_mutex); + + switch (mode) { + case SAR_PWR_SAVING: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_REG | RT5682S_CTRL_MB2_REG); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK, RT5682S_SAR_BUTDET_EN); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_2, + RT5682S_SAR_ADC_PSV_MASK, RT5682S_SAR_ADC_PSV_ENTRY); + break; + case SAR_PWR_NORMAL: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_FSM | RT5682S_CTRL_MB2_FSM); + if (!jd_step) { + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_SEL_MB1_2_AUTO); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK, + RT5682S_SAR_BUTDET_EN | RT5682S_SAR_BUTDET_POW_NORM); + } + break; + case SAR_PWR_OFF: + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + break; + default: + dev_err(component->dev, "Invalid SAR Power mode: %d\n", mode); + break; + } + + mutex_unlock(&rt5682s->sar_mutex); +} + +static void rt5682s_enable_push_button_irq(struct snd_soc_component *component) +{ + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682S_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK | RT5682S_4BTN_IL_RST_MASK, + RT5682S_4BTN_IL_EN | RT5682S_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682S_IRQ_CTRL_3, + RT5682S_IL_IRQ_MASK, RT5682S_IL_IRQ_EN); +} + +static void rt5682s_disable_push_button_irq(struct snd_soc_component *component) +{ + snd_soc_component_update_bits(component, RT5682S_IRQ_CTRL_3, + RT5682S_IL_IRQ_MASK, RT5682S_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); +} + +/** + * rt5682s_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_insert) +{ + unsigned int val, count; + int jack_type = 0; + + if (jack_insert) { + rt5682s_disable_push_button_irq(component); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF1 | RT5682S_PWR_VREF2 | RT5682S_PWR_MB, + RT5682S_PWR_VREF1 | RT5682S_PWR_VREF2 | RT5682S_PWR_MB); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_3, + RT5682S_PWR_CBJ, RT5682S_PWR_CBJ); + snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x0365); + snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, + RT5682S_OSW_L_DIS | RT5682S_OSW_R_DIS); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 1); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); + usleep_range(45000, 50000); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_HIGH); + + count = 0; + do { + usleep_range(10000, 15000); + val = snd_soc_component_read(component, RT5682S_CBJ_CTRL_2) + & RT5682S_JACK_TYPE_MASK; + count++; + } while (val == 0 && count < 50); + + pr_debug("%s, val=%d, count=%d\n", __func__, val, count); + + switch (val) { + case 0x1: + case 0x2: + jack_type = SND_JACK_HEADSET; + snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x024c); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_EN); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_MASK, val << RT5682S_SAR_SEL_MB1_2_SFT); + rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 1); + rt5682s_enable_push_button_irq(component); + break; + default: + jack_type = SND_JACK_HEADPHONE; + break; + } + snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, + RT5682S_OSW_L_EN | RT5682S_OSW_R_EN); + } else { + rt5682s_sar_power_mode(component, SAR_PWR_OFF, 1); + rt5682s_disable_push_button_irq(component); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); + + if (!snd_soc_dapm_get_pin_status(&component->dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(&component->dapm, "Vref2")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_VREF2, 0); + + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_3, + RT5682S_PWR_CBJ, 0); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_DIS); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", jack_type); + + return jack_type; +} + +static void rt5682s_jack_detect_handler(struct work_struct *work) +{ + struct rt5682s_priv *rt5682s = + container_of(work, struct rt5682s_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682s->component) + usleep_range(10000, 15000); + + while (!rt5682s->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682s->calibrate_mutex); + + val = snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) + & RT5682S_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682s->jack_type == 0) { + /* jack was out, report jack type */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 1); + rt5682s->irq_work_delay_time = 0; + } else if ((rt5682s->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + rt5682s->jack_type = SND_JACK_HEADSET; + btn_type = rt5682s_button_detect(rt5682s->component); + /** + * rt5682s can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682s->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682s->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682s->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682s->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5682s->component->dev, + "Unexpected button code 0x%04x\n", btn_type); + break; + } + } + } else { + /* jack out */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); + rt5682s->irq_work_delay_time = 50; + } + + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682s->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682s->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + mutex_unlock(&rt5682s->calibrate_mutex); +} + +static void rt5682s_jd_check_handler(struct work_struct *work) +{ + struct rt5682s_priv *rt5682s = + container_of(work, struct rt5682s_priv, jd_check_work.work); + + if (snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) + & RT5682S_JDH_RS_MASK) { + /* jack out */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); + + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682s->jd_check_work, 500); + } +} + +static irqreturn_t rt5682s_irq(int irq, void *data) +{ + struct rt5682s_priv *rt5682s = data; + + mod_delayed_work(system_power_efficient_wq, &rt5682s->jack_detect_work, + msecs_to_jiffies(rt5682s->irq_work_delay_time)); + + return IRQ_HANDLED; +} + +static int rt5682s_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int btndet_delay = 16; + + rt5682s->hs_jack = hs_jack; + + if (!hs_jack) { + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK, RT5682S_JD1_DIS); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_JDH, 0); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + + return 0; + } + + switch (rt5682s->pdata.jd_src) { + case RT5682S_JD1: + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_5, + RT5682S_JD_FAST_OFF_SRC_MASK, RT5682S_JD_FAST_OFF_SRC_JDH); + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_2, + RT5682S_EXT_JD_SRC, RT5682S_EXT_JD_SRC_MANUAL); + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_1, + RT5682S_EMB_JD_MASK | RT5682S_DET_TYPE | + RT5682S_POL_FAST_OFF_MASK | RT5682S_MIC_CAP_MASK, + RT5682S_EMB_JD_EN | RT5682S_DET_TYPE | + RT5682S_POL_FAST_OFF_HIGH | RT5682S_MIC_CAP_HS); + regmap_update_bits(rt5682s->regmap, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_POW_MASK, RT5682S_SAR_POW_EN); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP1_PIN_MASK, RT5682S_GP1_PIN_IRQ); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BGLDO, RT5682S_PWR_BGLDO); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_2, + RT5682S_PWR_JD_MASK, RT5682S_PWR_JD_ENABLE); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_IRQ | RT5682S_POW_JDH, RT5682S_POW_IRQ | RT5682S_POW_JDH); + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK | RT5682S_JD1_POL_MASK, + RT5682S_JD1_EN | RT5682S_JD1_POL_NOR); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_4, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_5, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_6, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_7, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682S_JD_NULL: + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK, RT5682S_JD1_DIS); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_JDH, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); + +static const struct snd_kcontrol_new rt5682s_snd_controls[] = { + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, + RT5682S_L_VOL_SFT + 2, RT5682S_R_VOL_SFT + 2, 63, 0, dac_vol_tlv), + + /* CBJ Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682S_REC_MIXER, + RT5682S_BST_CBJ_SFT, 35, 0, cbj_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE_SFT, RT5682S_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_VOL_SFT + 1, RT5682S_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682S_STO1_ADC_BOOST, + RT5682S_STO1_ADC_L_BST_SFT, RT5682S_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), +}; + +/** + * rt5682s_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682S can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + switch (clk_src) { + case RT5682S_CLK_SEL_SYS: + case RT5682S_CLK_SEL_I2S1_ASRC: + case RT5682S_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682S_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_2, + RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682S_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_3, + RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682s_sel_asrc_clk_src); + +static int rt5682s_div_sel(struct rt5682s_priv *rt5682s, + int target, const int div[], int size) +{ + int i; + + if (rt5682s->sysclk < target) { + dev_err(rt5682s->component->dev, + "sysclk rate %d is too low\n", rt5682s->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + dev_dbg(rt5682s->component->dev, "div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682s->sysclk) + return i; + if (target * div[i + 1] > rt5682s->sysclk) { + dev_dbg(rt5682s->component->dev, + "can't find div for sysclk %d\n", rt5682s->sysclk); + return i; + } + } + + if (target * div[i] < rt5682s->sysclk) + dev_err(rt5682s->component->dev, + "sysclk rate %d is too high\n", rt5682s->sysclk); + + return size - 1; +} + +static int get_clk_info(int sclk, int rate) +{ + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int idx, dmic_clk_rate = 3072000; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + if (rt5682s->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682s->pdata.dmic_clk_rate; + + idx = rt5682s_div_sel(rt5682s, dmic_clk_rate, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_CLK_MASK, idx << RT5682S_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int ref, val, reg, idx; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read(component, RT5682S_GPIO_CTRL_1) + & RT5682S_GP4_PIN_MASK; + + if (w->shift == RT5682S_PWR_ADC_S1F_BIT && val == RT5682S_GP4_PIN_ADCDAT2) + ref = 256 * rt5682s->lrck[RT5682S_AIF2]; + else + ref = 256 * rt5682s->lrck[RT5682S_AIF1]; + + idx = rt5682s_div_sel(rt5682s, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682S_PWR_ADC_S1F_BIT) + reg = RT5682S_PLL_TRACK_3; + else + reg = RT5682S_PLL_TRACK_2; + + snd_soc_component_update_bits(component, reg, + RT5682S_FILTER_CLK_DIV_MASK, idx << RT5682S_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682s->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_1, + RT5682S_ADC_OSR_MASK | RT5682S_DAC_OSR_MASK, + (idx << RT5682S_ADC_OSR_SFT) | (idx << RT5682S_DAC_OSR_SFT)); + + return 0; +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int delay = 50, val; + + if (rt5682s->pdata.dmic_delay) + delay = rt5682s->pdata.dmic_delay; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = (snd_soc_component_read(component, RT5682S_GLB_CLK) + & RT5682S_SCLK_SRC_MASK) >> RT5682S_SCLK_SRC_SFT; + if (val == RT5682S_CLK_SRC_PLL1 || val == RT5682S_CLK_SRC_PLL2) + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_MB, + RT5682S_PWR_VREF2 | RT5682S_PWR_MB); + + /*Add delay to avoid pop noise*/ + msleep(delay); + break; + + case SND_SOC_DAPM_POST_PMD: + if (!rt5682s->jack_type) { + if (!snd_soc_dapm_get_pin_status(w->dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(w->dapm, "Vref2")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_VREF2, 0); + } + break; + } + + return 0; +} + +static int set_i2s_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int pre_div, id; + unsigned int reg, mask, sft; + + if (event != SND_SOC_DAPM_PRE_PMU) + return 0; + + if (w->shift == RT5682S_PWR_I2S2_BIT) { + id = RT5682S_AIF2; + reg = RT5682S_I2S2_M_CLK_CTRL_1; + mask = RT5682S_I2S2_M_D_MASK; + sft = RT5682S_I2S2_M_D_SFT; + } else { + id = RT5682S_AIF1; + reg = RT5682S_ADDA_CLK_1; + mask = RT5682S_I2S_M_D_MASK; + sft = RT5682S_I2S_M_D_SFT; + } + + if (!rt5682s->master[id]) + return 0; + + pre_div = get_clk_info(rt5682s->sysclk, rt5682s->lrck[id]); + if (pre_div < 0) { + dev_err(component->dev, "get pre_div failed\n"); + return -EINVAL; + } + + dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d master\n", + rt5682s->lrck[id], pre_div, id); + snd_soc_component_update_bits(component, reg, mask, pre_div << sft); + + return 0; +} + +static int is_sys_clk_from_plla(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if ((rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL1) || + (rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL2 && rt5682s->pll_comb == USE_PLLAB)) + return 1; + + return 0; +} + +static int is_sys_clk_from_pllb(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if (rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL2) + return 1; + + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, sft, val; + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682S_ADC_STO1_ASRC_SFT: + reg = RT5682S_PLL_TRACK_3; + sft = RT5682S_FILTER_CLK_SEL_SFT; + break; + case RT5682S_DAC_STO1_ASRC_SFT: + reg = RT5682S_PLL_TRACK_2; + sft = RT5682S_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read(component, reg) >> sft) & 0xf; + switch (val) { + case RT5682S_CLK_SEL_I2S1_ASRC: + case RT5682S_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } +} + +static int is_headset_type(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if ((rt5682s->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) + return 1; + + return 0; +} + +static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN); + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(30000, 35000); + snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_11, 0x6666); + snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_12, 0xa82a); + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, + RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | + RT5682S_HPO_SEL_IP_EN_SW, RT5682S_HPO_L_PATH_EN | + RT5682S_HPO_R_PATH_EN | RT5682S_HPO_IP_EN_GATING); + snd_soc_component_write(component, RT5682S_HP_AMP_DET_CTL_1, 0x3050); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, + RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | + RT5682S_HPO_SEL_IP_EN_SW, 0); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, 0); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, 0); + break; + } + + return 0; +} + +static int sar_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 0); + break; + case SND_SOC_DAPM_POST_PMD: + rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 0); + break; + } + + return 0; +} + +/* Interface data select */ +static const char * const rt5682s_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_if2_adc_enum, RT5682S_DIG_INF2_DATA, + RT5682S_IF2_ADC_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_01_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC1_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_23_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC2_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_45_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC3_SEL_SFT, rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_67_adc_enum, RT5682S_TDM_ADDA_CTRL_1, + RT5682S_IF1_ADC4_SEL_SFT, rt5682s_data_select); + +static const struct snd_kcontrol_new rt5682s_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682s_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682s_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682s_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682s_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682s_if1_67_adc_enum); + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682s_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682s_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682S_REC_MIXER, + RT5682S_M_CBJ_RM1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_rec1_r_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682S_REC_MIXER, + RT5682S_M_CBJ_RM1_R_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682s_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc1l_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC1L_SRC_SFT, rt5682s_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682s_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc1r_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC1R_SRC_SFT, rt5682s_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682s_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682s_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adcl_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADCL_SRC_SFT, rt5682s_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682s_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adcr_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADCR_SRC_SFT, rt5682s_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682s_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682s_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc2l_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC2L_SRC_SFT, rt5682s_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682s_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_sto1_adc2r_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC2R_SRC_SFT, rt5682s_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682s_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682s_if1_adc_slot_values[] = { + 0, 2, 4, 6, +}; + +static const char * const rt5682s_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_if1_adc_slot_enum, + RT5682S_TDM_CTRL, RT5682S_TDM_ADC_LCA_SFT, RT5682S_TDM_ADC_LCA_MASK, + rt5682s_if1_adc_slot_src, rt5682s_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682s_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682s_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682s_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_alg_dac_l1_enum, RT5682S_A_DAC1_MUX, + RT5682S_A_DACL1_SFT, rt5682s_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682s_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682s_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682s_alg_dac_r1_enum, RT5682S_A_DAC1_MUX, + RT5682S_A_DACR1_SFT, rt5682s_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682s_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682s_alg_dac_r1_enum); + +static const unsigned int rt5682s_adcdat_pin_values[] = { + 1, 3, +}; + +static const char * const rt5682s_adcdat_pin_select[] = { + "ADCDAT1", "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, + RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, RT5682S_GP4_PIN_MASK, + rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682s_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682s_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO MB1", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO MB2", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_MB2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PLL Powers */ + SND_SOC_DAPM_SUPPLY_S("PLLA_LDO", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_LDO", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_PLLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLA_BIAS", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BIAS_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_BIAS", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BIAS_PLLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLA", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_PLLB_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("PLLA_RST", 1, RT5682S_PWR_ANLG_3, + RT5682S_RSTB_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_RST", 1, RT5682S_PWR_ANLG_3, + RT5682S_RSTB_PLLB_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682S_DMIC_CTRL_1, RT5682S_DMIC_1_EN_SFT, 0, + set_dmic_power, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682s_rec1_l_mix, + ARRAY_SIZE(rt5682s_rec1_l_mix)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5682s_rec1_r_mix, + ARRAY_SIZE(rt5682s_rec1_r_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682S_CAL_REC, + RT5682S_PWR_RM1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5682S_CAL_REC, + RT5682S_PWR_RM1_R_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682S_PWR_DIG_1, + RT5682S_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682S_PWR_DIG_1, + RT5682S_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682S_CHOP_ADC, + RT5682S_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682S_PWR_DIG_2, + RT5682S_PWR_ADC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE_SFT, 1, rt5682s_sto1_adc_l_mix, + ARRAY_SIZE(rt5682s_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_R_MUTE_SFT, 1, rt5682s_sto1_adc_r_mix, + ARRAY_SIZE(rt5682s_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682S_PWR_DIG_1, RT5682S_PWR_I2S1_BIT, + 0, set_i2s_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682S_PWR_DIG_1, RT5682S_PWR_I2S2_BIT, + 0, set_i2s_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, &rt5682s_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, RT5682S_I2S1_SDP, + RT5682S_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, RT5682S_I2S2_SDP, + RT5682S_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_dac_l_mix, ARRAY_SIZE(rt5682s_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682s_dac_r_mix, ARRAY_SIZE(rt5682s_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, &rt5682s_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, &rt5682s_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682S_PWR_DIG_2, + RT5682S_PWR_DAC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_dac_l_mix, ARRAY_SIZE(rt5682s_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_dac_r_mix, ARRAY_SIZE(rt5682s_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682S_PWR_DIG_1, RT5682S_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682S_PWR_DIG_1, RT5682S_PWR_DAC_R1_BIT, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682s_hp_amp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, + RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682S_CLK_DET, + RT5682S_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MCLK0 DET PWR", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MCLK0_WD_BIT, 0, NULL, 0), + + /* SAR */ + SND_SOC_DAPM_SUPPLY("SAR", SND_SOC_NOPM, 0, 0, sar_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLLA", is_sys_clk_from_plla}, + {"ADC Stereo1 Filter", NULL, "PLLB", is_sys_clk_from_pllb}, + {"DAC Stereo1 Filter", NULL, "PLLA", is_sys_clk_from_plla}, + {"DAC Stereo1 Filter", NULL, "PLLB", is_sys_clk_from_pllb}, + {"PLLA", NULL, "PLLA_LDO"}, + {"PLLA", NULL, "PLLA_BIAS"}, + {"PLLA", NULL, "PLLA_RST"}, + {"PLLB", NULL, "PLLB_LDO"}, + {"PLLB", NULL, "PLLB_BIAS"}, + {"PLLB", NULL, "PLLB_RST"}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + + {"CLKDET SYS", NULL, "MCLK0 DET PWR"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "SAR", is_headset_type}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + {"RECMIX1R", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1R", NULL, "RECMIX1R Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + {"ADC1 R", NULL, "RECMIX1R"}, + {"ADC1 R", NULL, "ADC1 R Power"}, + {"ADC1 R", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "I2S1"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "SAR", is_headset_type}, + + {"HPOL", NULL, "HP Amp"}, + {"HPOR", NULL, "HP Amp"}, +}; + +static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, + RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, RT5682S_TDM_EN); + else + snd_soc_component_update_bits(component, + RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682S_TDM_TX_CH_4; + val |= RT5682S_TDM_RX_CH_4; + break; + case 6: + val |= RT5682S_TDM_TX_CH_6; + val |= RT5682S_TDM_RX_CH_6; + break; + case 8: + val |= RT5682S_TDM_TX_CH_8; + val |= RT5682S_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_TDM_CTRL, + RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682S_I2S1_TX_CHL_8 | RT5682S_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682S_TDM_CL_16; + cl = RT5682S_I2S1_TX_CHL_16 | RT5682S_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682S_TDM_CL_20; + cl = RT5682S_I2S1_TX_CHL_20 | RT5682S_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682S_TDM_CL_24; + cl = RT5682S_I2S1_TX_CHL_24 | RT5682S_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682S_TDM_CL_32; + cl = RT5682S_I2S1_TX_CHL_32 | RT5682S_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_TX_CHL_MASK | RT5682S_I2S1_RX_CHL_MASK, cl); + + return 0; +} + +static int rt5682s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int frame_size; + + rt5682s->lrck[dai->id] = params_rate(params); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682S_I2S1_DL_20; + len_2 |= RT5682S_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682S_I2S1_DL_24; + len_2 |= RT5682S_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682S_I2S1_DL_32; + len_2 |= RT5682S_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682S_I2S2_DL_8; + len_2 |= RT5682S_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682S_AIF1: + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_DL_MASK, len_1); + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_MONO_MASK, RT5682S_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_MONO_MASK, RT5682S_I2S1_MONO_DIS); + break; + case RT5682S_AIF2: + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_DL_MASK, len_2); + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MONO_MASK, RT5682S_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MONO_MASK, RT5682S_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682s->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682s->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682S_I2S_BP_INV; + tdm_ctrl |= RT5682S_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682S_AIF1) + tdm_ctrl |= RT5682S_TDM_S_LP_INV | RT5682S_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682S_AIF1) + tdm_ctrl |= RT5682S_TDM_S_BP_INV | RT5682S_TDM_S_LP_INV | + RT5682S_TDM_M_BP_INV | RT5682S_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682S_I2S_DF_LEFT; + tdm_ctrl |= RT5682S_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682S_I2S_DF_PCM_A; + tdm_ctrl |= RT5682S_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682S_I2S_DF_PCM_B; + tdm_ctrl |= RT5682S_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682S_AIF1: + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_MS_MASK | RT5682S_TDM_S_BP_MASK | + RT5682S_TDM_DF_MASK | RT5682S_TDM_M_BP_MASK | + RT5682S_TDM_M_LP_MASK | RT5682S_TDM_S_LP_MASK, + tdm_ctrl | rt5682s->master[dai->id]); + break; + case RT5682S_AIF2: + if (rt5682s->master[dai->id] == 0) + reg_val |= RT5682S_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MS_MASK | RT5682S_I2S_BP_MASK | + RT5682S_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682s_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int src = 0; + + if (freq == rt5682s->sysclk && clk_id == rt5682s->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682S_SCLK_S_MCLK: + src = RT5682S_CLK_SRC_MCLK; + break; + case RT5682S_SCLK_S_PLL1: + src = RT5682S_CLK_SRC_PLL1; + break; + case RT5682S_SCLK_S_PLL2: + src = RT5682S_CLK_SRC_PLL2; + break; + case RT5682S_SCLK_S_RCCLK: + src = RT5682S_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_SCLK_SRC_MASK, src << RT5682S_SCLK_SRC_SFT); + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_1, + RT5682S_I2S_M_CLK_SRC_MASK, src << RT5682S_I2S_M_CLK_SRC_SFT); + snd_soc_component_update_bits(component, RT5682S_I2S2_M_CLK_CTRL_1, + RT5682S_I2S2_M_CLK_SRC_MASK, src << RT5682S_I2S2_M_CLK_SRC_SFT); + + rt5682s->sysclk = freq; + rt5682s->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static const struct pll_calc_map plla_table[] = { + {2048000, 24576000, 0, 46, 2, true, false, false, false}, + {256000, 24576000, 0, 382, 2, true, false, false, false}, + {512000, 24576000, 0, 190, 2, true, false, false, false}, + {4096000, 24576000, 0, 22, 2, true, false, false, false}, + {1024000, 24576000, 0, 94, 2, true, false, false, false}, + {11289600, 22579200, 1, 22, 2, false, false, false, false}, + {1411200, 22579200, 0, 62, 2, true, false, false, false}, + {2822400, 22579200, 0, 30, 2, true, false, false, false}, + {12288000, 24576000, 1, 22, 2, false, false, false, false}, + {1536000, 24576000, 0, 62, 2, true, false, false, false}, + {3072000, 24576000, 0, 30, 2, true, false, false, false}, + {24576000, 49152000, 4, 22, 0, false, false, false, false}, + {3072000, 49152000, 0, 30, 0, true, false, false, false}, + {6144000, 49152000, 0, 30, 0, false, false, false, false}, + {49152000, 98304000, 10, 22, 0, false, true, false, false}, + {6144000, 98304000, 0, 30, 0, false, true, false, false}, + {12288000, 98304000, 1, 22, 0, false, true, false, false}, + {48000000, 3840000, 10, 22, 23, false, false, false, false}, + {24000000, 3840000, 4, 22, 23, false, false, false, false}, + {19200000, 3840000, 3, 23, 23, false, false, false, false}, + {38400000, 3840000, 8, 23, 23, false, false, false, false}, +}; + +static const struct pll_calc_map pllb_table[] = { + {48000000, 24576000, 8, 6, 3, false, false, false, false}, + {48000000, 22579200, 23, 12, 3, false, false, false, true}, + {24000000, 24576000, 3, 6, 3, false, false, false, false}, + {24000000, 22579200, 23, 26, 3, false, false, false, true}, + {19200000, 24576000, 2, 6, 3, false, false, false, false}, + {19200000, 22579200, 3, 5, 3, false, false, false, true}, + {38400000, 24576000, 6, 6, 3, false, false, false, false}, + {38400000, 22579200, 8, 5, 3, false, false, false, true}, + {3840000, 49152000, 0, 6, 0, true, false, false, false}, +}; + +static int find_pll_inter_combination(unsigned int f_in, unsigned int f_out, + struct pll_calc_map *a, struct pll_calc_map *b) +{ + int i, j; + + /* Look at PLLA table */ + for (i = 0; i < ARRAY_SIZE(plla_table); i++) { + if (plla_table[i].freq_in == f_in && plla_table[i].freq_out == f_out) { + memcpy(a, plla_table + i, sizeof(*a)); + return USE_PLLA; + } + } + + /* Look at PLLB table */ + for (i = 0; i < ARRAY_SIZE(pllb_table); i++) { + if (pllb_table[i].freq_in == f_in && pllb_table[i].freq_out == f_out) { + memcpy(b, pllb_table + i, sizeof(*b)); + return USE_PLLB; + } + } + + /* Find a combination of PLLA & PLLB */ + for (i = ARRAY_SIZE(plla_table) - 1; i >= 0; i--) { + if (plla_table[i].freq_in == f_in && plla_table[i].freq_out == 3840000) { + for (j = ARRAY_SIZE(pllb_table) - 1; j >= 0; j--) { + if (pllb_table[j].freq_in == 3840000 && + pllb_table[j].freq_out == f_out) { + memcpy(a, plla_table + i, sizeof(*a)); + memcpy(b, pllb_table + j, sizeof(*b)); + return USE_PLLAB; + } + } + } + } + + return -EINVAL; +} + +static int rt5682s_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct pll_calc_map a_map, b_map; + + if (source == rt5682s->pll_src[pll_id] && freq_in == rt5682s->pll_in[pll_id] && + freq_out == rt5682s->pll_out[pll_id]) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + rt5682s->pll_in[pll_id] = 0; + rt5682s->pll_out[pll_id] = 0; + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_SCLK_SRC_MASK, RT5682S_CLK_SRC_MCLK << RT5682S_SCLK_SRC_SFT); + return 0; + } + + switch (source) { + case RT5682S_PLL_S_MCLK: + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_PLL_SRC_MASK, RT5682S_PLL_SRC_MCLK); + break; + case RT5682S_PLL_S_BCLK1: + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_PLL_SRC_MASK, RT5682S_PLL_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + rt5682s->pll_comb = find_pll_inter_combination(freq_in, freq_out, + &a_map, &b_map); + + if ((pll_id == RT5682S_PLL1 && rt5682s->pll_comb == USE_PLLA) || + (pll_id == RT5682S_PLL2 && (rt5682s->pll_comb == USE_PLLB || + rt5682s->pll_comb == USE_PLLAB))) { + dev_dbg(component->dev, + "Supported freq conversion for PLL%d:(%d->%d): %d\n", + pll_id + 1, freq_in, freq_out, rt5682s->pll_comb); + } else { + dev_err(component->dev, + "Unsupported freq conversion for PLL%d:(%d->%d): %d\n", + pll_id + 1, freq_in, freq_out, rt5682s->pll_comb); + return -EINVAL; + } + + if (rt5682s->pll_comb == USE_PLLA || rt5682s->pll_comb == USE_PLLAB) { + dev_dbg(component->dev, + "PLLA: fin=%d fout=%d m_bp=%d k_bp=%d m=%d n=%d k=%d\n", + a_map.freq_in, a_map.freq_out, a_map.m_bp, a_map.k_bp, + (a_map.m_bp ? 0 : a_map.m), a_map.n, (a_map.k_bp ? 0 : a_map.k)); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_1, + RT5682S_PLLA_N_MASK, a_map.n); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_2, + RT5682S_PLLA_M_MASK | RT5682S_PLLA_K_MASK, + a_map.m << RT5682S_PLLA_M_SFT | a_map.k); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_6, + RT5682S_PLLA_M_BP_MASK | RT5682S_PLLA_K_BP_MASK, + a_map.m_bp << RT5682S_PLLA_M_BP_SFT | + a_map.k_bp << RT5682S_PLLA_K_BP_SFT); + } + + if (rt5682s->pll_comb == USE_PLLB || rt5682s->pll_comb == USE_PLLAB) { + dev_dbg(component->dev, + "PLLB: fin=%d fout=%d m_bp=%d k_bp=%d m=%d n=%d k=%d byp_ps=%d sel_ps=%d\n", + b_map.freq_in, b_map.freq_out, b_map.m_bp, b_map.k_bp, + (b_map.m_bp ? 0 : b_map.m), b_map.n, (b_map.k_bp ? 0 : b_map.k), + b_map.byp_ps, b_map.sel_ps); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_3, + RT5682S_PLLB_N_MASK, b_map.n); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_4, + RT5682S_PLLB_M_MASK | RT5682S_PLLB_K_MASK, + b_map.m << RT5682S_PLLB_M_SFT | b_map.k); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_6, + RT5682S_PLLB_SEL_PS_MASK | RT5682S_PLLB_BYP_PS_MASK | + RT5682S_PLLB_M_BP_MASK | RT5682S_PLLB_K_BP_MASK, + b_map.sel_ps << RT5682S_PLLB_SEL_PS_SFT | + b_map.byp_ps << RT5682S_PLLB_BYP_PS_SFT | + b_map.m_bp << RT5682S_PLLB_M_BP_SFT | + b_map.k_bp << RT5682S_PLLB_K_BP_SFT); + } + + if (rt5682s->pll_comb == USE_PLLB) + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_7, + RT5682S_PLLB_SRC_MASK, RT5682S_PLLB_SRC_DFIN); + + rt5682s->pll_in[pll_id] = freq_in; + rt5682s->pll_out[pll_id] = freq_out; + rt5682s->pll_src[pll_id] = source; + + return 0; +} + +static int rt5682s_set_bclk1_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s->bclk[dai->id] = ratio; + + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_2, + RT5682S_I2S2_BCLK_MS2_MASK, RT5682S_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_2, + RT5682S_I2S2_BCLK_MS2_MASK, RT5682S_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_PWR_LDO, RT5682S_PWR_LDO); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_DIG_GATE_CTRL, RT5682S_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_DIG_GATE_CTRL | RT5682S_PWR_LDO, 0); + break; + case SND_SOC_BIAS_ON: + break; + } + + return 0; +} + +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_48 48000 +#define CLK_44 44100 + +static bool rt5682s_clk_check(struct rt5682s_priv *rt5682s) +{ + if (!rt5682s->master[RT5682S_AIF1]) { + dev_dbg(rt5682s->component->dev, "dai clk fmt not set correctly\n"); + return false; + } + return true; +} + +static int rt5682s_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_MB, RT5682S_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_FV2, RT5682S_PWR_VREF2); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV2, RT5682S_PWR_FV2); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + /* Only need to power PLLB due to the rate set restriction */ + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLLB"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682s_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + if (!rt5682s_clk_check(rt5682s)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); + if (!rt5682s->jack_type) + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_FV2 | RT5682S_PWR_MB, 0); + + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLLB"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682s_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + const char * const clk_name = clk_hw_get_name(hw); + + if (!rt5682s_clk_check(rt5682s)) + return 0; + /* + * Only accept to set wclk rate to 44.1k or 48kHz. + */ + if (rt5682s->lrck[RT5682S_AIF1] != CLK_48 && + rt5682s->lrck[RT5682S_AIF1] != CLK_44) { + dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + __func__, clk_name, CLK_44, CLK_48); + return 0; + } + + return rt5682s->lrck[RT5682S_AIF1]; +} + +static long rt5682s_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + const char * const clk_name = clk_hw_get_name(hw); + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + /* + * Only accept to set wclk rate to 44.1k or 48kHz. + * It will force to 48kHz if not both. + */ + if (rate != CLK_48 && rate != CLK_44) { + dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + __func__, clk_name, CLK_44, CLK_48); + rate = CLK_48; + } + + return rate; +} + +static int rt5682s_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct clk *parent_clk; + const char * const clk_name = clk_hw_get_name(hw); + unsigned int clk_pll2_fout; + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + /* + * Whether the wclk's parent clk (mclk) exists or not, please ensure + * it is fixed or set to 48MHz before setting wclk rate. It's a + * temporary limitation. Only accept 48MHz clk as the clk provider. + * + * It will set the codec anyway by assuming mclk is 48MHz. + */ + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + /* + * To achieve the rate conversion from 48MHz to 44.1k or 48kHz, + * PLL2 is needed. + */ + clk_pll2_fout = rate * 512; + rt5682s_set_component_pll(component, RT5682S_PLL2, RT5682S_PLL_S_MCLK, + CLK_PLL2_FIN, clk_pll2_fout); + + rt5682s_set_component_sysclk(component, RT5682S_SCLK_S_PLL2, 0, + clk_pll2_fout, SND_SOC_CLOCK_IN); + + rt5682s->lrck[RT5682S_AIF1] = rate; + + return 0; +} + +static unsigned long rt5682s_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + unsigned int bclks_per_wclk; + + bclks_per_wclk = snd_soc_component_read(component, RT5682S_TDM_TCON_CTRL_1); + + switch (bclks_per_wclk & RT5682S_TDM_BCLK_MS1_MASK) { + case RT5682S_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682S_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682S_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682S_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682s_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682s_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682s_clk_check(rt5682s)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682s_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682s_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dai *dai; + unsigned long factor; + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + factor = rt5682s_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682S_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682S_AIF1); + return -ENODEV; + } + + return rt5682s_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682s_dai_clk_ops[RT5682S_DAI_NUM_CLKS] = { + [RT5682S_DAI_WCLK_IDX] = { + .prepare = rt5682s_wclk_prepare, + .unprepare = rt5682s_wclk_unprepare, + .recalc_rate = rt5682s_wclk_recalc_rate, + .round_rate = rt5682s_wclk_round_rate, + .set_rate = rt5682s_wclk_set_rate, + }, + [RT5682S_DAI_BCLK_IDX] = { + .recalc_rate = rt5682s_bclk_recalc_rate, + .round_rate = rt5682s_bclk_round_rate, + .set_rate = rt5682s_bclk_set_rate, + }, +}; + +static int rt5682s_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct rt5682s_platform_data *pdata = &rt5682s->pdata; + struct clk_hw *dai_clk_hw; + int i, ret; + + for (i = 0; i < RT5682S_DAI_NUM_CLKS; ++i) { + struct clk_init_data init = { }; + + dai_clk_hw = &rt5682s->dai_clks_hw[i]; + + switch (i) { + case RT5682S_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682s->mclk) { + init.parent_data = &(struct clk_parent_data){ + .fw_name = "mclk", + }; + init.num_parents = 1; + } + break; + case RT5682S_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + init.parent_hws = &(const struct clk_hw *){ + &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX] + }; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + return -EINVAL; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682s_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + ret = devm_clk_hw_register(dev, dai_clk_hw); + if (ret) { + dev_warn(dev, "Failed to register %s: %d\n", init.name, ret); + return ret; + } + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + } else { + ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, + init.name, dev_name(dev)); + if (ret) + return ret; + } + } + + return 0; +} + +static int rt5682s_dai_probe_clks(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int ret; + + /* Check if MCLK provided */ + rt5682s->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682s->mclk)) { + if (PTR_ERR(rt5682s->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682s->mclk); + return ret; + } + rt5682s->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682s_register_dai_clks(component); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682s->lrck[RT5682S_AIF1] = CLK_48; + + return 0; +} +#else +static inline int rt5682s_dai_probe_clks(struct snd_soc_component *component) +{ + return 0; +} +#endif /* CONFIG_COMMON_CLK */ + +static int rt5682s_probe(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; + int ret; + + rt5682s->component = component; + + ret = rt5682s_dai_probe_clks(component); + if (ret) + return ret; + + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "Vref2"); + snd_soc_dapm_sync(dapm); + return 0; +} + +static void rt5682s_remove(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s_reset(rt5682s); +} + +#ifdef CONFIG_PM +static int rt5682s_suspend(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + if (rt5682s->hs_jack && rt5682s->jack_type == SND_JACK_HEADSET) + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + + regcache_cache_only(rt5682s->regmap, true); + regcache_mark_dirty(rt5682s->regmap); + + return 0; +} + +static int rt5682s_resume(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682s->regmap, false); + regcache_sync(rt5682s->regmap); + + if (rt5682s->hs_jack) { + rt5682s->jack_type = 0; + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} +#else +#define rt5682s_suspend NULL +#define rt5682s_resume NULL +#endif + +const struct snd_soc_dai_ops rt5682s_aif1_dai_ops = { + .hw_params = rt5682s_hw_params, + .set_fmt = rt5682s_set_dai_fmt, + .set_tdm_slot = rt5682s_set_tdm_slot, + .set_bclk_ratio = rt5682s_set_bclk1_ratio, +}; + +const struct snd_soc_dai_ops rt5682s_aif2_dai_ops = { + .hw_params = rt5682s_hw_params, + .set_fmt = rt5682s_set_dai_fmt, + .set_bclk_ratio = rt5682s_set_bclk2_ratio, +}; + +const struct snd_soc_component_driver rt5682s_soc_component_dev = { + .probe = rt5682s_probe, + .remove = rt5682s_remove, + .suspend = rt5682s_suspend, + .resume = rt5682s_resume, + .set_bias_level = rt5682s_set_bias_level, + .controls = rt5682s_snd_controls, + .num_controls = ARRAY_SIZE(rt5682s_snd_controls), + .dapm_widgets = rt5682s_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682s_dapm_widgets), + .dapm_routes = rt5682s_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682s_dapm_routes), + .set_sysclk = rt5682s_set_component_sysclk, + .set_pll = rt5682s_set_component_pll, + .set_jack = rt5682s_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682s->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682s->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682s->pdata.jd_src); + device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", + &rt5682s->pdata.dmic_clk_rate); + device_property_read_u32(dev, "realtek,dmic-delay-ms", + &rt5682s->pdata.dmic_delay); + + rt5682s->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + if (device_property_read_string_array(dev, "clock-output-names", + rt5682s->pdata.dai_clk_names, + RT5682S_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682s->pdata.dai_clk_names[RT5682S_DAI_WCLK_IDX], + rt5682s->pdata.dai_clk_names[RT5682S_DAI_BCLK_IDX]); + + rt5682s->pdata.dmic_clk_driving_high = device_property_read_bool(dev, + "realtek,dmic-clk-driving-high"); + + return 0; +} + +static void rt5682s_calibrate(struct rt5682s_priv *rt5682s) +{ + unsigned int count, value; + + mutex_lock(&rt5682s->calibrate_mutex); + + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0xaa80); + usleep_range(15000, 20000); + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0xfa80); + regmap_write(rt5682s->regmap, RT5682S_PWR_DIG_1, 0x01c0); + regmap_write(rt5682s->regmap, RT5682S_MICBIAS_2, 0x0380); + regmap_write(rt5682s->regmap, RT5682S_GLB_CLK, 0x8000); + regmap_write(rt5682s->regmap, RT5682S_ADDA_CLK_1, 0x1001); + regmap_write(rt5682s->regmap, RT5682S_CHOP_DAC_2, 0x3030); + regmap_write(rt5682s->regmap, RT5682S_CHOP_ADC, 0xb000); + regmap_write(rt5682s->regmap, RT5682S_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682s->regmap, RT5682S_CAL_REC, 0x5151); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682s->regmap, RT5682S_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682s->regmap, RT5682S_HP_CALIB_ST_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + dev_err(rt5682s->component->dev, "HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682s->regmap, RT5682S_MICBIAS_2, 0x0180); + regmap_write(rt5682s->regmap, RT5682S_CAL_REC, 0x5858); + regmap_write(rt5682s->regmap, RT5682S_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_2, 0x0320); + regmap_write(rt5682s->regmap, RT5682S_PWR_DIG_1, 0x00c0); + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0x0800); + regmap_write(rt5682s->regmap, RT5682S_GLB_CLK, 0x0000); + + mutex_unlock(&rt5682s->calibrate_mutex); +} + +static const struct regmap_config rt5682s_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682S_MAX_REG, + .volatile_reg = rt5682s_volatile_register, + .readable_reg = rt5682s_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682s_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682s_reg), + .use_single_read = true, + .use_single_write = true, +}; + +static struct snd_soc_dai_driver rt5682s_dai[] = { + { + .name = "rt5682s-aif1", + .id = RT5682S_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .ops = &rt5682s_aif1_dai_ops, + }, + { + .name = "rt5682s-aif2", + .id = RT5682S_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .ops = &rt5682s_aif2_dai_ops, + }, +}; + +static void rt5682s_i2c_disable_regulators(void *data) +{ + struct rt5682s_priv *rt5682s = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); +} + +static int rt5682s_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682s_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682s_priv *rt5682s; + int i, ret; + unsigned int val; + + rt5682s = devm_kzalloc(&i2c->dev, sizeof(struct rt5682s_priv), GFP_KERNEL); + if (!rt5682s) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682s); + + rt5682s->pdata = i2s_default_platform_data; + + if (pdata) + rt5682s->pdata = *pdata; + else + rt5682s_parse_dt(rt5682s, &i2c->dev); + + rt5682s->regmap = devm_regmap_init_i2c(i2c, &rt5682s_regmap); + if (IS_ERR(rt5682s->regmap)) { + ret = PTR_ERR(rt5682s->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682s->supplies); i++) + rt5682s->supplies[i].supply = rt5682s_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = devm_add_action_or_reset(&i2c->dev, rt5682s_i2c_disable_regulators, rt5682s); + if (ret) + return ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682s->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682s->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682s")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 50 ms minimum */ + usleep_range(50000, 55000); + + regmap_read(rt5682s->regmap, RT5682S_DEVICE_ID, &val); + if (val != DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %x is not rt5682s\n", val); + return -ENODEV; + } + + rt5682s_reset(rt5682s); + rt5682s_apply_patch_list(rt5682s, &i2c->dev); + + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_2, + RT5682S_DLDO_I_LIMIT_MASK, RT5682S_DLDO_I_LIMIT_DIS); + usleep_range(20000, 25000); + + mutex_init(&rt5682s->calibrate_mutex); + mutex_init(&rt5682s->sar_mutex); + rt5682s_calibrate(rt5682s); + + regmap_update_bits(rt5682s->regmap, RT5682S_MICBIAS_2, + RT5682S_PWR_CLK25M_MASK | RT5682S_PWR_CLK1M_MASK, + RT5682S_PWR_CLK25M_PD | RT5682S_PWR_CLK1M_PU); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_1, + RT5682S_PWR_BG, RT5682S_PWR_BG); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_LOGIC_CTRL_2, + RT5682S_HP_SIG_SRC_MASK, RT5682S_HP_SIG_SRC_1BIT_CTL); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_PM_HP_MASK, RT5682S_PM_HP_HV); + + /* DMIC data pin */ + switch (rt5682s->pdata.dmic1_data_pin) { + case RT5682S_DMIC1_DATA_NULL: + break; + case RT5682S_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682s->regmap, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_DP_MASK, RT5682S_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP2_PIN_MASK, RT5682S_GP2_PIN_DMIC_SDA); + break; + case RT5682S_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682s->regmap, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_DP_MASK, RT5682S_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP5_PIN_MASK, RT5682S_GP5_PIN_DMIC_SDA); + break; + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + /* DMIC clk pin */ + switch (rt5682s->pdata.dmic1_clk_pin) { + case RT5682S_DMIC1_CLK_NULL: + break; + case RT5682S_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP1_PIN_MASK, RT5682S_GP1_PIN_DMIC_CLK); + break; + case RT5682S_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP3_PIN_MASK, RT5682S_GP3_PIN_DMIC_CLK); + if (rt5682s->pdata.dmic_clk_driving_high) + regmap_update_bits(rt5682s->regmap, RT5682S_PAD_DRIVING_CTRL, + RT5682S_PAD_DRV_GP3_MASK, RT5682S_PAD_DRV_GP3_HIGH); + break; + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + + INIT_DELAYED_WORK(&rt5682s->jack_detect_work, rt5682s_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682s->jd_check_work, rt5682s_jd_check_handler); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682s_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5682s", rt5682s); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5682s_soc_component_dev, + rt5682s_dai, ARRAY_SIZE(rt5682s_dai)); +} + +static void rt5682s_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682s_priv *rt5682s = i2c_get_clientdata(client); + + disable_irq(client->irq); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + rt5682s_reset(rt5682s); +} + +static int rt5682s_i2c_remove(struct i2c_client *client) +{ + rt5682s_i2c_shutdown(client); + + return 0; +} + +static const struct of_device_id rt5682s_of_match[] = { + {.compatible = "realtek,rt5682s"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682s_of_match); + +static const struct acpi_device_id rt5682s_acpi_match[] = { + {"RTL5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682s_acpi_match); + +static const struct i2c_device_id rt5682s_i2c_id[] = { + {"rt5682s", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682s_i2c_id); + +static struct i2c_driver rt5682s_i2c_driver = { + .driver = { + .name = "rt5682s", + .of_match_table = rt5682s_of_match, + .acpi_match_table = rt5682s_acpi_match, + .probe_type = PROBE_PREFER_ASYNCHRONOUS, + }, + .probe = rt5682s_i2c_probe, + .remove = rt5682s_i2c_remove, + .shutdown = rt5682s_i2c_shutdown, + .id_table = rt5682s_i2c_id, +}; +module_i2c_driver(rt5682s_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682I-VS driver"); +MODULE_AUTHOR("Derek Fang <derek.fang(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h new file mode 100644 index 000000000000..7c755e5efb81 --- /dev/null +++ b/sound/soc/codecs/rt5682s.h @@ -0,0 +1,1455 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5682s.h -- RT5682I-VS ALSA SoC audio driver + * + * Copyright 2021 Realtek Microelectronics + * Author: Derek Fang <derek.fang(a)realtek.com> + */ + +#ifndef __RT5682S_H__ +#define __RT5682S_H__ + +#include <sound/rt5682s.h> +#include <linux/regulator/consumer.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> + + +/* Info */ +#define RT5682S_RESET 0x0000 +#define RT5682S_VERSION_ID 0x00fd +#define RT5682S_VENDOR_ID 0x00fe +#define RT5682S_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682S_HP_CTRL_1 0x0002 +#define RT5682S_HP_CTRL_2 0x0003 +#define RT5682S_HPL_GAIN 0x0005 +#define RT5682S_HPR_GAIN 0x0006 + +#define RT5682S_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682S_CBJ_BST_CTRL 0x000b +#define RT5682S_CBJ_DET_CTRL 0x000f +#define RT5682S_CBJ_CTRL_1 0x0010 +#define RT5682S_CBJ_CTRL_2 0x0011 +#define RT5682S_CBJ_CTRL_3 0x0012 +#define RT5682S_CBJ_CTRL_4 0x0013 +#define RT5682S_CBJ_CTRL_5 0x0014 +#define RT5682S_CBJ_CTRL_6 0x0015 +#define RT5682S_CBJ_CTRL_7 0x0016 +#define RT5682S_CBJ_CTRL_8 0x0017 +/* I/O - ADC/DAC/DMIC */ +#define RT5682S_DAC1_DIG_VOL 0x0019 +#define RT5682S_STO1_ADC_DIG_VOL 0x001c +#define RT5682S_STO1_ADC_BOOST 0x001f +#define RT5682S_HP_IMP_GAIN_1 0x0022 +#define RT5682S_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682S_SIDETONE_CTRL 0x0024 +#define RT5682S_STO1_ADC_MIXER 0x0026 +#define RT5682S_AD_DA_MIXER 0x0029 +#define RT5682S_STO1_DAC_MIXER 0x002a +#define RT5682S_A_DAC1_MUX 0x002b +#define RT5682S_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682S_REC_MIXER 0x003c +#define RT5682S_CAL_REC 0x0044 +/* HP Analog Offset Control */ +#define RT5682S_HP_ANA_OST_CTRL_1 0x004b +#define RT5682S_HP_ANA_OST_CTRL_2 0x004c +#define RT5682S_HP_ANA_OST_CTRL_3 0x004d +/* Power */ +#define RT5682S_PWR_DIG_1 0x0061 +#define RT5682S_PWR_DIG_2 0x0062 +#define RT5682S_PWR_ANLG_1 0x0063 +#define RT5682S_PWR_ANLG_2 0x0064 +#define RT5682S_PWR_ANLG_3 0x0065 +#define RT5682S_PWR_MIXER 0x0066 + +#define RT5682S_MB_CTRL 0x0067 +#define RT5682S_CLK_GATE_TCON_1 0x0068 +#define RT5682S_CLK_GATE_TCON_2 0x0069 +#define RT5682S_CLK_GATE_TCON_3 0x006a +/* Clock Detect */ +#define RT5682S_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682S_RESET_LPF_CTRL 0x006c +#define RT5682S_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682S_DMIC_CTRL_1 0x006e +#define RT5682S_LPF_AD_DMIC 0x006f +/* Format - ADC/DAC */ +#define RT5682S_I2S1_SDP 0x0070 +#define RT5682S_I2S2_SDP 0x0071 +#define RT5682S_ADDA_CLK_1 0x0073 +#define RT5682S_ADDA_CLK_2 0x0074 +#define RT5682S_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682S_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682S_TDM_CTRL 0x0079 +#define RT5682S_TDM_ADDA_CTRL_1 0x007a +#define RT5682S_TDM_ADDA_CTRL_2 0x007b +#define RT5682S_DATA_SEL_CTRL_1 0x007c +#define RT5682S_TDM_TCON_CTRL_1 0x007e +#define RT5682S_TDM_TCON_CTRL_2 0x007f +/* Function - Analog */ +#define RT5682S_GLB_CLK 0x0080 +#define RT5682S_PLL_TRACK_1 0x0083 +#define RT5682S_PLL_TRACK_2 0x0084 +#define RT5682S_PLL_TRACK_3 0x0085 +#define RT5682S_PLL_TRACK_4 0x0086 +#define RT5682S_PLL_TRACK_5 0x0087 +#define RT5682S_PLL_TRACK_6 0x0088 +#define RT5682S_PLL_TRACK_11 0x008c +#define RT5682S_DEPOP_1 0x008e +#define RT5682S_HP_CHARGE_PUMP_1 0x008f +#define RT5682S_HP_CHARGE_PUMP_2 0x0091 +#define RT5682S_HP_CHARGE_PUMP_3 0x0092 +#define RT5682S_MICBIAS_1 0x0093 +#define RT5682S_MICBIAS_2 0x0094 +#define RT5682S_MICBIAS_3 0x0095 + +#define RT5682S_PLL_TRACK_12 0x0096 +#define RT5682S_PLL_TRACK_14 0x0097 +#define RT5682S_PLL_CTRL_1 0x0098 +#define RT5682S_PLL_CTRL_2 0x0099 +#define RT5682S_PLL_CTRL_3 0x009a +#define RT5682S_PLL_CTRL_4 0x009b +#define RT5682S_PLL_CTRL_5 0x009c +#define RT5682S_PLL_CTRL_6 0x009d +#define RT5682S_PLL_CTRL_7 0x009e + +#define RT5682S_RC_CLK_CTRL 0x009f +#define RT5682S_I2S2_M_CLK_CTRL_1 0x00a0 +#define RT5682S_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682S_I2S2_F_DIV_CTRL_2 0x00a4 + +#define RT5682S_IRQ_CTRL_1 0x00b6 +#define RT5682S_IRQ_CTRL_2 0x00b7 +#define RT5682S_IRQ_CTRL_3 0x00b8 +#define RT5682S_IRQ_CTRL_4 0x00b9 +#define RT5682S_INT_ST_1 0x00be +#define RT5682S_GPIO_CTRL_1 0x00c0 +#define RT5682S_GPIO_CTRL_2 0x00c1 +#define RT5682S_GPIO_ST 0x00c2 +#define RT5682S_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682S_MID_HP_AMP_DET 0x00d2 +#define RT5682S_LOW_HP_AMP_DET 0x00d3 +#define RT5682S_DELAY_BUF_CTRL 0x00d4 +#define RT5682S_SV_ZCD_1 0x00d9 +#define RT5682S_SV_ZCD_2 0x00da +#define RT5682S_IL_CMD_1 0x00db +#define RT5682S_IL_CMD_2 0x00dc +#define RT5682S_IL_CMD_3 0x00dd +#define RT5682S_IL_CMD_4 0x00de +#define RT5682S_IL_CMD_5 0x00df +#define RT5682S_IL_CMD_6 0x00e0 +#define RT5682S_4BTN_IL_CMD_1 0x00e2 +#define RT5682S_4BTN_IL_CMD_2 0x00e3 +#define RT5682S_4BTN_IL_CMD_3 0x00e4 +#define RT5682S_4BTN_IL_CMD_4 0x00e5 +#define RT5682S_4BTN_IL_CMD_5 0x00e6 +#define RT5682S_4BTN_IL_CMD_6 0x00e7 +#define RT5682S_4BTN_IL_CMD_7 0x00e8 + +#define RT5682S_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682S_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682S_AJD1_CTRL 0x00f0 +#define RT5682S_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682S_DUMMY_1 0x00fa +#define RT5682S_DUMMY_2 0x00fb +#define RT5682S_DUMMY_3 0x00fc + +#define RT5682S_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682S_BIAS_CUR_CTRL_2 0x010b +#define RT5682S_BIAS_CUR_CTRL_3 0x010c +#define RT5682S_BIAS_CUR_CTRL_4 0x010d +#define RT5682S_BIAS_CUR_CTRL_5 0x010e +#define RT5682S_BIAS_CUR_CTRL_6 0x010f +#define RT5682S_BIAS_CUR_CTRL_7 0x0110 +#define RT5682S_BIAS_CUR_CTRL_8 0x0111 +#define RT5682S_BIAS_CUR_CTRL_9 0x0112 +#define RT5682S_BIAS_CUR_CTRL_10 0x0113 +#define RT5682S_VREF_REC_OP_FB_CAP_CTRL_1 0x0117 +#define RT5682S_VREF_REC_OP_FB_CAP_CTRL_2 0x0118 +#define RT5682S_CHARGE_PUMP_1 0x0125 +#define RT5682S_DIG_IN_CTRL_1 0x0132 +#define RT5682S_PAD_DRIVING_CTRL 0x0136 +#define RT5682S_CHOP_DAC_1 0x0139 +#define RT5682S_CHOP_DAC_2 0x013a +#define RT5682S_CHOP_ADC 0x013b +#define RT5682S_CALIB_ADC_CTRL 0x013c +#define RT5682S_VOL_TEST 0x013f +#define RT5682S_SPKVDD_DET_ST 0x0142 +#define RT5682S_TEST_MODE_CTRL_1 0x0145 +#define RT5682S_TEST_MODE_CTRL_2 0x0146 +#define RT5682S_TEST_MODE_CTRL_3 0x0147 +#define RT5682S_TEST_MODE_CTRL_4 0x0148 +#define RT5682S_PLL_INTERNAL_1 0x0156 +#define RT5682S_PLL_INTERNAL_2 0x0157 +#define RT5682S_PLL_INTERNAL_3 0x0158 +#define RT5682S_PLL_INTERNAL_4 0x0159 +#define RT5682S_STO_NG2_CTRL_1 0x0160 +#define RT5682S_STO_NG2_CTRL_2 0x0161 +#define RT5682S_STO_NG2_CTRL_3 0x0162 +#define RT5682S_STO_NG2_CTRL_4 0x0163 +#define RT5682S_STO_NG2_CTRL_5 0x0164 +#define RT5682S_STO_NG2_CTRL_6 0x0165 +#define RT5682S_STO_NG2_CTRL_7 0x0166 +#define RT5682S_STO_NG2_CTRL_8 0x0167 +#define RT5682S_STO_NG2_CTRL_9 0x0168 +#define RT5682S_STO_NG2_CTRL_10 0x0169 +#define RT5682S_STO1_DAC_SIL_DET 0x0190 +#define RT5682S_SIL_PSV_CTRL1 0x0194 +#define RT5682S_SIL_PSV_CTRL2 0x0195 +#define RT5682S_SIL_PSV_CTRL3 0x0197 +#define RT5682S_SIL_PSV_CTRL4 0x0198 +#define RT5682S_SIL_PSV_CTRL5 0x0199 +#define RT5682S_HP_IMP_SENS_CTRL_1 0x01ac +#define RT5682S_HP_IMP_SENS_CTRL_2 0x01ad +#define RT5682S_HP_IMP_SENS_CTRL_3 0x01ae +#define RT5682S_HP_IMP_SENS_CTRL_4 0x01af +#define RT5682S_HP_IMP_SENS_CTRL_5 0x01b0 +#define RT5682S_HP_IMP_SENS_CTRL_6 0x01b1 +#define RT5682S_HP_IMP_SENS_CTRL_7 0x01b2 +#define RT5682S_HP_IMP_SENS_CTRL_8 0x01b3 +#define RT5682S_HP_IMP_SENS_CTRL_9 0x01b4 +#define RT5682S_HP_IMP_SENS_CTRL_10 0x01b5 +#define RT5682S_HP_IMP_SENS_CTRL_11 0x01b6 +#define RT5682S_HP_IMP_SENS_CTRL_12 0x01b7 +#define RT5682S_HP_IMP_SENS_CTRL_13 0x01b8 +#define RT5682S_HP_IMP_SENS_CTRL_14 0x01b9 +#define RT5682S_HP_IMP_SENS_CTRL_15 0x01ba +#define RT5682S_HP_IMP_SENS_CTRL_16 0x01bb +#define RT5682S_HP_IMP_SENS_CTRL_17 0x01bc +#define RT5682S_HP_IMP_SENS_CTRL_18 0x01bd +#define RT5682S_HP_IMP_SENS_CTRL_19 0x01be +#define RT5682S_HP_IMP_SENS_CTRL_20 0x01bf +#define RT5682S_HP_IMP_SENS_CTRL_21 0x01c0 +#define RT5682S_HP_IMP_SENS_CTRL_22 0x01c1 +#define RT5682S_HP_IMP_SENS_CTRL_23 0x01c2 +#define RT5682S_HP_IMP_SENS_CTRL_24 0x01c3 +#define RT5682S_HP_IMP_SENS_CTRL_25 0x01c4 +#define RT5682S_HP_IMP_SENS_CTRL_26 0x01c5 +#define RT5682S_HP_IMP_SENS_CTRL_27 0x01c6 +#define RT5682S_HP_IMP_SENS_CTRL_28 0x01c7 +#define RT5682S_HP_IMP_SENS_CTRL_29 0x01c8 +#define RT5682S_HP_IMP_SENS_CTRL_30 0x01c9 +#define RT5682S_HP_IMP_SENS_CTRL_31 0x01ca +#define RT5682S_HP_IMP_SENS_CTRL_32 0x01cb +#define RT5682S_HP_IMP_SENS_CTRL_33 0x01cc +#define RT5682S_HP_IMP_SENS_CTRL_34 0x01cd +#define RT5682S_HP_IMP_SENS_CTRL_35 0x01ce +#define RT5682S_HP_IMP_SENS_CTRL_36 0x01cf +#define RT5682S_HP_IMP_SENS_CTRL_37 0x01d0 +#define RT5682S_HP_IMP_SENS_CTRL_38 0x01d1 +#define RT5682S_HP_IMP_SENS_CTRL_39 0x01d2 +#define RT5682S_HP_IMP_SENS_CTRL_40 0x01d3 +#define RT5682S_HP_IMP_SENS_CTRL_41 0x01d4 +#define RT5682S_HP_IMP_SENS_CTRL_42 0x01d5 +#define RT5682S_HP_IMP_SENS_CTRL_43 0x01d6 +#define RT5682S_HP_IMP_SENS_CTRL_44 0x01d7 +#define RT5682S_HP_IMP_SENS_CTRL_45 0x01d8 +#define RT5682S_HP_IMP_SENS_CTRL_46 0x01d9 +#define RT5682S_HP_LOGIC_CTRL_1 0x01da +#define RT5682S_HP_LOGIC_CTRL_2 0x01db +#define RT5682S_HP_LOGIC_CTRL_3 0x01dc +#define RT5682S_HP_CALIB_CTRL_1 0x01de +#define RT5682S_HP_CALIB_CTRL_2 0x01df +#define RT5682S_HP_CALIB_CTRL_3 0x01e0 +#define RT5682S_HP_CALIB_CTRL_4 0x01e1 +#define RT5682S_HP_CALIB_CTRL_5 0x01e2 +#define RT5682S_HP_CALIB_CTRL_6 0x01e3 +#define RT5682S_HP_CALIB_CTRL_7 0x01e4 +#define RT5682S_HP_CALIB_CTRL_8 0x01e5 +#define RT5682S_HP_CALIB_CTRL_9 0x01e6 +#define RT5682S_HP_CALIB_CTRL_10 0x01e7 +#define RT5682S_HP_CALIB_CTRL_11 0x01e8 +#define RT5682S_HP_CALIB_ST_1 0x01ea +#define RT5682S_HP_CALIB_ST_2 0x01eb +#define RT5682S_HP_CALIB_ST_3 0x01ec +#define RT5682S_HP_CALIB_ST_4 0x01ed +#define RT5682S_HP_CALIB_ST_5 0x01ee +#define RT5682S_HP_CALIB_ST_6 0x01ef +#define RT5682S_HP_CALIB_ST_7 0x01f0 +#define RT5682S_HP_CALIB_ST_8 0x01f1 +#define RT5682S_HP_CALIB_ST_9 0x01f2 +#define RT5682S_HP_CALIB_ST_10 0x01f3 +#define RT5682S_HP_CALIB_ST_11 0x01f4 +#define RT5682S_SAR_IL_CMD_1 0x0210 +#define RT5682S_SAR_IL_CMD_2 0x0211 +#define RT5682S_SAR_IL_CMD_3 0x0212 +#define RT5682S_SAR_IL_CMD_4 0x0213 +#define RT5682S_SAR_IL_CMD_5 0x0214 +#define RT5682S_SAR_IL_CMD_6 0x0215 +#define RT5682S_SAR_IL_CMD_7 0x0216 +#define RT5682S_SAR_IL_CMD_8 0x0217 +#define RT5682S_SAR_IL_CMD_9 0x0218 +#define RT5682S_SAR_IL_CMD_10 0x0219 +#define RT5682S_SAR_IL_CMD_11 0x021a +#define RT5682S_SAR_IL_CMD_12 0x021b +#define RT5682S_SAR_IL_CMD_13 0x021c +#define RT5682S_SAR_IL_CMD_14 0x021d +#define RT5682S_DUMMY_4 0x02fa +#define RT5682S_DUMMY_5 0x02fb +#define RT5682S_DUMMY_6 0x02fc +#define RT5682S_VERSION_ID_HIDE 0x03fe +#define RT5682S_VERSION_ID_CUS 0x03ff +#define RT5682S_SCAN_CTL 0x0500 +#define RT5682S_HP_AMP_DET 0x0600 +#define RT5682S_BIAS_CUR_CTRL_11 0x0610 +#define RT5682S_BIAS_CUR_CTRL_12 0x0611 +#define RT5682S_BIAS_CUR_CTRL_13 0x0620 +#define RT5682S_BIAS_CUR_CTRL_14 0x0621 +#define RT5682S_BIAS_CUR_CTRL_15 0x0630 +#define RT5682S_BIAS_CUR_CTRL_16 0x0631 +#define RT5682S_BIAS_CUR_CTRL_17 0x0640 +#define RT5682S_BIAS_CUR_CTRL_18 0x0641 +#define RT5682S_I2C_TRANS_CTRL 0x07fa +#define RT5682S_DUMMY_7 0x08fa +#define RT5682S_DUMMY_8 0x08fb +#define RT5682S_DMIC_FLOAT_DET 0x0d00 +#define RT5682S_HA_CMP_OP_1 0x1100 +#define RT5682S_HA_CMP_OP_2 0x1101 +#define RT5682S_HA_CMP_OP_3 0x1102 +#define RT5682S_HA_CMP_OP_4 0x1103 +#define RT5682S_HA_CMP_OP_5 0x1104 +#define RT5682S_HA_CMP_OP_6 0x1105 +#define RT5682S_HA_CMP_OP_7 0x1106 +#define RT5682S_HA_CMP_OP_8 0x1107 +#define RT5682S_HA_CMP_OP_9 0x1108 +#define RT5682S_HA_CMP_OP_10 0x1109 +#define RT5682S_HA_CMP_OP_11 0x110a +#define RT5682S_HA_CMP_OP_12 0x110b +#define RT5682S_HA_CMP_OP_13 0x110c +#define RT5682S_HA_CMP_OP_14 0x1111 +#define RT5682S_HA_CMP_OP_15 0x1112 +#define RT5682S_HA_CMP_OP_16 0x1113 +#define RT5682S_HA_CMP_OP_17 0x1114 +#define RT5682S_HA_CMP_OP_18 0x1115 +#define RT5682S_HA_CMP_OP_19 0x1116 +#define RT5682S_HA_CMP_OP_20 0x1117 +#define RT5682S_HA_CMP_OP_21 0x1118 +#define RT5682S_HA_CMP_OP_22 0x1119 +#define RT5682S_HA_CMP_OP_23 0x111a +#define RT5682S_HA_CMP_OP_24 0x111b +#define RT5682S_HA_CMP_OP_25 0x111c +#define RT5682S_NEW_CBJ_DET_CTL_1 0x1401 +#define RT5682S_NEW_CBJ_DET_CTL_2 0x1402 +#define RT5682S_NEW_CBJ_DET_CTL_3 0x1403 +#define RT5682S_NEW_CBJ_DET_CTL_4 0x1404 +#define RT5682S_NEW_CBJ_DET_CTL_5 0x1406 +#define RT5682S_NEW_CBJ_DET_CTL_6 0x1407 +#define RT5682S_NEW_CBJ_DET_CTL_7 0x1408 +#define RT5682S_NEW_CBJ_DET_CTL_8 0x1409 +#define RT5682S_NEW_CBJ_DET_CTL_9 0x140a +#define RT5682S_NEW_CBJ_DET_CTL_10 0x140b +#define RT5682S_NEW_CBJ_DET_CTL_11 0x140c +#define RT5682S_NEW_CBJ_DET_CTL_12 0x140d +#define RT5682S_NEW_CBJ_DET_CTL_13 0x140e +#define RT5682S_NEW_CBJ_DET_CTL_14 0x140f +#define RT5682S_NEW_CBJ_DET_CTL_15 0x1410 +#define RT5682S_NEW_CBJ_DET_CTL_16 0x1411 +#define RT5682S_DA_FILTER_1 0x1801 +#define RT5682S_DA_FILTER_2 0x1802 +#define RT5682S_DA_FILTER_3 0x1803 +#define RT5682S_DA_FILTER_4 0x1804 +#define RT5682S_DA_FILTER_5 0x1805 +#define RT5682S_CLK_SW_TEST_1 0x2c00 +#define RT5682S_CLK_SW_TEST_2 0x3400 +#define RT5682S_CLK_SW_TEST_3 0x3404 +#define RT5682S_CLK_SW_TEST_4 0x3405 +#define RT5682S_CLK_SW_TEST_5 0x3406 +#define RT5682S_CLK_SW_TEST_6 0x3407 +#define RT5682S_CLK_SW_TEST_7 0x3408 +#define RT5682S_CLK_SW_TEST_8 0x3409 +#define RT5682S_CLK_SW_TEST_9 0x340a +#define RT5682S_CLK_SW_TEST_10 0x340b +#define RT5682S_CLK_SW_TEST_11 0x340c +#define RT5682S_CLK_SW_TEST_12 0x340d +#define RT5682S_CLK_SW_TEST_13 0x340e +#define RT5682S_CLK_SW_TEST_14 0x340f +#define RT5682S_EFUSE_MANU_WRITE_1 0x3410 +#define RT5682S_EFUSE_MANU_WRITE_2 0x3411 +#define RT5682S_EFUSE_MANU_WRITE_3 0x3412 +#define RT5682S_EFUSE_MANU_WRITE_4 0x3413 +#define RT5682S_EFUSE_MANU_WRITE_5 0x3414 +#define RT5682S_EFUSE_MANU_WRITE_6 0x3415 +#define RT5682S_EFUSE_READ_1 0x3424 +#define RT5682S_EFUSE_READ_2 0x3425 +#define RT5682S_EFUSE_READ_3 0x3426 +#define RT5682S_EFUSE_READ_4 0x3427 +#define RT5682S_EFUSE_READ_5 0x3428 +#define RT5682S_EFUSE_READ_6 0x3429 +#define RT5682S_EFUSE_READ_7 0x342a +#define RT5682S_EFUSE_READ_8 0x342b +#define RT5682S_EFUSE_READ_9 0x342c +#define RT5682S_EFUSE_READ_10 0x342d +#define RT5682S_EFUSE_READ_11 0x342e +#define RT5682S_EFUSE_READ_12 0x342f +#define RT5682S_EFUSE_READ_13 0x3430 +#define RT5682S_EFUSE_READ_14 0x3431 +#define RT5682S_EFUSE_READ_15 0x3432 +#define RT5682S_EFUSE_READ_16 0x3433 +#define RT5682S_EFUSE_READ_17 0x3434 +#define RT5682S_EFUSE_READ_18 0x3435 +#define RT5682S_EFUSE_TIMING_CTL_1 0x3440 +#define RT5682S_EFUSE_TIMING_CTL_2 0x3441 +#define RT5682S_PILOT_DIG_CTL_1 0x3500 +#define RT5682S_PILOT_DIG_CTL_2 0x3501 +#define RT5682S_HP_AMP_DET_CTL_1 0x3b00 +#define RT5682S_HP_AMP_DET_CTL_2 0x3b01 +#define RT5682S_HP_AMP_DET_CTL_3 0x3b02 +#define RT5682S_HP_AMP_DET_CTL_4 0x3b03 + +#define RT5682S_MAX_REG (RT5682S_HP_AMP_DET_CTL_4) + +/* global definition */ +#define RT5682S_L_MUTE (0x1 << 15) +#define RT5682S_L_MUTE_SFT 15 +#define RT5682S_R_MUTE (0x1 << 7) +#define RT5682S_R_MUTE_SFT 7 +#define RT5682S_L_VOL_SFT 8 +#define RT5682S_R_VOL_SFT 0 +#define RT5682S_CLK_SRC_MCLK (0x0) +#define RT5682S_CLK_SRC_PLL1 (0x1) +#define RT5682S_CLK_SRC_PLL2 (0x2) +#define RT5682S_CLK_SRC_RCCLK (0x4) /* 25M */ + + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5682S_HPO_L_PATH_MASK (0x1 << 14) +#define RT5682S_HPO_L_PATH_EN (0x1 << 14) +#define RT5682S_HPO_L_PATH_DIS (0x0 << 14) +#define RT5682S_HPO_R_PATH_MASK (0x1 << 13) +#define RT5682S_HPO_R_PATH_EN (0x1 << 13) +#define RT5682S_HPO_R_PATH_DIS (0x0 << 13) +#define RT5682S_HPO_SEL_IP_EN_SW (0x1) +#define RT5682S_HPO_IP_EN_GATING (0x1) +#define RT5682S_HPO_IP_NO_GATING (0x0) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682S_G_HP (0xf << 8) +#define RT5682S_G_HP_SFT 8 +#define RT5682S_G_STO_DA_DMIX (0xf) +#define RT5682S_G_STO_DA_SFT 0 + +/* Embeeded Jack and Type Detection Control 2 (0x0010) */ +#define RT5682S_EMB_JD_MASK (0x1 << 15) +#define RT5682S_EMB_JD_EN (0x1 << 15) +#define RT5682S_EMB_JD_EN_SFT 15 +#define RT5682S_EMB_JD_RST (0x1 << 14) +#define RT5682S_JD_MODE (0x1 << 13) +#define RT5682S_JD_MODE_SFT 13 +#define RT5682S_DET_TYPE (0x1 << 12) +#define RT5682S_DET_TYPE_SFT 12 +#define RT5682S_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682S_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682S_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682S_SEL_FAST_OFF_MASK (0x3 << 9) +#define RT5682S_SEL_FAST_OFF_SFT 9 +#define RT5682S_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682S_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682S_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682S_FAST_OFF_MASK (0x1 << 7) +#define RT5682S_FAST_OFF_EN (0x1 << 7) +#define RT5682S_FAST_OFF_DIS (0x0 << 7) +#define RT5682S_VREF_POW_MASK (0x1 << 6) +#define RT5682S_VREF_POW_FSM (0x0 << 6) +#define RT5682S_VREF_POW_REG (0x1 << 6) +#define RT5682S_MB1_PATH_BIT 5 +#define RT5682S_MB1_PATH_MASK (0x1 << 5) +#define RT5682S_CTRL_MB1_REG (0x1 << 5) +#define RT5682S_CTRL_MB1_FSM (0x0 << 5) +#define RT5682S_MB2_PATH_BIT 4 +#define RT5682S_MB2_PATH_MASK (0x1 << 4) +#define RT5682S_CTRL_MB2_REG (0x1 << 4) +#define RT5682S_CTRL_MB2_FSM (0x0 << 4) +#define RT5682S_TRIG_JD_MASK (0x1 << 3) +#define RT5682S_TRIG_JD_HIGH (0x1 << 3) +#define RT5682S_TRIG_JD_LOW (0x0 << 3) +#define RT5682S_MIC_CAP_MASK (0x1 << 1) +#define RT5682S_MIC_CAP_HS (0x1 << 1) +#define RT5682S_MIC_CAP_HP (0x0 << 1) +#define RT5682S_MIC_CAP_SRC_MASK (0x1) +#define RT5682S_MIC_CAP_SRC_REG (0x1) +#define RT5682S_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 3 (0x0011) */ +#define RT5682S_SEL_CBJ_TYPE_SLOW (0x1 << 15) +#define RT5682S_SEL_CBJ_TYPE_NORM (0x0 << 15) +#define RT5682S_SEL_CBJ_TYPE_MASK (0x1 << 15) +#define RT5682S_POW_BG_MB1_MASK (0x1 << 13) +#define RT5682S_POW_BG_MB1_REG (0x1 << 13) +#define RT5682S_POW_BG_MB1_FSM (0x0 << 13) +#define RT5682S_POW_BG_MB2_MASK (0x1 << 12) +#define RT5682S_POW_BG_MB2_REG (0x1 << 12) +#define RT5682S_POW_BG_MB2_FSM (0x0 << 12) +#define RT5682S_EXT_JD_SRC (0x7 << 4) +#define RT5682S_EXT_JD_SRC_SFT 4 +#define RT5682S_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682S_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682S_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682S_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682S_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682S_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 4 (0x0012) */ +#define RT5682S_CBJ_IN_BUF_MASK (0x1 << 7) +#define RT5682S_CBJ_IN_BUF_EN (0x1 << 7) +#define RT5682S_CBJ_IN_BUF_DIS (0x0 << 7) +#define RT5682S_CBJ_IN_BUF_BIT 7 + +/* Combo Jack and Type Detection Control 5 (0x0013) */ +#define RT5682S_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682S_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682S_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682S_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682S_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682S_CBJ_JD_TEST_MODE (0x1 << 6) + +/* Combo Jack and Type Detection Control 6 (0x0014) */ +#define RT5682S_JD_FAST_OFF_SRC_MASK (0x7 << 8) +#define RT5682S_JD_FAST_OFF_SRC_JDH (0x6 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO6 (0x5 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO5 (0x4 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO4 (0x3 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO3 (0x2 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO2 (0x1 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO1 (0x0 << 8) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682S_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682S_DAC_L1_VOL_SFT 8 +#define RT5682S_DAC_R1_VOL_MASK (0xff) +#define RT5682S_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682S_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682S_ADC_L_VOL_SFT 8 +#define RT5682S_ADC_R_VOL_MASK (0x7f) +#define RT5682S_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682S_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682S_STO1_ADC_L_BST_SFT 14 +#define RT5682S_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682S_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682S_ST_SRC_SEL (0x1 << 8) +#define RT5682S_ST_SRC_SFT 8 +#define RT5682S_ST_EN_MASK (0x1 << 6) +#define RT5682S_ST_DIS (0x0 << 6) +#define RT5682S_ST_EN (0x1 << 6) +#define RT5682S_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682S_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682S_M_STO1_ADC_L1_SFT 15 +#define RT5682S_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682S_M_STO1_ADC_L2_SFT 14 +#define RT5682S_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682S_STO1_ADC1L_SRC_SFT 13 +#define RT5682S_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682S_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682S_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682S_STO1_ADC2L_SRC_SFT 12 +#define RT5682S_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682S_STO1_ADCL_SRC_SFT 10 +#define RT5682S_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682S_M_STO1_ADC_R1_SFT 7 +#define RT5682S_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682S_M_STO1_ADC_R2_SFT 6 +#define RT5682S_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682S_STO1_ADC1R_SRC_SFT 5 +#define RT5682S_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682S_STO1_ADC2R_SRC_SFT 4 +#define RT5682S_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682S_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682S_M_ADCMIX_L (0x1 << 15) +#define RT5682S_M_ADCMIX_L_SFT 15 +#define RT5682S_M_DAC1_L (0x1 << 14) +#define RT5682S_M_DAC1_L_SFT 14 +#define RT5682S_M_ADCMIX_R (0x1 << 7) +#define RT5682S_M_ADCMIX_R_SFT 7 +#define RT5682S_M_DAC1_R (0x1 << 6) +#define RT5682S_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682S_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682S_M_DAC_L1_STO_L_SFT 15 +#define RT5682S_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682S_G_DAC_L1_STO_L_SFT 14 +#define RT5682S_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682S_M_DAC_R1_STO_L_SFT 13 +#define RT5682S_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682S_G_DAC_R1_STO_L_SFT 12 +#define RT5682S_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682S_M_DAC_L1_STO_R_SFT 7 +#define RT5682S_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682S_G_DAC_L1_STO_R_SFT 6 +#define RT5682S_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682S_M_DAC_R1_STO_R_SFT 5 +#define RT5682S_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682S_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682S_M_ST_STO_L (0x1 << 9) +#define RT5682S_M_ST_STO_L_SFT 9 +#define RT5682S_M_ST_STO_R (0x1 << 8) +#define RT5682S_M_ST_STO_R_SFT 8 +#define RT5682S_DAC_L1_SRC_MASK (0x1 << 4) +#define RT5682S_A_DACL1_SFT 4 +#define RT5682S_DAC_R1_SRC_MASK (0x1) +#define RT5682S_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682S_IF2_DAC_SEL_MASK (0x3 << 2) +#define RT5682S_IF2_DAC_SEL_SFT 2 +#define RT5682S_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682S_IF2_ADC_SEL_SFT 0 + +/* REC Left/Right Mixer Control 2 (0x003c) */ +#define RT5682S_BST_CBJ_MASK (0x3f << 8) +#define RT5682S_BST_CBJ_SFT 8 +#define RT5682S_M_CBJ_RM1_L (0x1 << 7) +#define RT5682S_M_CBJ_RM1_L_SFT 7 +#define RT5682S_M_CBJ_RM1_R (0x1 << 6) +#define RT5682S_M_CBJ_RM1_R_SFT 6 + +/* REC Left/Right Mixer Calibration Control(0x0044) */ +#define RT5682S_PWR_RM1_R_BIT 8 +#define RT5682S_PWR_RM1_L_BIT 0 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682S_PWR_I2S1 (0x1 << 15) +#define RT5682S_PWR_I2S1_BIT 15 +#define RT5682S_PWR_I2S2 (0x1 << 14) +#define RT5682S_PWR_I2S2_BIT 14 +#define RT5682S_PRE_CHR_DAC_L1 (0x1 << 13) +#define RT5682S_PRE_CHR_DAC_L1_BIT 13 +#define RT5682S_PRE_CHR_DAC_R1 (0x1 << 12) +#define RT5682S_PRE_CHR_DAC_R1_BIT 12 +#define RT5682S_PWR_DAC_L1 (0x1 << 11) +#define RT5682S_PWR_DAC_L1_BIT 11 +#define RT5682S_PWR_DAC_R1 (0x1 << 10) +#define RT5682S_PWR_DAC_R1_BIT 10 +#define RT5682S_PWR_LDO (0x1 << 8) +#define RT5682S_PWR_LDO_BIT 8 +#define RT5682S_PWR_D2S_L (0x1 << 7) +#define RT5682S_PWR_D2S_L_BIT 7 +#define RT5682S_PWR_D2S_R (0x1 << 6) +#define RT5682S_PWR_D2S_R_BIT 6 +#define RT5682S_PWR_ADC_L1 (0x1 << 4) +#define RT5682S_PWR_ADC_L1_BIT 4 +#define RT5682S_PWR_ADC_R1 (0x1 << 3) +#define RT5682S_PWR_ADC_R1_BIT 3 +#define RT5682S_EFUSE_SW_EN (0x1 << 2) +#define RT5682S_EFUSE_SW_DIS (0x0 << 2) +#define RT5682S_PWR_EFUSE (0x1 << 1) +#define RT5682S_PWR_EFUSE_BIT 1 +#define RT5682S_DIG_GATE_CTRL (0x1 << 0) +#define RT5682S_DIG_GATE_CTRL_SFT 0 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682S_PWR_ADC_S1F (0x1 << 15) +#define RT5682S_PWR_ADC_S1F_BIT 15 +#define RT5682S_PWR_DAC_S1F (0x1 << 10) +#define RT5682S_PWR_DAC_S1F_BIT 10 +#define RT5682S_DLDO_I_LIMIT_MASK (0x1 << 7) +#define RT5682S_DLDO_I_LIMIT_EN (0x1 << 7) +#define RT5682S_DLDO_I_LIMIT_DIS (0x0 << 7) +#define RT5682S_DLDO_I_BIAS_SEL_4 (0x1 << 6) +#define RT5682S_DLDO_I_BIAS_SEL_0 (0x0 << 6) +#define RT5682S_DLDO_REG_TEST_1 (0x1 << 5) +#define RT5682S_DLDO_REG_TEST_0 (0x0 << 5) +#define RT5682S_DLDO_SRC_REG (0x1 << 4) +#define RT5682S_DLDO_SRC_EFUSE (0x0 << 4) + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682S_PWR_VREF1 (0x1 << 15) +#define RT5682S_PWR_VREF1_BIT 15 +#define RT5682S_PWR_FV1 (0x1 << 14) +#define RT5682S_PWR_FV1_BIT 14 +#define RT5682S_PWR_VREF2 (0x1 << 13) +#define RT5682S_PWR_VREF2_BIT 13 +#define RT5682S_PWR_FV2 (0x1 << 12) +#define RT5682S_PWR_FV2_BIT 12 +#define RT5682S_LDO1_DBG_MASK (0x3 << 10) +#define RT5682S_PWR_MB (0x1 << 9) +#define RT5682S_PWR_MB_BIT 9 +#define RT5682S_PWR_BG (0x1 << 7) +#define RT5682S_PWR_BG_BIT 7 +#define RT5682S_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682S_LDO1_BYPASS (0x1 << 6) +#define RT5682S_LDO1_NOT_BYPASS (0x0 << 6) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682S_PWR_MCLK0_WD (0x1 << 15) +#define RT5682S_PWR_MCLK0_WD_BIT 15 +#define RT5682S_PWR_MCLK1_WD (0x1 << 14) +#define RT5682S_PWR_MCLK1_WD_BIT 14 +#define RT5682S_RST_MCLK0 (0x1 << 13) +#define RT5682S_RST_MCLK0_BIT 13 +#define RT5682S_RST_MCLK1 (0x1 << 12) +#define RT5682S_RST_MCLK1_BIT 12 +#define RT5682S_PWR_MB1 (0x1 << 11) +#define RT5682S_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682S_PWR_MB1_BIT 11 +#define RT5682S_PWR_MB2 (0x1 << 10) +#define RT5682S_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682S_PWR_MB2_BIT 10 +#define RT5682S_PWR_JD_MASK (0x1 << 0) +#define RT5682S_PWR_JD_ENABLE (0x1 << 0) +#define RT5682S_PWR_JD_DISABLE (0x0 << 0) + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682S_PWR_LDO_PLLA (0x1 << 15) +#define RT5682S_PWR_LDO_PLLA_BIT 15 +#define RT5682S_PWR_LDO_PLLB (0x1 << 14) +#define RT5682S_PWR_LDO_PLLB_BIT 14 +#define RT5682S_PWR_BIAS_PLLA (0x1 << 13) +#define RT5682S_PWR_BIAS_PLLA_BIT 13 +#define RT5682S_PWR_BIAS_PLLB (0x1 << 12) +#define RT5682S_PWR_BIAS_PLLB_BIT 12 +#define RT5682S_PWR_CBJ (0x1 << 9) +#define RT5682S_PWR_CBJ_BIT 9 +#define RT5682S_RSTB_PLLB (0x1 << 7) +#define RT5682S_RSTB_PLLB_BIT 7 +#define RT5682S_RSTB_PLLA (0x1 << 6) +#define RT5682S_RSTB_PLLA_BIT 6 +#define RT5682S_PWR_PLLB (0x1 << 5) +#define RT5682S_PWR_PLLB_BIT 5 +#define RT5682S_PWR_PLLA (0x1 << 4) +#define RT5682S_PWR_PLLA_BIT 4 +#define RT5682S_PWR_LDO_MB2 (0x1 << 2) +#define RT5682S_PWR_LDO_MB2_BIT 2 +#define RT5682S_PWR_LDO_MB1 (0x1 << 1) +#define RT5682S_PWR_LDO_MB1_BIT 1 +#define RT5682S_PWR_BGLDO (0x1 << 0) +#define RT5682S_PWR_BGLDO_BIT 0 + +/* Power Management for Mixer (0x0066) */ +#define RT5682S_PWR_CLK_COMP_8FS (0x1 << 15) +#define RT5682S_PWR_CLK_COMP_8FS_BIT 15 +#define RT5682S_DBG_BGLDO_MASK (0x3 << 12) +#define RT5682S_DBG_BGLDO_SFT 12 +#define RT5682S_DBG_BGLDO_MB1_MASK (0x3 << 10) +#define RT5682S_DBG_BGLDO_MB1_SFT 10 +#define RT5682S_DBG_BGLDO_MB2_MASK (0x3 << 8) +#define RT5682S_DBG_BGLDO_MB2_SFT 8 +#define RT5682S_DLDO_BGLDO_MASK (0x3 << 6) +#define RT5682S_DLDO_BGLDO_MB2_SFT 6 +#define RT5682S_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682S_PWR_STO1_DAC_L_BIT 5 +#define RT5682S_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682S_PWR_STO1_DAC_R_BIT 4 +#define RT5682S_DVO_BGLDO_MB1_MASK (0x3 << 2) +#define RT5682S_DVO_BGLDO_MB1_SFT 2 +#define RT5682S_DVO_BGLDO_MB2_MASK (0x3 << 0) + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682S_SYS_CLK_DET (0x1 << 15) +#define RT5682S_SYS_CLK_DET_SFT 15 +#define RT5682S_PLL1_CLK_DET (0x1 << 14) +#define RT5682S_PLL1_CLK_DET_SFT 14 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682S_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682S_DMIC_1_EN_SFT 15 +#define RT5682S_DMIC_1_DIS (0x0 << 15) +#define RT5682S_DMIC_1_EN (0x1 << 15) +#define RT5682S_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682S_FIFO_CLK_DIV_2 (0x1 << 12) +#define RT5682S_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682S_DMIC_1_DP_SFT 4 +#define RT5682S_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682S_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682S_DMIC_CLK_MASK (0xf << 0) +#define RT5682S_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682S_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682S_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682S_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682S_SEL_ADCDAT_SFT 15 +#define RT5682S_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682S_I2S1_TX_CHL_SFT 12 +#define RT5682S_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682S_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682S_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682S_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682S_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682S_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682S_I2S1_RX_CHL_SFT 8 +#define RT5682S_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682S_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682S_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682S_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682S_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682S_I2S1_MONO_MASK (0x1 << 7) +#define RT5682S_I2S1_MONO_EN (0x1 << 7) +#define RT5682S_I2S1_MONO_DIS (0x0 << 7) +#define RT5682S_I2S1_DL_MASK (0x7 << 4) +#define RT5682S_I2S1_DL_SFT 4 +#define RT5682S_I2S1_DL_16 (0x0 << 4) +#define RT5682S_I2S1_DL_20 (0x1 << 4) +#define RT5682S_I2S1_DL_24 (0x2 << 4) +#define RT5682S_I2S1_DL_32 (0x3 << 4) +#define RT5682S_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0071) */ +#define RT5682S_I2S2_MS_MASK (0x1 << 15) +#define RT5682S_I2S2_MS_SFT 15 +#define RT5682S_I2S2_MS_M (0x0 << 15) +#define RT5682S_I2S2_MS_S (0x1 << 15) +#define RT5682S_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682S_I2S2_PIN_CFG_SFT 14 +#define RT5682S_I2S2_OUT_MASK (0x1 << 9) +#define RT5682S_I2S2_OUT_SFT 9 +#define RT5682S_I2S2_OUT_UM (0x0 << 9) +#define RT5682S_I2S2_OUT_M (0x1 << 9) +#define RT5682S_I2S_BP_MASK (0x1 << 8) +#define RT5682S_I2S_BP_SFT 8 +#define RT5682S_I2S_BP_NOR (0x0 << 8) +#define RT5682S_I2S_BP_INV (0x1 << 8) +#define RT5682S_I2S2_MONO_MASK (0x1 << 7) +#define RT5682S_I2S2_MONO_EN (0x1 << 7) +#define RT5682S_I2S2_MONO_DIS (0x0 << 7) +#define RT5682S_I2S2_DL_MASK (0x7 << 4) +#define RT5682S_I2S2_DL_SFT 4 +#define RT5682S_I2S2_DL_8 (0x0 << 4) +#define RT5682S_I2S2_DL_16 (0x1 << 4) +#define RT5682S_I2S2_DL_20 (0x2 << 4) +#define RT5682S_I2S2_DL_24 (0x3 << 4) +#define RT5682S_I2S2_DL_32 (0x4 << 4) +#define RT5682S_I2S_DF_MASK (0x7) +#define RT5682S_I2S_DF_SFT 0 +#define RT5682S_I2S_DF_I2S (0x0) +#define RT5682S_I2S_DF_LEFT (0x1) +#define RT5682S_I2S_DF_PCM_A (0x2) +#define RT5682S_I2S_DF_PCM_B (0x3) +#define RT5682S_I2S_DF_PCM_A_N (0x6) +#define RT5682S_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682S_ADC_OSR_MASK (0xf << 12) +#define RT5682S_ADC_OSR_SFT 12 +#define RT5682S_ADC_OSR_D_1 (0x0 << 12) +#define RT5682S_ADC_OSR_D_2 (0x1 << 12) +#define RT5682S_ADC_OSR_D_4 (0x2 << 12) +#define RT5682S_ADC_OSR_D_6 (0x3 << 12) +#define RT5682S_ADC_OSR_D_8 (0x4 << 12) +#define RT5682S_ADC_OSR_D_12 (0x5 << 12) +#define RT5682S_ADC_OSR_D_16 (0x6 << 12) +#define RT5682S_ADC_OSR_D_24 (0x7 << 12) +#define RT5682S_ADC_OSR_D_32 (0x8 << 12) +#define RT5682S_ADC_OSR_D_48 (0x9 << 12) +#define RT5682S_I2S_M_D_MASK (0xf << 8) +#define RT5682S_I2S_M_D_SFT 8 +#define RT5682S_I2S_M_D_1 (0x0 << 8) +#define RT5682S_I2S_M_D_2 (0x1 << 8) +#define RT5682S_I2S_M_D_3 (0x2 << 8) +#define RT5682S_I2S_M_D_4 (0x3 << 8) +#define RT5682S_I2S_M_D_6 (0x4 << 8) +#define RT5682S_I2S_M_D_8 (0x5 << 8) +#define RT5682S_I2S_M_D_12 (0x6 << 8) +#define RT5682S_I2S_M_D_16 (0x7 << 8) +#define RT5682S_I2S_M_D_24 (0x8 << 8) +#define RT5682S_I2S_M_D_32 (0x9 << 8) +#define RT5682S_I2S_M_D_48 (0x10 << 8) +#define RT5682S_I2S_M_CLK_SRC_MASK (0x7 << 4) +#define RT5682S_I2S_M_CLK_SRC_SFT 4 +#define RT5682S_DAC_OSR_MASK (0xf << 0) +#define RT5682S_DAC_OSR_SFT 0 +#define RT5682S_DAC_OSR_D_1 (0x0 << 0) +#define RT5682S_DAC_OSR_D_2 (0x1 << 0) +#define RT5682S_DAC_OSR_D_4 (0x2 << 0) +#define RT5682S_DAC_OSR_D_6 (0x3 << 0) +#define RT5682S_DAC_OSR_D_8 (0x4 << 0) +#define RT5682S_DAC_OSR_D_12 (0x5 << 0) +#define RT5682S_DAC_OSR_D_16 (0x6 << 0) +#define RT5682S_DAC_OSR_D_24 (0x7 << 0) +#define RT5682S_DAC_OSR_D_32 (0x8 << 0) +#define RT5682S_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682S_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682S_I2S2_BCLK_MS2_SFT 11 +#define RT5682S_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682S_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682S_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682S_TDM_TX_CH_2 (0x0 << 12) +#define RT5682S_TDM_TX_CH_4 (0x1 << 12) +#define RT5682S_TDM_TX_CH_6 (0x2 << 12) +#define RT5682S_TDM_TX_CH_8 (0x3 << 12) +#define RT5682S_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682S_TDM_RX_CH_2 (0x0 << 8) +#define RT5682S_TDM_RX_CH_4 (0x1 << 8) +#define RT5682S_TDM_RX_CH_6 (0x2 << 8) +#define RT5682S_TDM_RX_CH_8 (0x3 << 8) +#define RT5682S_TDM_ADC_LCA_MASK (0x7 << 4) +#define RT5682S_TDM_ADC_LCA_SFT 4 +#define RT5682S_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682S_IF1_ADC1_SEL_SFT 14 +#define RT5682S_IF1_ADC2_SEL_SFT 12 +#define RT5682S_IF1_ADC3_SEL_SFT 10 +#define RT5682S_IF1_ADC4_SEL_SFT 8 +#define RT5682S_TDM_ADC_SEL_SFT 3 + +/* TDM control 3 (0x007b) */ +#define RT5682S_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682S_TDM_S_BP_MASK (0x1 << 15) +#define RT5682S_TDM_S_BP_SFT 15 +#define RT5682S_TDM_S_BP_NOR (0x0 << 15) +#define RT5682S_TDM_S_BP_INV (0x1 << 15) +#define RT5682S_TDM_S_LP_MASK (0x1 << 14) +#define RT5682S_TDM_S_LP_SFT 14 +#define RT5682S_TDM_S_LP_NOR (0x0 << 14) +#define RT5682S_TDM_S_LP_INV (0x1 << 14) +#define RT5682S_TDM_DF_MASK (0x7 << 11) +#define RT5682S_TDM_DF_SFT 11 +#define RT5682S_TDM_DF_I2S (0x0 << 11) +#define RT5682S_TDM_DF_LEFT (0x1 << 11) +#define RT5682S_TDM_DF_PCM_A (0x2 << 11) +#define RT5682S_TDM_DF_PCM_B (0x3 << 11) +#define RT5682S_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682S_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682S_TDM_BCLK_MS1_MASK (0x3 << 8) +#define RT5682S_TDM_BCLK_MS1_SFT 8 +#define RT5682S_TDM_BCLK_MS1_32 (0x0 << 8) +#define RT5682S_TDM_BCLK_MS1_64 (0x1 << 8) +#define RT5682S_TDM_BCLK_MS1_128 (0x2 << 8) +#define RT5682S_TDM_BCLK_MS1_256 (0x3 << 8) +#define RT5682S_TDM_BCLK_MS1_16 (0x4 << 8) +#define RT5682S_TDM_CL_MASK (0x3 << 4) +#define RT5682S_TDM_CL_16 (0x0 << 4) +#define RT5682S_TDM_CL_20 (0x1 << 4) +#define RT5682S_TDM_CL_24 (0x2 << 4) +#define RT5682S_TDM_CL_32 (0x3 << 4) +#define RT5682S_TDM_M_BP_MASK (0x1 << 2) +#define RT5682S_TDM_M_BP_SFT 2 +#define RT5682S_TDM_M_BP_NOR (0x0 << 2) +#define RT5682S_TDM_M_BP_INV (0x1 << 2) +#define RT5682S_TDM_M_LP_MASK (0x1 << 1) +#define RT5682S_TDM_M_LP_SFT 1 +#define RT5682S_TDM_M_LP_NOR (0x0 << 1) +#define RT5682S_TDM_M_LP_INV (0x1 << 1) +#define RT5682S_TDM_MS_MASK (0x1 << 0) +#define RT5682S_TDM_MS_SFT 0 +#define RT5682S_TDM_MS_S (0x0 << 0) +#define RT5682S_TDM_MS_M (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682S_SCLK_SRC_MASK (0x7 << 13) +#define RT5682S_SCLK_SRC_SFT 13 +#define RT5682S_PLL_SRC_MASK (0x3 << 8) +#define RT5682S_PLL_SRC_SFT 8 +#define RT5682S_PLL_SRC_MCLK (0x0 << 8) +#define RT5682S_PLL_SRC_BCLK1 (0x1 << 8) +#define RT5682S_PLL_SRC_RC (0x3 << 8) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682S_DA_ASRC_MASK (0x1 << 13) +#define RT5682S_DA_ASRC_SFT 13 +#define RT5682S_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682S_DAC_STO1_ASRC_SFT 12 +#define RT5682S_AD_ASRC_MASK (0x1 << 8) +#define RT5682S_AD_ASRC_SFT 8 +#define RT5682S_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682S_AD_ASRC_SEL_SFT 4 +#define RT5682S_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682S_DMIC_ASRC_SFT 3 +#define RT5682S_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682S_ADC_STO1_ASRC_SFT 2 +#define RT5682S_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682S_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682S_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682S_FILTER_CLK_SEL_SFT 12 +#define RT5682S_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682S_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682S_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682S_ASRCIN_FTK_N1_SFT 14 +#define RT5682S_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682S_ASRCIN_FTK_N2_SFT 12 +#define RT5682S_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682S_ASRCIN_FTK_M1_SFT 8 +#define RT5682S_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682S_ASRCIN_FTK_M2_SFT 4 + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682S_OUT_HP_L_EN (0x1 << 6) +#define RT5682S_OUT_HP_R_EN (0x1 << 5) +#define RT5682S_LDO_PUMP_EN (0x1 << 4) +#define RT5682S_LDO_PUMP_EN_SFT 4 +#define RT5682S_PUMP_EN (0x1 << 3) +#define RT5682S_PUMP_EN_SFT 3 +#define RT5682S_CAPLESS_L_EN (0x1 << 1) +#define RT5682S_CAPLESS_L_EN_SFT 1 +#define RT5682S_CAPLESS_R_EN (0x1 << 0) +#define RT5682S_CAPLESS_R_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682S_RAMP_MASK (0x1 << 12) +#define RT5682S_RAMP_SFT 12 +#define RT5682S_RAMP_DIS (0x0 << 12) +#define RT5682S_RAMP_EN (0x1 << 12) +#define RT5682S_BPS_MASK (0x1 << 11) +#define RT5682S_BPS_SFT 11 +#define RT5682S_BPS_DIS (0x0 << 11) +#define RT5682S_BPS_EN (0x1 << 11) +#define RT5682S_FAST_UPDN_MASK (0x1 << 10) +#define RT5682S_FAST_UPDN_SFT 10 +#define RT5682S_FAST_UPDN_DIS (0x0 << 10) +#define RT5682S_FAST_UPDN_EN (0x1 << 10) +#define RT5682S_VLO_MASK (0x1 << 7) +#define RT5682S_VLO_SFT 7 +#define RT5682S_VLO_3V (0x0 << 7) +#define RT5682S_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682S_OSW_L_MASK (0x1 << 11) +#define RT5682S_OSW_L_SFT 11 +#define RT5682S_OSW_L_DIS (0x0 << 11) +#define RT5682S_OSW_L_EN (0x1 << 11) +#define RT5682S_OSW_R_MASK (0x1 << 10) +#define RT5682S_OSW_R_SFT 10 +#define RT5682S_OSW_R_DIS (0x0 << 10) +#define RT5682S_OSW_R_EN (0x1 << 10) +#define RT5682S_PM_HP_MASK (0x3 << 8) +#define RT5682S_PM_HP_SFT 8 +#define RT5682S_PM_HP_LV (0x0 << 8) +#define RT5682S_PM_HP_MV (0x1 << 8) +#define RT5682S_PM_HP_HV (0x2 << 8) + +/* Micbias Control1 (0x93) */ +#define RT5682S_MIC1_OV_MASK (0x3 << 14) +#define RT5682S_MIC1_OV_SFT 14 +#define RT5682S_MIC1_OV_2V7 (0x0 << 14) +#define RT5682S_MIC1_OV_2V4 (0x1 << 14) +#define RT5682S_MIC1_OV_2V25 (0x3 << 14) +#define RT5682S_MIC1_OV_1V8 (0x4 << 14) +#define RT5682S_MIC2_OV_MASK (0x3 << 8) +#define RT5682S_MIC2_OV_SFT 8 +#define RT5682S_MIC2_OV_2V7 (0x0 << 8) +#define RT5682S_MIC2_OV_2V4 (0x1 << 8) +#define RT5682S_MIC2_OV_2V25 (0x3 << 8) +#define RT5682S_MIC2_OV_1V8 (0x4 << 8) + +/* Micbias Control2 (0x0094) */ +#define RT5682S_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682S_PWR_CLK25M_SFT 9 +#define RT5682S_PWR_CLK25M_PD (0x0 << 9) +#define RT5682S_PWR_CLK25M_PU (0x1 << 9) +#define RT5682S_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682S_PWR_CLK1M_SFT 8 +#define RT5682S_PWR_CLK1M_PD (0x0 << 8) +#define RT5682S_PWR_CLK1M_PU (0x1 << 8) + +/* PLL M/N/K Code Control 1 (0x0098) */ +#define RT5682S_PLLA_N_MASK (0x1ff << 0) + +/* PLL M/N/K Code Control 2 (0x0099) */ +#define RT5682S_PLLA_M_MASK (0x1f << 8) +#define RT5682S_PLLA_M_SFT 8 +#define RT5682S_PLLA_K_MASK (0x1f << 0) + +/* PLL M/N/K Code Control 3 (0x009a) */ +#define RT5682S_PLLB_N_MASK (0x3ff << 0) + +/* PLL M/N/K Code Control 4 (0x009b) */ +#define RT5682S_PLLB_M_MASK (0x1f << 8) +#define RT5682S_PLLB_M_SFT 8 +#define RT5682S_PLLB_K_MASK (0x1f << 0) + +/* PLL M/N/K Code Control 6 (0x009d) */ +#define RT5682S_PLLB_SEL_PS_MASK (0x1 << 13) +#define RT5682S_PLLB_SEL_PS_SFT 13 +#define RT5682S_PLLB_BYP_PS_MASK (0x1 << 12) +#define RT5682S_PLLB_BYP_PS_SFT 12 +#define RT5682S_PLLB_M_BP_MASK (0x1 << 11) +#define RT5682S_PLLB_M_BP_SFT 11 +#define RT5682S_PLLB_K_BP_MASK (0x1 << 10) +#define RT5682S_PLLB_K_BP_SFT 10 +#define RT5682S_PLLA_M_BP_MASK (0x1 << 7) +#define RT5682S_PLLA_M_BP_SFT 7 +#define RT5682S_PLLA_K_BP_MASK (0x1 << 6) +#define RT5682S_PLLA_K_BP_SFT 6 + +/* PLL M/N/K Code Control 7 (0x009e) */ +#define RT5682S_PLLB_SRC_MASK (0x3 << 0) +#define RT5682S_PLLB_SRC_DFIN (0x1) +#define RT5682S_PLLB_SRC_PLLA (0x0) + +/* RC Clock Control (0x009f) */ +#define RT5682S_POW_IRQ (0x1 << 15) +#define RT5682S_POW_JDH (0x1 << 14) + +/* I2S2 Master Mode Clock Control 1 (0x00a0) */ +#define RT5682S_I2S2_M_CLK_SRC_MASK (0x7 << 4) +#define RT5682S_I2S2_M_CLK_SRC_SFT 4 +#define RT5682S_I2S2_M_D_MASK (0xf << 0) +#define RT5682S_I2S2_M_D_1 (0x0) +#define RT5682S_I2S2_M_D_2 (0x1) +#define RT5682S_I2S2_M_D_3 (0x2) +#define RT5682S_I2S2_M_D_4 (0x3) +#define RT5682S_I2S2_M_D_6 (0x4) +#define RT5682S_I2S2_M_D_8 (0x5) +#define RT5682S_I2S2_M_D_12 (0x6) +#define RT5682S_I2S2_M_D_16 (0x7) +#define RT5682S_I2S2_M_D_24 (0x8) +#define RT5682S_I2S2_M_D_32 (0x9) +#define RT5682S_I2S2_M_D_48 (0xa) +#define RT5682S_I2S2_M_D_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682S_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682S_JD1_PULSE_EN_SFT 10 +#define RT5682S_JD1_PULSE_DIS (0x0 << 10) +#define RT5682S_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682S_JD1_EN_MASK (0x1 << 15) +#define RT5682S_JD1_EN_SFT 15 +#define RT5682S_JD1_DIS (0x0 << 15) +#define RT5682S_JD1_EN (0x1 << 15) +#define RT5682S_JD1_POL_MASK (0x1 << 13) +#define RT5682S_JD1_POL_NOR (0x0 << 13) +#define RT5682S_JD1_POL_INV (0x1 << 13) +#define RT5682S_JD1_IRQ_MASK (0x1 << 10) +#define RT5682S_JD1_IRQ_LEV (0x0 << 10) +#define RT5682S_JD1_IRQ_PUL (0x1 << 10) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682S_IL_IRQ_MASK (0x1 << 7) +#define RT5682S_IL_IRQ_DIS (0x0 << 7) +#define RT5682S_IL_IRQ_EN (0x1 << 7) +#define RT5682S_IL_IRQ_TYPE_MASK (0x1 << 4) +#define RT5682S_IL_IRQ_LEV (0x0 << 4) +#define RT5682S_IL_IRQ_PUL (0x1 << 4) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682S_GP1_PIN_MASK (0x3 << 14) +#define RT5682S_GP1_PIN_SFT 14 +#define RT5682S_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682S_GP1_PIN_IRQ (0x1 << 14) +#define RT5682S_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682S_GP2_PIN_MASK (0x3 << 12) +#define RT5682S_GP2_PIN_SFT 12 +#define RT5682S_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682S_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682S_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682S_GP3_PIN_MASK (0x3 << 10) +#define RT5682S_GP3_PIN_SFT 10 +#define RT5682S_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682S_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682S_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682S_GP4_PIN_MASK (0x3 << 8) +#define RT5682S_GP4_PIN_SFT 8 +#define RT5682S_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682S_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682S_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682S_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682S_GP5_PIN_MASK (0x3 << 6) +#define RT5682S_GP5_PIN_SFT 6 +#define RT5682S_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682S_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682S_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682S_GP6_PIN_MASK (0x1 << 5) +#define RT5682S_GP6_PIN_SFT 5 +#define RT5682S_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682S_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682S_GP1_PF_MASK (0x1 << 15) +#define RT5682S_GP1_PF_IN (0x0 << 15) +#define RT5682S_GP1_PF_OUT (0x1 << 15) +#define RT5682S_GP1_OUT_MASK (0x1 << 14) +#define RT5682S_GP1_OUT_L (0x0 << 14) +#define RT5682S_GP1_OUT_H (0x1 << 14) +#define RT5682S_GP2_PF_MASK (0x1 << 13) +#define RT5682S_GP2_PF_IN (0x0 << 13) +#define RT5682S_GP2_PF_OUT (0x1 << 13) +#define RT5682S_GP2_OUT_MASK (0x1 << 12) +#define RT5682S_GP2_OUT_L (0x0 << 12) +#define RT5682S_GP2_OUT_H (0x1 << 12) +#define RT5682S_GP3_PF_MASK (0x1 << 11) +#define RT5682S_GP3_PF_IN (0x0 << 11) +#define RT5682S_GP3_PF_OUT (0x1 << 11) +#define RT5682S_GP3_OUT_MASK (0x1 << 10) +#define RT5682S_GP3_OUT_L (0x0 << 10) +#define RT5682S_GP3_OUT_H (0x1 << 10) +#define RT5682S_GP4_PF_MASK (0x1 << 9) +#define RT5682S_GP4_PF_IN (0x0 << 9) +#define RT5682S_GP4_PF_OUT (0x1 << 9) +#define RT5682S_GP4_OUT_MASK (0x1 << 8) +#define RT5682S_GP4_OUT_L (0x0 << 8) +#define RT5682S_GP4_OUT_H (0x1 << 8) +#define RT5682S_GP5_PF_MASK (0x1 << 7) +#define RT5682S_GP5_PF_IN (0x0 << 7) +#define RT5682S_GP5_PF_OUT (0x1 << 7) +#define RT5682S_GP5_OUT_MASK (0x1 << 6) +#define RT5682S_GP5_OUT_L (0x0 << 6) +#define RT5682S_GP5_OUT_H (0x1 << 6) +#define RT5682S_GP6_PF_MASK (0x1 << 5) +#define RT5682S_GP6_PF_IN (0x0 << 5) +#define RT5682S_GP6_PF_OUT (0x1 << 5) +#define RT5682S_GP6_OUT_MASK (0x1 << 4) +#define RT5682S_GP6_OUT_L (0x0 << 4) +#define RT5682S_GP6_OUT_H (0x1 << 4) + +/* GPIO Status (0x00c2) */ +#define RT5682S_GP6_ST (0x1 << 6) +#define RT5682S_GP5_ST (0x1 << 5) +#define RT5682S_GP4_ST (0x1 << 4) +#define RT5682S_GP3_ST (0x1 << 3) +#define RT5682S_GP2_ST (0x1 << 2) +#define RT5682S_GP1_ST (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682S_ZCD_MASK (0x1 << 10) +#define RT5682S_ZCD_SFT 10 +#define RT5682S_ZCD_PD (0x0 << 10) +#define RT5682S_ZCD_PU (0x1 << 10) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682S_4BTN_IL_MASK (0x1 << 15) +#define RT5682S_4BTN_IL_EN (0x1 << 15) +#define RT5682S_4BTN_IL_DIS (0x0 << 15) +#define RT5682S_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682S_4BTN_IL_NOR (0x1 << 14) +#define RT5682S_4BTN_IL_RST (0x0 << 14) + +/* 4 Button Inline Command Control 3~6 (0x00e5~0x00e8) */ +#define RT5682S_4BTN_IL_HOLD_WIN_MASK (0x7f << 8) +#define RT5682S_4BTN_IL_HOLD_WIN_SFT 8 +#define RT5682S_4BTN_IL_CLICK_WIN_MASK (0x7f) +#define RT5682S_4BTN_IL_CLICK_WIN_SFT 0 + +/* Analog JD Control (0x00f0) */ +#define RT5682S_JDH_RS_MASK (0x1 << 4) +#define RT5682S_JDH_NO_PLUG (0x1 << 4) +#define RT5682S_JDH_PLUG (0x0 << 4) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682S_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682S_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682S_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682S_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682S_CP_CLK_HP_600KHZ (0x3 << 4) + +/* Pad Driving Control (0x0136) */ +#define RT5682S_PAD_DRV_GP1_MASK (0x1 << 14) +#define RT5682S_PAD_DRV_GP1_HIGH (0x1 << 14) +#define RT5682S_PAD_DRV_GP1_LOW (0x0 << 14) +#define RT5682S_PAD_DRV_GP2_MASK (0x1 << 12) +#define RT5682S_PAD_DRV_GP2_HIGH (0x1 << 12) +#define RT5682S_PAD_DRV_GP2_LOW (0x0 << 12) +#define RT5682S_PAD_DRV_GP3_MASK (0x1 << 10) +#define RT5682S_PAD_DRV_GP3_HIGH (0x1 << 10) +#define RT5682S_PAD_DRV_GP3_LOW (0x0 << 10) +#define RT5682S_PAD_DRV_GP4_MASK (0x1 << 8) +#define RT5682S_PAD_DRV_GP4_HIGH (0x1 << 8) +#define RT5682S_PAD_DRV_GP4_LOW (0x0 << 8) +#define RT5682S_PAD_DRV_GP5_MASK (0x1 << 6) +#define RT5682S_PAD_DRV_GP5_HIGH (0x1 << 6) +#define RT5682S_PAD_DRV_GP5_LOW (0x0 << 6) +#define RT5682S_PAD_DRV_GP6_MASK (0x1 << 4) +#define RT5682S_PAD_DRV_GP6_HIGH (0x1 << 4) +#define RT5682S_PAD_DRV_GP6_LOW (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682S_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682S_CKXEN_DAC1_SFT 13 +#define RT5682S_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682S_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682S_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682S_CKXEN_ADC1_SFT 13 +#define RT5682S_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682S_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682S_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682S_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682S_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682S_AD2DA_LB_MASK (0x1 << 10) +#define RT5682S_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682S_NG2_EN_MASK (0x1 << 15) +#define RT5682S_NG2_EN (0x1 << 15) +#define RT5682S_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682S_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682S_DEB_80_MS (0x0 << 4) + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5682S_HP_SIG_SRC_MASK (0x3) +#define RT5682S_HP_SIG_SRC_1BIT_CTL (0x3) +#define RT5682S_HP_SIG_SRC_REG (0x2) +#define RT5682S_HP_SIG_SRC_IMPE_REG (0x1) +#define RT5682S_HP_SIG_SRC_DC_CALI (0x0) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682S_SAR_BUTDET_MASK (0x1 << 15) +#define RT5682S_SAR_BUTDET_EN (0x1 << 15) +#define RT5682S_SAR_BUTDET_DIS (0x0 << 15) +#define RT5682S_SAR_BUTDET_POW_MASK (0x1 << 14) +#define RT5682S_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682S_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682S_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682S_SAR_BUTDET_RST_NORM (0x1 << 13) +#define RT5682S_SAR_BUTDET_RST (0x0 << 13) +#define RT5682S_SAR_POW_MASK (0x1 << 12) +#define RT5682S_SAR_POW_EN (0x1 << 12) +#define RT5682S_SAR_POW_DIS (0x0 << 12) +#define RT5682S_SAR_RST_MASK (0x1 << 11) +#define RT5682S_SAR_RST_NORMAL (0x1 << 11) +#define RT5682S_SAR_RST (0x0 << 11) +#define RT5682S_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682S_SAR_BYPASS_EN (0x1 << 10) +#define RT5682S_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682S_SAR_SEL_MB1_2_MASK (0x3 << 8) +#define RT5682S_SAR_SEL_MB1_2_SFT 8 +#define RT5682S_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682S_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682S_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682S_SAR_SEL_MB1_2_CTL_MASK (0x1 << 5) +#define RT5682S_SAR_SEL_MB1_2_AUTO (0x1 << 5) +#define RT5682S_SAR_SEL_MB1_2_MANU (0x0 << 5) +#define RT5682S_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682S_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682S_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 2 (0x0211) */ +#define RT5682S_SAR_ADC_PSV_MASK (0x1 << 4) +#define RT5682S_SAR_ADC_PSV_ENTRY (0x1 << 4) + + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682S_SAR_SOUR_MASK (0x3f) +#define RT5682S_SAR_SOUR_BTN (0x3f) +#define RT5682S_SAR_SOUR_TYPE (0x0) + + +#define RT5682S_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +/* System Clock Source */ +enum { + RT5682S_SCLK_S_MCLK, + RT5682S_SCLK_S_PLL1, + RT5682S_SCLK_S_PLL2, + RT5682S_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682S_PLL_S_MCLK, + RT5682S_PLL_S_BCLK1, + RT5682S_PLL_S_BCLK2, + RT5682S_PLL_S_RCCLK, +}; + +enum { + RT5682S_PLL1, + RT5682S_PLL2, + RT5682S_PLLS, +}; + +enum { + RT5682S_AIF1, + RT5682S_AIF2, + RT5682S_AIFS +}; + +/* filter mask */ +enum { + RT5682S_DA_STEREO1_FILTER = 0x1, + RT5682S_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682S_CLK_SEL_SYS, + RT5682S_CLK_SEL_I2S1_ASRC, + RT5682S_CLK_SEL_I2S2_ASRC, +}; + +enum { + USE_PLLA, + USE_PLLB, + USE_PLLAB, +}; + +struct pll_calc_map { + unsigned int freq_in; + unsigned int freq_out; + int m; + int n; + int k; + bool m_bp; + bool k_bp; + bool byp_ps; + bool sel_ps; +}; + +#define RT5682S_NUM_SUPPLIES 2 + +struct rt5682s_priv { + struct snd_soc_component *component; + struct rt5682s_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682S_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + struct mutex sar_mutex; + +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682S_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + + int sysclk; + int sysclk_src; + int lrck[RT5682S_AIFS]; + int bclk[RT5682S_AIFS]; + int master[RT5682S_AIFS]; + + int pll_src[RT5682S_PLLS]; + int pll_in[RT5682S_PLLS]; + int pll_out[RT5682S_PLLS]; + int pll_comb; + + int jack_type; + int irq_work_delay_time; +}; + +int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682S_H__ */ -- 2.17.1

7 10

[PATCH v7 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 15 Sep '21

15 Sep '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v7 changes: Remove property 'classh-bst-max-limit' David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 151 ++ include/sound/cs35l41.h | 34 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 114 ++ sound/soc/codecs/cs35l41-spi.c | 143 ++ sound/soc/codecs/cs35l41-tables.c | 597 +++++++ sound/soc/codecs/cs35l41.c | 1545 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 775 +++++++++ 9 files changed, 3375 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

3 4

Location Of _snd_ctl_ops Definitions
by Ryan McClue 15 Sep '21

15 Sep '21

The struct _snd_ctl_ops contains a number of function pointers. Where are these functions defined in the alsa-lib source tree, e.g int (*subscribe_events)(snd_ctl_t *handle, int subscribe) -- Ryan McClue, Sydney

2 2

[PATCH 0/2] ASoC: cs42l42: Implement Manual Type detection as fallback
by Vitaly Rodionov 15 Sep '21

15 Sep '21

For some headsets CS42L42 autodetect mode is not working correctly. They will be detected as unknown types or as headphones. According to the CS42L42 datasheet, if the headset autodetect failed, then the driver should switch to manual mode and perform a manual steps sequence. These steps were missing in the current driver code. This patch will add manual mode fallback steps in case autodetect failed. The default behavior is not affected, manual mode runs only when autodetect failed. Tested for regression with autodetect with all known headsets - no regression. Tested with all headsets customers reported as false detected: Gumdrop DropTech B1 - detected as headset OK HUAWEI AM115 - detected as headset OK UGREEN EP103 - detected as headset OK HONOR AM116 - detected as headset OK Also, fixes some errors reported by checkpatch.pl script. Stefan Binding (1): ASoC: cs42l42: Implement Manual Type detection as fallback Vitaly Rodionov (1): ASoC: cs42l42: Minor fix all errors reported by checkpatch.pl script sound/soc/codecs/cs42l42.c | 114 +++++++++++++++++++++++++++++++------ sound/soc/codecs/cs42l42.h | 54 ++++++++++++++++++ 2 files changed, 151 insertions(+), 17 deletions(-) -- 2.25.1

2 4

[PATCH 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 15 Sep '21

15 Sep '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. These patches depends on the dt-bindings header patch -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver .../bindings/sound/google,sc7280-herobrine.yaml | 169 ++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 347 +++++++++++++++++++++ 4 files changed, 530 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 8

[PATCH v4] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 15 Sep '21

15 Sep '21

The document was merged as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [1] revealed that the commit 1afc60e00de3 breaks dt_binding_check due to dt-bindings/clock/mt8192-clk.h doesn't exist. As a temporary fix, commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") reverted commit 1afc60e00de3. dt-bindings/clock/mt8192-clk.h is in mainline per commit f35f1a23e0e1 ("clk: mediatek: Add dt-bindings of MT8192 clocks"). Re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v3 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/189705…): - Fix the indent errors. Changes from v2 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189058.ht…): - Simplify the commit message. Changes from v1 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189048.ht…): - Add more context to the commit message. .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..5e9fe067f440 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.309.g3052b89438-goog

1 0

[PATCH v2] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 14 Sep '21

14 Sep '21

These support 3 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2 or 4 speakers. Left/Right speakers and 4 speakers tdm playback add echo reference capture add BT_OFFLOAD support add DMI_OEM_STRING for board varients Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> --- Changes logs: -- add clearer hw boards info -- move Kconfig MAX98390 up, correct SSP2 wrong quirk -- modify snd_soc_dai_set_tdm_slot() supporting 2/4 dai codecs -- aligning topology name: sof-adl-rt5682-ssp0-max98390-ssp2.tplg sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 170 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 69 +++++++ .../intel/common/soc-acpi-intel-adl-match.c | 13 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 278 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 61b71d6c44cf..9723d04cbe74 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -466,6 +466,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..eeaa96648c5a 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,176 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; +EXPORT_SYMBOL_NS(max_98390_tt_dapm_routes, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, + ARRAY_SIZE(max_98373_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "Speaker Left, Right map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Tweeter Speaker Left, Right map addition failed: %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..412b9dd06bad 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern struct snd_soc_ops max_98390_ops; +extern const struct snd_soc_dapm_route max_98390_dapm_routes[]; +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f096bd6d69be..9acadfe03ef3 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,8 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(24) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -176,6 +178,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -459,6 +491,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -745,6 +778,21 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + links[id].init = max_98390_spk_codec_init; + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + } + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -881,6 +929,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1007,6 +1063,18 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { } }; MODULE_DEVICE_TABLE(platform, board_ids); @@ -1026,6 +1094,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

3 2

git clone failed
by GitHub issues - edited 14 Sep '21

14 Sep '21

alsa-project/tinycompress issue #12 was edited from BalajiMalar: i was building yocto image which uses tinycomrpess as part of NXP's recipe-multimedia. the yocto build failed since it wasnt able to clone the package. i tried to clone from CLI using the command "**git clone git://git.alsa-project.org/tinycompress.git**" and still it failed Issue URL : https://github.com/alsa-project/tinycompress/issues/12 Repository URL: https://github.com/alsa-project/tinycompress

1 0

git clone failed
by GitHub issues - opened 14 Sep '21

14 Sep '21

alsa-project/tinycompress issue #12 was opened from BalajiMalar: i was building yocto image which uses tinycomrpess as part of NXP's recipe-multimedia. the yocto build failed since it wasnt able to clone the package. i tried to clone from CLI using the command "**git clone git clone git://git.alsa-project.org/tinycompress.git**" and still it failed Issue URL : https://github.com/alsa-project/tinycompress/issues/12 Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH v2] ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
by Cameron Berkenpas 14 Sep '21

14 Sep '21

This patch initializes and enables speaker output on the Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 series of laptops using the HDA verb sequence specific to each model. Speaker automute is suppressed for the Lenovo Legion 7i 15IMHG05 to avoid breaking speaker output on resume and when devices are unplugged from its headphone jack. Thanks to: Andreas Holzer, Vincent Morel, sycxyc, Max Christian Pohle and all others that helped. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aed..89a6ed26b7b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.suppress_auto_mute = 1; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6646,6 +6660,10 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS }; static const struct hda_fixup alc269_fixups[] = { @@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.30.2

2 1

[PATCH] soundwire: export sdw_nwrite_no_pm and sdw_nread_no_pm functions
by Simon Trimmer 14 Sep '21

14 Sep '21

The patch "soundwire: export sdw_write/read_no_pm functions" exposed the single byte no_pm versions of the IO functions that can be used without touching PM, export the multi byte no_pm versions for the same reason. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 8 ++++---- include/linux/soundwire/sdw.h | 2 ++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 1b115734a8f6..859de302dee2 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -379,8 +379,7 @@ int sdw_fill_msg(struct sdw_msg *msg, struct sdw_slave *slave, * all clients need to use the pm versions */ -static int -sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { struct sdw_msg msg; int ret; @@ -395,9 +394,9 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) ret = 0; return ret; } +EXPORT_SYMBOL(sdw_nread_no_pm); -static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) +int sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; @@ -412,6 +411,7 @@ sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) ret = 0; return ret; } +EXPORT_SYMBOL(sdw_nwrite_no_pm); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value) { diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 76ce3f3ac0f2..2a5395f0dcf1 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1042,7 +1042,9 @@ int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value); int sdw_read_no_pm(struct sdw_slave *slave, u32 addr); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); +int sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); int sdw_update(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); int sdw_update_no_pm(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); -- 2.33.0

4 3

[PATCH AUTOSEL 5.13 07/19] ASoC: audio-graph: respawn Platform Support
by Sasha Levin 14 Sep '21

14 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 5f939f49771002f347039edf984aca42f30fc31a ] commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Link: https://lore.kernel.org/r/878s0jzrpf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 09/25] ASoC: audio-graph: respawn Platform Support
by Sasha Levin 14 Sep '21

14 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 5f939f49771002f347039edf984aca42f30fc31a ] commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Link: https://lore.kernel.org/r/878s0jzrpf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.30.2

1 0

[PATCH] ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
by Cameron Berkenpas 14 Sep '21

14 Sep '21

This patch initializes and enables speaker output on the Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 series of laptops using the HDA verb sequence specific to each model. Speaker automute is disabled for the Lenovo Legion 7i 15IMHG05 to avoid breaking speaker output on resume and when devices are unplugged from its headphone jack. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aed..7e94107e8e59 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_INIT: + spec->gen.automute_speaker = 0; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6646,6 +6660,10 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS }; static const struct hda_fixup alc269_fixups[] = { @@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.30.2

2 5

kernel threads preventing wakeup of thread using snd_pcm_readi?
by Bert Schiettecatte 13 Sep '21

13 Sep '21

Hi all I am working on a realtime audio application on the rk3288 platform. I am using stable kernel 5.13.13. I'm using the open source tracy profiler to profile my application. My development PC receives trace data via ethernet from my platform. I have custom alsa drivers on my platform for my DACs and ADCs. My application has a thread in which I use snd_pcm_readi which blocks until it can read enough samples. I have another thread in which I write the samples to disk (sandisk extreme SD card). I use a ring-buffer so the code is lock-free. I've noticed that over a timespan of a few minutes I get a few clicks, which seem to come from the thread blocking on snd_pcm_readi not waking up in time. I am observing this in the traces I collect via tracy. Right before this happens, I can see kworker/kblockd being scheduled on one of the cores many times and rapidly, or ocasionally ksoftirqd, mmc_complete, or events_highpri. From the trace I collect, i have the impression nothing is being scheduled on the other cores, so I do not see a reason why my thread should not be woken up and scheduled. Why is the thread blocking on snd_pcm_readi not waking up in time, and is it possible that the kworker threads are preventing this, and why? Any suggestions on how to debug this further are greatly appreciated. Thanks Bert

1 0

[PATCH 0/2] ARM: dts: sun8i: r40: Add I2S nodes
by Jernej Skrabec 13 Sep '21

13 Sep '21

Allwinner R40 has 3 I2S controllers, compatible to those in H3. Patch 1 adds R40/H3 compatible pair to DT bindings. Patch 2 adds I2S nodes to R40 DT. Please take a look. Best regards, Jernej Jernej Skrabec (2): dt-bindings: sound: sun4i-i2s: add Allwinner R40 I2S compatible ARM: dts: sun8i: r40: Add I2S nodes .../sound/allwinner,sun4i-a10-i2s.yaml | 3 ++ arch/arm/boot/dts/sun8i-r40.dtsi | 39 +++++++++++++++++++ 2 files changed, 42 insertions(+) -- 2.33.0

3 6

[PATCH v2] ASoC: mediatek: common: handle NULL case in suspend/resume function
by Trevor Wu 13 Sep '21

13 Sep '21

When memory allocation for afe->reg_back_up fails, reg_back_up can't be used. Keep the suspend/resume flow but skip register backup when afe->reg_back_up is NULL, in case illegal memory access happens. Fixes: 283b612429a2 ("ASoC: mediatek: implement mediatek common structure") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Change since v1: add Reported-by tag --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d4..e95c7c018e7d 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component) devm_kcalloc(dev, afe->reg_back_up_list_num, sizeof(unsigned int), GFP_KERNEL); - for (i = 0; i < afe->reg_back_up_list_num; i++) - regmap_read(regmap, afe->reg_back_up_list[i], - &afe->reg_back_up[i]); + if (afe->reg_back_up) { + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + } afe->suspended = true; afe->runtime_suspend(dev); @@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component) afe->runtime_resume(dev); - if (!afe->reg_back_up) + if (!afe->reg_back_up) { dev_dbg(dev, "%s no reg_backup\n", __func__); - - for (i = 0; i < afe->reg_back_up_list_num; i++) - mtk_regmap_write(regmap, afe->reg_back_up_list[i], - afe->reg_back_up[i]); + } else { + for (i = 0; i < afe->reg_back_up_list_num; i++) + mtk_regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + } afe->suspended = false; return 0; -- 2.18.0

2 4

[PATCH] Revert "ALSA: hda: Drop workaround for a hang at shutdown again"
by Takashi Iwai 13 Sep '21

13 Sep '21

This reverts commit 8fc8e903156f42c66245838441d03607e9067381. It was expected that the fixes in HD-audio codec side would make the workaround redundant, but unfortunately it doesn't seem sufficing. Resurrect the workaround for now. Fixes: 8fc8e903156f ("ALSA: hda: Drop workaround for a hang at shutdown again") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214045 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3aa432d814a2..47777439961c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void azx_shutdown_chip(struct azx *chip) +static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) { azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!skip_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); +static void azx_shutdown_chip(struct azx *chip) +{ + __azx_shutdown_chip(chip, false); +} + static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2357,7 +2363,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_shutdown_chip(chip); + __azx_shutdown_chip(chip, true); } /* PCI IDs */ -- 2.26.2

1 0

[PATCH] ASoC: pcm5102a: increase rate from 192k to 384k
by gearhead 13 Sep '21

13 Sep '21

the pcm5102a is capable of 384k, but the current code limits it to 192k. This commit extends to 384k Signed-off-by: gearhead <ys3al35l(a)gmail.com> --- sound/soc/codecs/pcm5102a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index b8cfc250612c..f39f98bbc97f 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -17,7 +17,7 @@ static struct snd_soc_dai_driver pcm5102a_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_8000_384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.33.0

2 1

[PATCH] ASoC: mediatek: mt8195: Make use of the helper function devm_platform_ioremap_resource()
by Cai Huoqing 13 Sep '21

13 Sep '21

Use the devm_platform_ioremap_resource() helper instead of calling platform_get_resource() and devm_ioremap_resource() separately Signed-off-by: Cai Huoqing <caihuoqing(a)baidu.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 6635c3f72ecc..8c697f41d51d 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3057,7 +3057,6 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; struct mt8195_afe_private *afe_priv; - struct resource *res; struct device *dev = &pdev->dev; int i, irq_id, ret; struct snd_soc_component *component; @@ -3078,8 +3077,7 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) afe_priv = afe->platform_priv; afe->dev = &pdev->dev; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - afe->base_addr = devm_ioremap_resource(&pdev->dev, res); + afe->base_addr = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(afe->base_addr)) return PTR_ERR(afe->base_addr); -- 2.25.1

2 1

[PATCH] ASoC: SOF: core: allow module parameter to override dma trace Kconfig
by Pierre-Louis Bossart 13 Sep '21

13 Sep '21

Most distributions do not enable the SOF developer options and specifically the DMA trace. This is problematic for end-user/community support since the sof-logger tool cannot extract valuable information. Conversely in rare cases the DMA trace can lead to Heisenbugs by creating more traffic to system memory and more interrupts. This patch changes the logic so that the Kconfig value is used as a default value for a module parameter, but this value can be changed as needed. Users can override the distro DMA trace selection. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e4dd4a86363..6be4f159ee35 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -19,7 +19,7 @@ #endif /* see SOF_DBG_ flags */ -int sof_core_debug; +int sof_core_debug = IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_FIRMWARE_TRACE); module_param_named(sof_debug, sof_core_debug, int, 0444); MODULE_PARM_DESC(sof_debug, "SOF core debug options (0x0 all off)"); @@ -202,8 +202,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) goto fw_run_err; } - if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_FIRMWARE_TRACE) || - (sof_core_debug & SOF_DBG_ENABLE_TRACE)) { + if (sof_core_debug & SOF_DBG_ENABLE_TRACE) { sdev->dtrace_is_supported = true; /* init DMA trace */ -- 2.25.1

2 1

[PATCH v7 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 13 Sep '21

13 Sep '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v7 changes: Remove property 'classh-bst-max-limit' David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 151 ++ include/sound/cs35l41.h | 34 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 114 ++ sound/soc/codecs/cs35l41-spi.c | 143 ++ sound/soc/codecs/cs35l41-tables.c | 597 +++++++ sound/soc/codecs/cs35l41.c | 1545 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 775 +++++++++ 9 files changed, 3375 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

4 6

[PATCH] ASoC: mt8195: remove unnecessary CONFIG_PM
by Trevor Wu 13 Sep '21

13 Sep '21

The unnecessary conditional inclusion caused the following warning. >> sound/soc/mediatek/mt8195/mt8195-afe-pcm.c:3260:32: warning: unused >> variable 'mt8195_afe_pm_ops' [-Wunused-const-variable] static const struct dev_pm_ops mt8195_afe_pm_ops = { ^ 1 warning generated. Because runtime_pm already handles the case without CONFIG_PM, we can remove CONFIG_PM condition. Fixes: 6746cc858259 ("ASoC: mediatek: mt8195: add platform driver") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 6635c3f72ecc..f8dd50c8a7da 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3266,9 +3266,7 @@ static struct platform_driver mt8195_afe_pcm_driver = { .driver = { .name = "mt8195-audio", .of_match_table = mt8195_afe_pcm_dt_match, -#ifdef CONFIG_PM .pm = &mt8195_afe_pm_ops, -#endif }, .probe = mt8195_afe_pcm_dev_probe, .remove = mt8195_afe_pcm_dev_remove, -- 2.18.0

2 1

[PATCH] ASoC: SOF: Handle control change notification from firmware
by Peter Ujfalusi 13 Sep '21

13 Sep '21

If the value/data associated with a control changes in SOF it will send a notification (SOF_IPC_GLB_COMP_MSG with SOF_IPC_COMP_GET_VALUE/DATA). We have support for binary volatile control type, but we might have features where enum/switch/volume changes. Re-implementing everything as volatile as well would be not much of a gain for several reasons: - volatile controls would do an IPC all the time, regardless if there is a need or not. - We still don't have notification which forces userspace to continuously poll. When such notification arrives we use snd_ctl_notify_one() to signal userspace about the change. The kernel is prepared for two types of notification: - the notification carries the new data for the control (num_elems != 0) The new value/data is copied to the control's local data - blank message about a change The new flag for the scontrol (comp_data_dirty) is set and when next time user space reads the value via the kcontrol's get callback we will refresh the control's local data from the firmware. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Tested-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> --- sound/soc/sof/control.c | 186 ++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc.c | 35 +++++++ sound/soc/sof/sof-audio.h | 5 + 3 files changed, 226 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 504500dd4d43..58bb89af4de1 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -65,6 +65,40 @@ static inline u32 ipc_to_mixer(u32 value, u32 *volume_map, int size) return i - 1; } +static void snd_sof_refresh_control(struct snd_sof_control *scontrol) +{ + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct snd_soc_component *scomp = scontrol->scomp; + enum sof_ipc_ctrl_type ctrl_type; + int ret; + + if (!scontrol->comp_data_dirty) + return; + + if (!pm_runtime_active(scomp->dev)) + return; + + if (scontrol->cmd == SOF_CTRL_CMD_BINARY) + ctrl_type = SOF_IPC_COMP_GET_DATA; + else + ctrl_type = SOF_IPC_COMP_GET_VALUE; + + /* set the ABI header values */ + cdata->data->magic = SOF_ABI_MAGIC; + cdata->data->abi = SOF_ABI_VERSION; + + /* refresh the component data from DSP */ + scontrol->comp_data_dirty = false; + ret = snd_sof_ipc_set_get_comp_data(scontrol, ctrl_type, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + scontrol->cmd, false); + if (ret < 0) { + dev_err(scomp->dev, "error: failed to get control data: %d\n", ret); + /* Set the flag to re-try next time to get the data */ + scontrol->comp_data_dirty = true; + } +} + int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -74,6 +108,8 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + snd_sof_refresh_control(scontrol); + /* read back each channel */ for (i = 0; i < channels; i++) ucontrol->value.integer.value[i] = @@ -145,6 +181,8 @@ int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + snd_sof_refresh_control(scontrol); + /* read back each channel */ for (i = 0; i < channels; i++) ucontrol->value.integer.value[i] = cdata->chanv[i].value; @@ -195,6 +233,8 @@ int snd_sof_enum_get(struct snd_kcontrol *kcontrol, struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + snd_sof_refresh_control(scontrol); + /* read back each channel */ for (i = 0; i < channels; i++) ucontrol->value.enumerated.item[i] = cdata->chanv[i].value; @@ -244,6 +284,8 @@ int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, struct sof_abi_hdr *data = cdata->data; size_t size; + snd_sof_refresh_control(scontrol); + if (be->max > sizeof(ucontrol->value.bytes.data)) { dev_err_ratelimited(scomp->dev, "error: data max %d exceeds ucontrol data array size\n", @@ -475,6 +517,8 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, (struct snd_ctl_tlv __user *)binary_data; size_t data_size; + snd_sof_refresh_control(scontrol); + /* * Decrement the limit by ext bytes header size to * ensure the user space buffer is not exceeded. @@ -511,3 +555,145 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, return 0; } + +static void snd_sof_update_control(struct snd_sof_control *scontrol, + struct sof_ipc_ctrl_data *cdata) +{ + struct snd_soc_component *scomp = scontrol->scomp; + struct sof_ipc_ctrl_data *local_cdata; + int i; + + local_cdata = scontrol->control_data; + + if (cdata->cmd == SOF_CTRL_CMD_BINARY) { + if (cdata->num_elems != local_cdata->data->size) { + dev_err(scomp->dev, + "error: cdata binary size mismatch %u - %u\n", + cdata->num_elems, local_cdata->data->size); + return; + } + + /* copy the new binary data */ + memcpy(local_cdata->data, cdata->data, cdata->num_elems); + } else if (cdata->num_elems != scontrol->num_channels) { + dev_err(scomp->dev, + "error: cdata channel count mismatch %u - %d\n", + cdata->num_elems, scontrol->num_channels); + } else { + /* copy the new values */ + for (i = 0; i < cdata->num_elems; i++) + local_cdata->chanv[i].value = cdata->chanv[i].value; + } +} + +void snd_sof_control_notify(struct snd_sof_dev *sdev, + struct sof_ipc_ctrl_data *cdata) +{ + struct snd_soc_dapm_widget *widget; + struct snd_sof_control *scontrol; + struct snd_sof_widget *swidget; + struct snd_kcontrol *kc = NULL; + struct soc_mixer_control *sm; + struct soc_bytes_ext *be; + size_t expected_size; + struct soc_enum *se; + bool found = false; + int i, type; + + /* Find the swidget first */ + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (swidget->comp_id == cdata->comp_id) { + found = true; + break; + } + } + + if (!found) + return; + + /* Translate SOF cmd to TPLG type */ + switch (cdata->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_SWITCH: + type = SND_SOC_TPLG_TYPE_MIXER; + break; + case SOF_CTRL_CMD_BINARY: + type = SND_SOC_TPLG_TYPE_BYTES; + break; + case SOF_CTRL_CMD_ENUM: + type = SND_SOC_TPLG_TYPE_ENUM; + break; + default: + dev_err(sdev->dev, "error: unknown cmd %u\n", cdata->cmd); + return; + } + + widget = swidget->widget; + for (i = 0; i < widget->num_kcontrols; i++) { + /* skip non matching types or non matching indexes within type */ + if (widget->dobj.widget.kcontrol_type[i] == type && + widget->kcontrol_news[i].index == cdata->index) { + kc = widget->kcontrols[i]; + break; + } + } + + if (!kc) + return; + + switch (cdata->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_SWITCH: + sm = (struct soc_mixer_control *)kc->private_value; + scontrol = sm->dobj.private; + break; + case SOF_CTRL_CMD_BINARY: + be = (struct soc_bytes_ext *)kc->private_value; + scontrol = be->dobj.private; + break; + case SOF_CTRL_CMD_ENUM: + se = (struct soc_enum *)kc->private_value; + scontrol = se->dobj.private; + break; + default: + return; + } + + expected_size = sizeof(struct sof_ipc_ctrl_data); + switch (cdata->type) { + case SOF_CTRL_TYPE_VALUE_CHAN_GET: + case SOF_CTRL_TYPE_VALUE_CHAN_SET: + expected_size += cdata->num_elems * + sizeof(struct sof_ipc_ctrl_value_chan); + break; + case SOF_CTRL_TYPE_VALUE_COMP_GET: + case SOF_CTRL_TYPE_VALUE_COMP_SET: + expected_size += cdata->num_elems * + sizeof(struct sof_ipc_ctrl_value_comp); + break; + case SOF_CTRL_TYPE_DATA_GET: + case SOF_CTRL_TYPE_DATA_SET: + expected_size += cdata->num_elems + sizeof(struct sof_abi_hdr); + break; + default: + return; + } + + if (cdata->rhdr.hdr.size != expected_size) { + dev_err(sdev->dev, "error: component notification size mismatch\n"); + return; + } + + if (cdata->num_elems) + /* + * The message includes the updated value/data, update the + * control's local cache using the received notification + */ + snd_sof_update_control(scontrol, cdata); + else + /* Mark the scontrol that the value/data is changed in SOF */ + scontrol->comp_data_dirty = true; + + snd_ctl_notify_one(swidget->scomp->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, kc, 0); +} diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index c2d07b783f60..6260f4ba5618 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -369,6 +369,39 @@ void snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) } EXPORT_SYMBOL(snd_sof_ipc_reply); +static void ipc_comp_notification(struct snd_sof_dev *sdev, + struct sof_ipc_cmd_hdr *hdr) +{ + u32 msg_type = hdr->cmd & SOF_CMD_TYPE_MASK; + struct sof_ipc_ctrl_data *cdata; + int ret; + + switch (msg_type) { + case SOF_IPC_COMP_GET_VALUE: + case SOF_IPC_COMP_GET_DATA: + cdata = kmalloc(hdr->size, GFP_KERNEL); + if (!cdata) + return; + + /* read back full message */ + ret = snd_sof_ipc_msg_data(sdev, NULL, cdata, hdr->size); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to read component event: %d\n", ret); + goto err; + } + break; + default: + dev_err(sdev->dev, "error: unhandled component message %#x\n", msg_type); + return; + } + + snd_sof_control_notify(sdev, cdata); + +err: + kfree(cdata); +} + /* DSP firmware has sent host a message */ void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev) { @@ -404,7 +437,9 @@ void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev) case SOF_IPC_GLB_COMPOUND: case SOF_IPC_GLB_TPLG_MSG: case SOF_IPC_GLB_PM_MSG: + break; case SOF_IPC_GLB_COMP_MSG: + ipc_comp_notification(sdev, &hdr); break; case SOF_IPC_GLB_STREAM_MSG: /* need to pass msg id into the function */ diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index dc274e63ed9a..9a8d005e75a0 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -75,6 +75,9 @@ struct snd_sof_control { struct list_head list; /* list in sdev control list */ struct snd_sof_led_control led_ctl; + + /* if true, the control's data needs to be updated from Firmware */ + bool comp_data_dirty; }; /* ASoC SOF DAPM widget */ @@ -148,6 +151,8 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, unsigned int size); int snd_sof_bytes_ext_volatile_get(struct snd_kcontrol *kcontrol, unsigned int __user *binary_data, unsigned int size); +void snd_sof_control_notify(struct snd_sof_dev *sdev, + struct sof_ipc_ctrl_data *cdata); /* * Topology. -- 2.33.0

2 1

[PATCH] ASoC: ti: rename CONFIG_SND_SOC_DM365_VOICE_CODEC_MODULE
by Masahiro Yamada 13 Sep '21

13 Sep '21

Kconfig generates include/generated/autoconf.h to make CONFIG options available to the pre-processor. Symbols with the value 'm' are suffixed with '_MODULE' Here is a conflict; CONFIG_FOO=m results in '#define CONFIG_FOO_MODULE 1', but CONFIG_FOO_MODULE=y also results in the same define. Also, CONFIG options that end with '_MODULE' confuse the Kconfig/fixdep interaction; fixdep always assumes CONFIG_FOO_MODULE comes from CONFIG_FOO=m, so the dependency is not properly tracked for symbols that end with '_MODULE'. For these reasons, CONFIG options that end with '_MODULE' should be avoided in general. (I am planning on adding a check in Kconfig.) This is the only case in the current kernel. The new option name was suggested by Péter Ujfalusi. [1] [1] https://lore.kernel.org/all/d9e777dc-d274-92ee-4d77-711bfd553611@gmail.com/ Fixes: 147162f57515 ("ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies") Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- sound/soc/ti/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 698d7bc84dcf..3260a30d3d54 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -211,7 +211,7 @@ config SND_SOC_DM365_VOICE_CODEC Say Y if you want to add support for SoC On-chip voice codec endchoice -config SND_SOC_DM365_VOICE_CODEC_MODULE +config SND_SOC_DM365_SELECT_VOICE_CODECS def_tristate y depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC select MFD_DAVINCI_VOICECODEC -- 2.30.2

3 2

[PATCH] ASoC: soc-topology: Move template info print soc_tplg_dapm_widget_create()
by Peter Ujfalusi 13 Sep '21

13 Sep '21

A DAPM widget now can have different types of controls, it is no longer correct to print the type as it is just the type of the first control. Move it after the loop where we create the controls and print the number of the control types. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-topology.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f6e5ac3e0314..73e1b7b48ce9 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1473,10 +1473,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, goto widget; } - control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; - dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", - w->name, w->num_kcontrols, control_hdr->type); - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); if (!kc) @@ -1536,6 +1532,8 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } template.kcontrol_news = kc; + dev_dbg(tplg->dev, "ASoC: template %s with %d/%d/%d (mixer/enum/bytes) control\n", + w->name, mixer_count, enum_count, bytes_count); widget: ret = soc_tplg_widget_load(tplg, &template, w); -- 2.33.0

2 1

[PATCH v2] ASoC: SOF: Handle control change notification from firmware
by Peter Ujfalusi 13 Sep '21

13 Sep '21

If the value/data associated with a control changes in SOF it will send a notification (SOF_IPC_GLB_COMP_MSG with SOF_IPC_COMP_GET_VALUE/DATA). We have support for binary volatile control type, but we might have features where enum/switch/volume changes. Re-implementing everything as volatile as well would be not much of a gain for several reasons: - volatile controls would do an IPC all the time, regardless if there is a need or not. - We still don't have notification which forces userspace to continuously poll. When such notification arrives we use snd_ctl_notify_one() to signal userspace about the change. The kernel is prepared for two types of notification: - the notification carries the new data for the control (num_elems != 0) The new value/data is copied to the control's local data - blank message about a change The new flag for the scontrol (comp_data_dirty) is set and when next time user space reads the value via the kcontrol's get callback we will refresh the control's local data from the firmware. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Tested-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> --- Hi, Changes since v1: - Fix error reported by kernel test robot <lkp(a)intel.com>: snd_sof_ipc_msg_data() returns void in upstream (downstream it is int) I don't know how I missed with my sparse build, sorry about that. Regards, Peter sound/soc/sof/control.c | 186 ++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc.c | 28 ++++++ sound/soc/sof/sof-audio.h | 5 + 3 files changed, 219 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 504500dd4d43..58bb89af4de1 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -65,6 +65,40 @@ static inline u32 ipc_to_mixer(u32 value, u32 *volume_map, int size) return i - 1; } +static void snd_sof_refresh_control(struct snd_sof_control *scontrol) +{ + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct snd_soc_component *scomp = scontrol->scomp; + enum sof_ipc_ctrl_type ctrl_type; + int ret; + + if (!scontrol->comp_data_dirty) + return; + + if (!pm_runtime_active(scomp->dev)) + return; + + if (scontrol->cmd == SOF_CTRL_CMD_BINARY) + ctrl_type = SOF_IPC_COMP_GET_DATA; + else + ctrl_type = SOF_IPC_COMP_GET_VALUE; + + /* set the ABI header values */ + cdata->data->magic = SOF_ABI_MAGIC; + cdata->data->abi = SOF_ABI_VERSION; + + /* refresh the component data from DSP */ + scontrol->comp_data_dirty = false; + ret = snd_sof_ipc_set_get_comp_data(scontrol, ctrl_type, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + scontrol->cmd, false); + if (ret < 0) { + dev_err(scomp->dev, "error: failed to get control data: %d\n", ret); + /* Set the flag to re-try next time to get the data */ + scontrol->comp_data_dirty = true; + } +} + int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -74,6 +108,8 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + snd_sof_refresh_control(scontrol); + /* read back each channel */ for (i = 0; i < channels; i++) ucontrol->value.integer.value[i] = @@ -145,6 +181,8 @@ int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + snd_sof_refresh_control(scontrol); + /* read back each channel */ for (i = 0; i < channels; i++) ucontrol->value.integer.value[i] = cdata->chanv[i].value; @@ -195,6 +233,8 @@ int snd_sof_enum_get(struct snd_kcontrol *kcontrol, struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + snd_sof_refresh_control(scontrol); + /* read back each channel */ for (i = 0; i < channels; i++) ucontrol->value.enumerated.item[i] = cdata->chanv[i].value; @@ -244,6 +284,8 @@ int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, struct sof_abi_hdr *data = cdata->data; size_t size; + snd_sof_refresh_control(scontrol); + if (be->max > sizeof(ucontrol->value.bytes.data)) { dev_err_ratelimited(scomp->dev, "error: data max %d exceeds ucontrol data array size\n", @@ -475,6 +517,8 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, (struct snd_ctl_tlv __user *)binary_data; size_t data_size; + snd_sof_refresh_control(scontrol); + /* * Decrement the limit by ext bytes header size to * ensure the user space buffer is not exceeded. @@ -511,3 +555,145 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, return 0; } + +static void snd_sof_update_control(struct snd_sof_control *scontrol, + struct sof_ipc_ctrl_data *cdata) +{ + struct snd_soc_component *scomp = scontrol->scomp; + struct sof_ipc_ctrl_data *local_cdata; + int i; + + local_cdata = scontrol->control_data; + + if (cdata->cmd == SOF_CTRL_CMD_BINARY) { + if (cdata->num_elems != local_cdata->data->size) { + dev_err(scomp->dev, + "error: cdata binary size mismatch %u - %u\n", + cdata->num_elems, local_cdata->data->size); + return; + } + + /* copy the new binary data */ + memcpy(local_cdata->data, cdata->data, cdata->num_elems); + } else if (cdata->num_elems != scontrol->num_channels) { + dev_err(scomp->dev, + "error: cdata channel count mismatch %u - %d\n", + cdata->num_elems, scontrol->num_channels); + } else { + /* copy the new values */ + for (i = 0; i < cdata->num_elems; i++) + local_cdata->chanv[i].value = cdata->chanv[i].value; + } +} + +void snd_sof_control_notify(struct snd_sof_dev *sdev, + struct sof_ipc_ctrl_data *cdata) +{ + struct snd_soc_dapm_widget *widget; + struct snd_sof_control *scontrol; + struct snd_sof_widget *swidget; + struct snd_kcontrol *kc = NULL; + struct soc_mixer_control *sm; + struct soc_bytes_ext *be; + size_t expected_size; + struct soc_enum *se; + bool found = false; + int i, type; + + /* Find the swidget first */ + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (swidget->comp_id == cdata->comp_id) { + found = true; + break; + } + } + + if (!found) + return; + + /* Translate SOF cmd to TPLG type */ + switch (cdata->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_SWITCH: + type = SND_SOC_TPLG_TYPE_MIXER; + break; + case SOF_CTRL_CMD_BINARY: + type = SND_SOC_TPLG_TYPE_BYTES; + break; + case SOF_CTRL_CMD_ENUM: + type = SND_SOC_TPLG_TYPE_ENUM; + break; + default: + dev_err(sdev->dev, "error: unknown cmd %u\n", cdata->cmd); + return; + } + + widget = swidget->widget; + for (i = 0; i < widget->num_kcontrols; i++) { + /* skip non matching types or non matching indexes within type */ + if (widget->dobj.widget.kcontrol_type[i] == type && + widget->kcontrol_news[i].index == cdata->index) { + kc = widget->kcontrols[i]; + break; + } + } + + if (!kc) + return; + + switch (cdata->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_SWITCH: + sm = (struct soc_mixer_control *)kc->private_value; + scontrol = sm->dobj.private; + break; + case SOF_CTRL_CMD_BINARY: + be = (struct soc_bytes_ext *)kc->private_value; + scontrol = be->dobj.private; + break; + case SOF_CTRL_CMD_ENUM: + se = (struct soc_enum *)kc->private_value; + scontrol = se->dobj.private; + break; + default: + return; + } + + expected_size = sizeof(struct sof_ipc_ctrl_data); + switch (cdata->type) { + case SOF_CTRL_TYPE_VALUE_CHAN_GET: + case SOF_CTRL_TYPE_VALUE_CHAN_SET: + expected_size += cdata->num_elems * + sizeof(struct sof_ipc_ctrl_value_chan); + break; + case SOF_CTRL_TYPE_VALUE_COMP_GET: + case SOF_CTRL_TYPE_VALUE_COMP_SET: + expected_size += cdata->num_elems * + sizeof(struct sof_ipc_ctrl_value_comp); + break; + case SOF_CTRL_TYPE_DATA_GET: + case SOF_CTRL_TYPE_DATA_SET: + expected_size += cdata->num_elems + sizeof(struct sof_abi_hdr); + break; + default: + return; + } + + if (cdata->rhdr.hdr.size != expected_size) { + dev_err(sdev->dev, "error: component notification size mismatch\n"); + return; + } + + if (cdata->num_elems) + /* + * The message includes the updated value/data, update the + * control's local cache using the received notification + */ + snd_sof_update_control(scontrol, cdata); + else + /* Mark the scontrol that the value/data is changed in SOF */ + scontrol->comp_data_dirty = true; + + snd_ctl_notify_one(swidget->scomp->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, kc, 0); +} diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index c2d07b783f60..a4fe007a0e4d 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -369,6 +369,32 @@ void snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) } EXPORT_SYMBOL(snd_sof_ipc_reply); +static void ipc_comp_notification(struct snd_sof_dev *sdev, + struct sof_ipc_cmd_hdr *hdr) +{ + u32 msg_type = hdr->cmd & SOF_CMD_TYPE_MASK; + struct sof_ipc_ctrl_data *cdata; + + switch (msg_type) { + case SOF_IPC_COMP_GET_VALUE: + case SOF_IPC_COMP_GET_DATA: + cdata = kmalloc(hdr->size, GFP_KERNEL); + if (!cdata) + return; + + /* read back full message */ + snd_sof_ipc_msg_data(sdev, NULL, cdata, hdr->size); + break; + default: + dev_err(sdev->dev, "error: unhandled component message %#x\n", msg_type); + return; + } + + snd_sof_control_notify(sdev, cdata); + + kfree(cdata); +} + /* DSP firmware has sent host a message */ void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev) { @@ -404,7 +430,9 @@ void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev) case SOF_IPC_GLB_COMPOUND: case SOF_IPC_GLB_TPLG_MSG: case SOF_IPC_GLB_PM_MSG: + break; case SOF_IPC_GLB_COMP_MSG: + ipc_comp_notification(sdev, &hdr); break; case SOF_IPC_GLB_STREAM_MSG: /* need to pass msg id into the function */ diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index dc274e63ed9a..9a8d005e75a0 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -75,6 +75,9 @@ struct snd_sof_control { struct list_head list; /* list in sdev control list */ struct snd_sof_led_control led_ctl; + + /* if true, the control's data needs to be updated from Firmware */ + bool comp_data_dirty; }; /* ASoC SOF DAPM widget */ @@ -148,6 +151,8 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, unsigned int size); int snd_sof_bytes_ext_volatile_get(struct snd_kcontrol *kcontrol, unsigned int __user *binary_data, unsigned int size); +void snd_sof_control_notify(struct snd_sof_dev *sdev, + struct sof_ipc_ctrl_data *cdata); /* * Topology. -- 2.33.0

2 1

[PATCH] ASoC: mediatek: mt8195: Fix unused initialization of pointer etdm_data
by Colin King 13 Sep '21

13 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer etdm_data is being inintialized with a value that is never read, it is later being re-assigned a new value. Remove the redundant initialization. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c index 7378e42f2766..ac591d453e1e 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c @@ -2094,7 +2094,7 @@ static int mtk_dai_etdm_set_sysclk(struct snd_soc_dai *dai, { struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8195_afe_private *afe_priv = afe->platform_priv; - struct mtk_dai_etdm_priv *etdm_data = afe_priv->dai_priv[dai->id]; + struct mtk_dai_etdm_priv *etdm_data; int dai_id; dev_dbg(dai->dev, "%s id %d freq %u, dir %d\n", -- 2.32.0

3 2

[PATCH] ASoC: mediatek: mt8195: Remove unsued irqs_lock.
by Sebastian Andrzej Siewior 13 Sep '21

13 Sep '21

irqs_lock is not used, never was. Remove irqs_lock. Fixes: 283b612429a27 ("ASoC: mediatek: implement mediatek common structure") Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Matthias Brugger <matthias.bgg(a)gmail.com> Signed-off-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d48..0c2482b7ce584 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -288,7 +288,6 @@ const struct snd_soc_dai_ops mtk_afe_fe_ops = { }; EXPORT_SYMBOL_GPL(mtk_afe_fe_ops); -static DEFINE_MUTEX(irqs_lock); int mtk_dynamic_irq_acquire(struct mtk_base_afe *afe) { int i; -- 2.33.0

2 1

[PATCH v2] ASoC: fsl_rpmsg: add soc specific data structure
by Shengjiu Wang 13 Sep '21

13 Sep '21

Each platform has different supported rates and formats, so add soc specific data for each platform. This soc specific data is attached with compatible string. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> --- changes in v2: - remove checking rpmsg->soc_data is NULL - add Reviewed-by Fabio sound/soc/fsl/fsl_rpmsg.c | 46 +++++++++++++++++++++++++++++++++++---- sound/soc/fsl/fsl_rpmsg.h | 12 ++++++++++ 2 files changed, 54 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index d60f4dac6c1b..07abad7fe372 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -138,11 +138,42 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", }; +static const struct fsl_rpmsg_soc_data imx7ulp_data = { + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mm_data = { + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U8 | + SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mn_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mp_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + static const struct of_device_id fsl_rpmsg_ids[] = { - { .compatible = "fsl,imx7ulp-rpmsg-audio"}, - { .compatible = "fsl,imx8mm-rpmsg-audio"}, - { .compatible = "fsl,imx8mn-rpmsg-audio"}, - { .compatible = "fsl,imx8mp-rpmsg-audio"}, + { .compatible = "fsl,imx7ulp-rpmsg-audio", .data = &imx7ulp_data}, + { .compatible = "fsl,imx8mm-rpmsg-audio", .data = &imx8mm_data}, + { .compatible = "fsl,imx8mn-rpmsg-audio", .data = &imx8mn_data}, + { .compatible = "fsl,imx8mp-rpmsg-audio", .data = &imx8mp_data}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); @@ -157,6 +188,13 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) if (!rpmsg) return -ENOMEM; + rpmsg->soc_data = of_device_get_match_data(&pdev->dev); + + fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; + fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; rpmsg->buffer_size = LPA_LARGE_BUFFER_SIZE; diff --git a/sound/soc/fsl/fsl_rpmsg.h b/sound/soc/fsl/fsl_rpmsg.h index 4f5b49eb18d8..b04086fbf828 100644 --- a/sound/soc/fsl/fsl_rpmsg.h +++ b/sound/soc/fsl/fsl_rpmsg.h @@ -6,6 +6,16 @@ #ifndef __FSL_RPMSG_H #define __FSL_RPMSG_H +/* + * struct fsl_rpmsg_soc_data + * @rates: supported rates + * @formats: supported formats + */ +struct fsl_rpmsg_soc_data { + int rates; + u64 formats; +}; + /* * struct fsl_rpmsg - rpmsg private data * @@ -15,6 +25,7 @@ * @pll8k: parent clock for multiple of 8kHz frequency * @pll11k: parent clock for multiple of 11kHz frequency * @card_pdev: Platform_device pointer to register a sound card + * @soc_data: soc specific data * @mclk_streams: Active streams that are using baudclk * @force_lpa: force enable low power audio routine if condition satisfy * @enable_lpa: enable low power audio routine according to dts setting @@ -27,6 +38,7 @@ struct fsl_rpmsg { struct clk *pll8k; struct clk *pll11k; struct platform_device *card_pdev; + const struct fsl_rpmsg_soc_data *soc_data; unsigned int mclk_streams; int force_lpa; int enable_lpa; -- 2.17.1

2 1

[PATCH] ASoC: rt1011: add i2s reference control for rt1011
by Jack Yu 13 Sep '21

13 Sep '21

Add i2s reference control for rt1011 amp. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1011.c | 54 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1011.h | 7 +++++ 2 files changed, 61 insertions(+) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index faff2b558687..508597866dff 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,6 +1311,57 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } +static const char * const rt1011_i2s_ref[] = { + "None", "Left Channel", "Right Channel" +}; + +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, + rt1011_i2s_ref); + +static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + int i2s_ref_ch = ucontrol->value.integer.value[0]; + + switch (i2s_ref_ch) { + case RT1011_I2S_REF_LEFT_CH: + regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); + regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); + break; + case RT1011_I2S_REF_RIGHT_CH: + regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); + regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); + break; + default: + dev_info(component->dev, "I2S Reference: Do nothing\n"); + } + + rt1011->i2s_ref = ucontrol->value.integer.value[0]; + + return 0; +} + +static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt1011->i2s_ref; + + return 0; +} + static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ SOC_ENUM("DIN Source", rt1011_din_source_enum), @@ -1349,6 +1400,9 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* R0 temperature */ SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), + /* I2S Reference */ + SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, + rt1011_i2s_ref_get, rt1011_i2s_ref_put), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 68fadc15fa8c..afb2fad94216 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,6 +654,12 @@ enum { RT1011_AIFS }; +enum { + RT1011_I2S_REF_NONE, + RT1011_I2S_REF_LEFT_CH, + RT1011_I2S_REF_RIGHT_CH, +}; + /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -692,6 +698,7 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; + unsigned int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- 2.31.1

3 7

[PATCH] ASoC: SOF: control: fix a typo in put operations for kcontrol
by Peter Ujfalusi 13 Sep '21

13 Sep '21

From: Rander Wang <rander.wang(a)intel.com> SOF_CTRL_TYPE_VALUE_CHAN_SET should be used for put operations for consistency. The current use of _GET is obviously incorrect but _GET and _SET result in the same action so there is no functional change introduced by this patch. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 456f3cd36678..58bb89af4de1 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -144,7 +144,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_VOLUME, true); return change; @@ -217,7 +217,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_SWITCH, true); @@ -266,7 +266,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_ENUM, true); -- 2.33.0

2 1

[PATCH] ASoC: amd: acp: declare and add prefix to 'bt_uart_enable' symbol
by Pierre-Louis Bossart 13 Sep '21

13 Sep '21

Sparse reports the following warning: sound/soc/amd/acp-pcm-dma.c:39:6: error: symbol 'bt_uart_enable' was not declared. Should it be static? It's not very good practice to export such symbols that can easily conflict, add the acp_ prefix and add declaration in header file. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/amd/acp-da7219-max98357a.c | 6 +++--- sound/soc/amd/acp-pcm-dma.c | 6 +++--- sound/soc/amd/acp.h | 2 ++ 3 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b3df98a9f9f3..b2065f3fe42c 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,7 +33,7 @@ static struct clk *da7219_dai_wclk; static struct clk *da7219_dai_bclk; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; -extern bool bt_uart_enable; + void *acp_soc_is_rltk_max(struct device *dev); static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -760,8 +760,8 @@ static int cz_probe(struct platform_device *pdev) "devm_snd_soc_register_card(%s) failed\n", card->name); } - bt_uart_enable = !device_property_read_bool(&pdev->dev, - "bt-pad-enable"); + acp_bt_uart_enable = !device_property_read_bool(&pdev->dev, + "bt-pad-enable"); return 0; } diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 11b3c4f39eba..1f322accd9ea 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -36,8 +36,8 @@ #define ST_MIN_BUFFER ST_MAX_BUFFER #define DRV_NAME "acp_audio_dma" -bool bt_uart_enable = true; -EXPORT_SYMBOL(bt_uart_enable); +bool acp_bt_uart_enable = true; +EXPORT_SYMBOL(acp_bt_uart_enable); static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | @@ -596,7 +596,7 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) acp_reg_write(val, acp_mmio, mmACP_SOFT_RESET); /* For BT instance change pins from UART to BT */ - if (!bt_uart_enable) { + if (!acp_bt_uart_enable) { val = acp_reg_read(acp_mmio, mmACP_BT_UART_PAD_SEL); val |= ACP_BT_UART_PAD_SELECT_MASK; acp_reg_write(val, acp_mmio, mmACP_BT_UART_PAD_SEL); diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index e5ab6c6040a6..85529ed7e5f5 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -204,4 +204,6 @@ typedef struct acp_dma_dscr_transfer { u32 reserved; } acp_dma_dscr_transfer_t; +extern bool acp_bt_uart_enable; + #endif /*__ACP_HW_H */ -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: lpass: add binding headers for digital codecs
by Srinivasa Rao Mandadapu 13 Sep '21

13 Sep '21

Add header defining for lpass internal digital codecs rx,tx and va dai node id's. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- include/dt-bindings/sound/qcom,lpass.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/dt-bindings/sound/qcom,lpass.h b/include/dt-bindings/sound/qcom,lpass.h index 7b0b80b..187af45 100644 --- a/include/dt-bindings/sound/qcom,lpass.h +++ b/include/dt-bindings/sound/qcom,lpass.h @@ -10,6 +10,11 @@ #define LPASS_DP_RX 5 +#define LPASS_CDC_DMA_RX0 6 +#define LPASS_CDC_DMA_TX3 7 +#define LPASS_CDC_DMA_VA0 8 +#define LPASS_MAX_PORTS 9 + #define LPASS_MCLK0 0 #endif /* __DT_QCOM_LPASS_H */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 7

Mackie Onyx 1640i: Long run playback results in xrun and hang.
by AreYouLoco? 13 Sep '21

13 Sep '21

Hi, I am using custom build real-time kernel 5.12.2-rt3-fw. JACK with ALSA backend on Mackie Onyx 1640i. Lenovo Thinkpad T420 with FireWire. Debian Bullseye (11.0 - stable) JACKD version 1.9.17. I am using 512 samples 3 periods @48kHz. The problem is that long run playback only around 1h (but the time differs), result in XRUN detected by JACK logs and the whole audio stops. So I have to restart JACK in order to make it work. DMESG is silent about that issue. But sometimes I get: snd_oxfw fw1.0: Detect discontinuity of CIP: E8 00 And different hex values there. Here is interesting part from the log: Sun Sep 12 22:10:12 2021: Jack: JackRequest::Notification Sun Sep 12 22:10:12 2021: Jack: **** alsa_pcm: xrun of at least 0.027 msecs Sun Sep 12 22:10:12 2021: Jack: Repreparing capture Sun Sep 12 22:10:12 2021: Jack: Repreparing playback Sun Sep 12 22:10:12 2021: Jack: ALSA XRun wait_status = 0 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackClient::ClientNotify ref = 2 name = dbusapi notify = 3 Sun Sep 12 22:10:12 2021: Jack: JackClient::kXRunCallback Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:28 2021: ERROR: ALSA: poll time out, polled for 15998655 usecs, Retrying with a recovery, retry cnt = 1 Sun Sep 12 22:10:44 2021: ERROR: ALSA: poll time out, polled for 15998654 usecs, Retrying with a recovery, retry cnt = 2 Sun Sep 12 22:11:00 2021: ERROR: ALSA: poll time out, polled for 15998659 usecs, Retrying with a recovery, retry cnt = 3 Sun Sep 12 22:11:16 2021: ERROR: ALSA: poll time out, polled for 15998649 usecs, Retrying with a recovery, retry cnt = 4 Anyone have idea what might be the cause of this XRUN? Or how to debug that? Its reproducible on my setup. Another log catched the same issue: Sun Sep 12 16:50:44 2021: Jack: **** alsa_pcm: xrun of at least 0.030 msecs Sun Sep 12 16:50:44 2021: Jack: Repreparing capture Sun Sep 12 16:50:44 2021: Jack: Repreparing playback Sun Sep 12 16:50:44 2021: Jack: ALSA XRun wait_status = 0 Sun Sep 12 16:50:44 2021: Jack: JackRequest::Notification Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackClient::ClientNotify ref = 2 name = dbusapi notify = 3 Sun Sep 12 16:50:44 2021: Jack: JackClient::kXRunCallback Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:51:00 2021: ERROR: ALSA: poll time out, polled for 15998366 usecs, Retrying with a recovery, retry cnt = 1 Sun Sep 12 16:51:16 2021: ERROR: ALSA: poll time out, polled for 15998353 usecs, Retrying with a recovery, retry cnt = 2 Sun Sep 12 16:51:32 2021: ERROR: ALSA: poll time out, polled for 15998352 usecs, Retrying with a recovery, retry cnt = 3 Sun Sep 12 16:51:48 2021: ERROR: ALSA: poll time out, polled for 15998367 usecs, Retrying with a recovery, retry cnt = 4 Sun Sep 12 16:52:04 2021: ERROR: ALSA: poll time out, polled for 15998353 usecs, Retrying with a recovery, retry cnt = 5 Sun Sep 12 16:52:20 2021: ERROR: ALSA: poll time out, polled for 15998371 usecs, Reached max retry cnt = 5, Exiting Sun Sep 12 16:52:20 2021: ERROR: JackAudioDriver::ProcessAsync: read error, stopping... Sun Sep 12 16:52:20 2021: Jack: JackPosixThread::ThreadHandler : exit I am willing to provide additional info. Thank you in advance, AreYouLoco?

1 0

[PATCH] ALSA: oxfw: fix transmission method for Loud models based on OXFW971
by Takashi Sakamoto 13 Sep '21

13 Sep '21

Loud Technologies Mackie Onyx 1640i (former model) is identified as the model which uses OXFW971. The analysis of packet dump shows that it transfers events in blocking method of IEC 61883-6, however the default behaviour of ALSA oxfw driver is for non-blocking method. This commit adds code to detect it assuming that all of loud models based on OXFW971 have such quirk. It brings no functional change except for alignment rule of PCM buffer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b..daf731364695 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } -- 2.30.2

2 1

[PATCH AUTOSEL 4.4 29/35] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 13 Sep '21

13 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index d0d338533eb6..64079423b298 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -134,7 +134,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 2

[PATCH AUTOSEL 5.13 176/219] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 11 Sep '21

11 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 905c7965f653..5db2f4865bbb 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 2

Raw ALSA PCM
by Ryan McClue 11 Sep '21

11 Sep '21

Hello, I'm trying to play a simple PCM buffer with ALSA without asoundlib. This has proven very difficult as there are very few resources that show usage without asoundlib. Through my investigations I was fortunate enough to obtain a gist that attempts to do this: https://gist.github.com/takaswie/d8a9553c230971d974a3dfc759b2ff67 Unfortunately this application does not work. I tested the application using the virtual snd-dummy driver and the application produced the same error. So, it is most likely an application error, rather than an Intel HDA error. I'm emailing to ask for help in getting the program to work. I'm really lost as to what should be changed in the code. I'm sure that a working example of this will be useful for many programmers. Thank you

1 0

ASoC/ALSA Out of Memory Issue
by Alex Roberts 10 Sep '21

10 Sep '21

Hello, I am developing a dummy codec to interface with an 8-channel, 24-bit ADC. I've got it working on an NXP imx8m through the fsl_sai driver on kernel 5.4.85. I can capture all 8 channels at varying sample rates using arecord, and I've verified correct data capture via opening the resulting .wav file in Audacity. The problem I am having is that occasionally, upon starting arecord - after a fresh power cycle - I get an out of memory error. Other times I get an out of memory after a non-deterministic period of capture. Starting capture again also reports out of memory, but if I wait several minutes and start capture it will start recording again. A power cycle usually helps, but as stated earlier, not 100% of the time. I'm trying to track down where the oom error is coming from, but haven't had much luck. My colleague tried running arecord with valgrind to check for memory leaks and nothing of note was observed. My suspicion is there's something going on with allocated memory for DMA, like fragmentation starts to happen and it can't get a contiguous region for operation. Reserving a larger pool - either via device tree or kernel cmdline arguments in the bootoader - did not seem to help. Another thought is that it's a boundary/alignment issue due to the 24-bit data, and the error is the result of trying to allocate a chunk of memory for DMA that doesn't align. I'm very new to ALSA dev with some exposure to kernel dev in general, so please correct me if I'm wrong or completely mis-understanding something. Any suggestions on where I should / how I can debug this memory error? Thanks, Alex. PS: Previously sent this to just alsa-devel mailing list on 7/21, but never saw it show up in the archives. Here is more info since then: The goal is 8-channel, 96k sampling rate. I've reduced sampling rate and still have the issue. Reducing down to 4-channels helps, but haven't tested long term enough to evaluate by how much. Narrowed it down to device_prep_dma_cyclic(..) returning NULL within dmaengine_prep_dma_cyclic(..)..... still tracing through source to learn exactly what is going on.

2 3

[PATCH] ASoC: mediatek: common: handle NULL case in suspend/resume function
by Trevor Wu 10 Sep '21

10 Sep '21

When memory allocation for afe->reg_back_up fails, reg_back_up can't be used. Keep the suspend/resume flow but skip register backup when afe->reg_back_up is NULL, in case illegal memory access happens. Fixes: 283b612429a2 ("ASoC: mediatek: implement mediatek common structure") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d4..e95c7c018e7d 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component) devm_kcalloc(dev, afe->reg_back_up_list_num, sizeof(unsigned int), GFP_KERNEL); - for (i = 0; i < afe->reg_back_up_list_num; i++) - regmap_read(regmap, afe->reg_back_up_list[i], - &afe->reg_back_up[i]); + if (afe->reg_back_up) { + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + } afe->suspended = true; afe->runtime_suspend(dev); @@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component) afe->runtime_resume(dev); - if (!afe->reg_back_up) + if (!afe->reg_back_up) { dev_dbg(dev, "%s no reg_backup\n", __func__); - - for (i = 0; i < afe->reg_back_up_list_num; i++) - mtk_regmap_write(regmap, afe->reg_back_up_list[i], - afe->reg_back_up[i]); + } else { + for (i = 0; i < afe->reg_back_up_list_num; i++) + mtk_regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + } afe->suspended = false; return 0; -- 2.18.0

2 1

[PATCH for-5.15 0/5] ASoC: fsl: register platform component before registering cpu dai
by Shengjiu Wang 10 Sep '21

10 Sep '21

There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. This patch set is to fix this issue for SAI, ESAI, MICFIL, SPDIF, XCVR drivers. Shengjiu Wang (5): ASoC: fsl_sai: register platform component before registering cpu dai ASoC: fsl_esai: register platform component before registering cpu dai ASoC: fsl_micfil: register platform component before registering cpu dai ASoC: fsl_spdif: register platform component before registering cpu dai ASoC: fsl_xcvr: register platform component before registering cpu dai sound/soc/fsl/fsl_esai.c | 16 ++++++++++------ sound/soc/fsl/fsl_micfil.c | 15 ++++++++++----- sound/soc/fsl/fsl_sai.c | 14 +++++++++----- sound/soc/fsl/fsl_spdif.c | 14 +++++++++----- sound/soc/fsl/fsl_xcvr.c | 15 ++++++++++----- 5 files changed, 48 insertions(+), 26 deletions(-) -- 2.17.1

3 9

[PATCH] ALSA: usb-audio: Unify mixer resume and reset_resume procedure
by Takashi Iwai 10 Sep '21

10 Sep '21

USB-audio driver assumes that the normal resume would preserve the device configuration while reset_resume wouldn't, and tries to restore the mixer elements only at reset_resume callback. However, this seems too naive, and some devices do behave differently, resetting the volume at the normal resume; this resulted in the inconsistent volume that surprised users. This patch changes the mixer resume code to handle both the normal and reset resume in the same way, always restoring the original mixer element values. This allows us to unify the both callbacks as well as dropping the no longer used reset_resume field, which ends up with a good code reduction. A slight behavior change by this patch is that now we assign restore_mixer_value() as the default resume callback, and the function is no longer called at reset-resume when the resume callback is overridden by the quirk function. That is, if needed, the quirk resume function would have to handle similarly as restore_mixer_value() by itself. Reported-by: En-Shuo Hsu <enshuo(a)chromium.org> Cc: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsM… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 18 ++++-------------- sound/usb/mixer.c | 26 ++++---------------------- sound/usb/mixer.h | 3 +-- sound/usb/mixer_quirks.c | 2 +- 4 files changed, 10 insertions(+), 39 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd570a42f043..1764b9302d46 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1100,20 +1100,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 43bc59575a6e..a2ce535df14b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3653,33 +3653,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3700,7 +3683,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 876bbc9a71ad..98ea24d91d80 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a66ce0375fd9..46082dc57be0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); -- 2.26.2

1 0

A portion of USB Headsets loses previous sound volume setting after a suspend resume
by En-Shuo Hsu 10 Sep '21

10 Sep '21

Hi We recently found that some USB headsets may fall back to their full volume after a suspend and resume. We think the issue is caused by the logic of mixer_ctl_feature_put <https://github.com/torvalds/linux/blob/a3fa7a101dcff93791d1b1bdb3affcad1410…> in sound/usb/mixer.c: err = snd_usb_get_cur_mix_value(cval, c + 1, cnt, &oval); if (err < 0) return filter_error(cval, err); val = ucontrol->value.integer.value[cnt]; val = get_abs_value(cval, val); if (oval != val) { snd_usb_set_cur_mix_value(cval, c + 1, cnt, val); changed = 1; } The existing codes get the existing mixer control value and ignore the set if the val doesn't change. However, in the suspend and resume case, the USB headset's control value is actually changed. Removing the cache logic is a potential fix, but a better solution may be to properly handle the suspend resume scenario in snd_usb_get_cur_mix_value. We may need to mark the cache in usb_mixer_elem_info to be dirty. The issue is verified to be reproduced with Dell WH3022 and Logitech USB Headset H340 by: 1. Boot to OS. 2. Plug in the headset and check sound output. 3. Play an audio/video and keep with low volume. 4. Suspend. 5. Resume. 6. When audio/video is played, the headset's sound output can't keep the original volume. --> issue " Would like to know your thoughts on this issue.

3 3

[PATCH v3] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Randy Dunlap 10 Sep '21

10 Sep '21

On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Fixes: 3951e4aae2ce ("ASoC: atmel-pcm: dma support based on pcm dmaengine") Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Fixes: 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Cc: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- v2: rebase & resend; drop Cc: to Bo Shen <voice.shen(a)atmel.com> # bounced v3: drop "depends on HAS_DMA" (as suggested by Codrin; Thanks) --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) --- linux-next-20210707.orig/sound/soc/atmel/Kconfig +++ linux-next-20210707/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool

5 10

[PATCH] ASoC: mediatek: add required config dependency
by Trevor Wu 10 Sep '21

10 Sep '21

Because SND_SOC_MT8195 depends on COMPILE_TEST, it's possible to build MT8195 driver in many different config combinations. Add all dependent config for SND_SOC_MT8195 in case some errors happen when COMPILE_TEST is enabled. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> --- sound/soc/mediatek/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5a2f4667d50b..81ad2dcee9eb 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -1,6 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SOC_MEDIATEK tristate + select REGMAP_MMIO config SND_SOC_MT2701 tristate "ASoC support for Mediatek MT2701 chip" @@ -188,7 +189,9 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 config SND_SOC_MT8195 tristate "ASoC support for Mediatek MT8195 chip" depends on ARCH_MEDIATEK || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_MEDIATEK + select MFD_SYSCON if SND_SOC_MT6359 help This adds ASoC platform driver support for Mediatek MT8195 chip that can be used with other codecs. -- 2.18.0

3 2

[PATCH v2 0/2] Add code to manage DSP related clocks
by Daniel Baluta 09 Sep '21

09 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and Device Tree communities then it will be merged to SOF tree and then merged into ALSA tree. DSP node on the Linux kernel side must also take care of enabling DAI/DMA related clocks. By design we choose to manage DAI/DMA clocks from the kernel side because of the architecture of some i.MX8 boards. Clocks are handled by a special M4 core which runs a special firmware called SCFW (System Controler firmware). This communicates with A cores running Linux via a special Messaging Unit and implements a custom API which is already implemented by the Linux kernel i.MX clocks implementation. Note that these clocks are optional. We can use the DSP without them. Changes since v1: - used clk bulk API as suggested by mark Daniel Baluta (2): ASoC: SOF: imx: Add code to manage DSP related clocks dt-bindings: dsp: fsl: Add DSP optional clocks documentation .../devicetree/bindings/dsp/fsl,dsp.yaml | 33 ++++++++++++++ sound/soc/sof/imx/imx-common.c | 44 +++++++++++++++++++ sound/soc/sof/imx/imx-common.h | 13 ++++++ sound/soc/sof/imx/imx8.c | 37 ++++++++++++++++ sound/soc/sof/imx/imx8m.c | 34 ++++++++++++++ 5 files changed, 161 insertions(+) -- 2.27.0

4 8

[PATCH] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 09 Sep '21

09 Sep '21

Left/Right speakers and 4ch tdm playback add echo reference capture add bt offload support add DMI_OEM_STRING for board varients Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 155 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 69 ++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 13 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 263 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 61b71d6c44cf..2c24d8a929a2 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -470,6 +470,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1015 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..d2c236197f2e 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,161 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; +EXPORT_SYMBOL_NS(max_98390_tt_dapm_routes, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME) || + !strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME) || + !strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, + ARRAY_SIZE(max_98373_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "Speaker Left, Right map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Tweeter Speaker Left, Right map addition failed: %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..412b9dd06bad 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern struct snd_soc_ops max_98390_ops; +extern const struct snd_soc_dapm_route max_98390_dapm_routes[]; +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f096bd6d69be..e46438cca643 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,8 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(24) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -176,6 +178,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-ADL_MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-ADL_MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -459,6 +491,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -745,6 +778,21 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + links[id].init = max_98390_spk_codec_init; + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + } + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -881,6 +929,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1007,6 +1063,18 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { } }; MODULE_DEVICE_TABLE(platform, board_ids); @@ -1026,6 +1094,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..eee73a91b7ff 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-ADL_MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-rt5682-ssp0-max98390-ssp2.tplg", + }, + {} }; -- 2.20.1

3 4

[PATCH] MAINTAINERS: fix update references to stm32 audio bindings
by Arnaud Pouliquen 09 Sep '21

09 Sep '21

The 00d38fd8d2524 ("MAINTAINERS: update references to stm32 audio bindings") commit update the bindings reference, by removing bindings/sound/st,stm32-adfsdm.txt, to set the new reference to bindings/iio/adc/st,stm32-*.yaml. This leads to "get_maintainer finds" the match for the Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml, but also to the IIO bindings Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml And The commit fixes only a part of the problem: Documentation/devicetree/bindings/sound/st,stm32-*.txt file have been also moved to yaml. Update references to include all stm32 audio bindings file and exclude the st,stm32-adc.yaml bindings file. cc: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> Fixes: 0d38fd8d2524 ("MAINTAINERS: update references to stm32 audio bindings") Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 3a9a7cbf9456..3f6dc482660e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -17886,7 +17886,8 @@ M: Olivier Moysan <olivier.moysan(a)foss.st.com> M: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/iio/adc/st,stm32-*.yaml +F: Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +F: Documentation/devicetree/bindings/sound/st,stm32-*.yaml F: sound/soc/stm/ STM32 TIMER/LPTIMER DRIVERS -- 2.17.1

2 1

[PATCH] ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic
by Pierre-Louis Bossart 09 Sep '21

09 Sep '21

The SoundWire BEs make use of 'stream' functions for .prepare and .trigger. These functions will in turn force a Bank Switch, which implies a wait operation. Mark SoundWire BEs as nonatomic for consistency, but keep all other types of BEs as is. The initialization of .nonatomic is done outside of the create_sdw_dailink helper to avoid adding more parameters to deal with a single exception to the rule that BEs are atomic. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6602eda89e8e..6b06248a9327 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -929,6 +929,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, cpus + *cpu_id, cpu_dai_num, codecs, codec_num, NULL, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + dai_links[*be_index].nonatomic = true; ret = set_codec_init_func(card, link, dai_links + (*be_index)++, playback, group_id); -- 2.25.1

2 1

[PATCH v2] ASoC: qcom: lpass-platform: Reset irq clear reg post handling interrupts
by Srinivasa Rao Mandadapu 09 Sep '21

09 Sep '21

Update interrupt clear register with reset value after addressing all interrupts. This is to fix playback or capture hanging issue in simultaneous playback and capture usecase. Fixes: 4f629e4b8705f ("ASoC: qcom: Add ability to handle interrupts per dma channel") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v1: -- Update comments Header information with fixes tag sound/soc/qcom/lpass-platform.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f9df76d37858..1a0a4b0b1a03 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -749,6 +749,12 @@ static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data) } } + rv = regmap_write(drvdata->lpaif_map, LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST), 0x0); + if (rv) { + pr_err("error writing to irqstat reg: %d\n", rv); + return IRQ_NONE; + } + return IRQ_HANDLED; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH AUTOSEL 4.9 39/48] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 1b6dedfc33e3..6303b2d3982d 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -135,7 +135,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 1

[PATCH AUTOSEL 4.14 44/59] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 96f7facd0fa0..c877326cb0a6 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -135,7 +135,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 1

[PATCH AUTOSEL 4.19 57/74] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 682ee41ec75c..501ac836777a 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -135,7 +135,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 1

[PATCH AUTOSEL 5.4 084/109] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 7d59846808b5..216e88624c5f 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 1

[PATCH AUTOSEL 5.10 141/176] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 255b4d528a66..2784611196f0 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 1

[PATCH AUTOSEL 4.9 28/48] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c17f262f0834..18e002fef401 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -172,9 +172,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -182,19 +179,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -236,6 +237,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -243,6 +245,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 54/59] ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
by Sasha Levin 09 Sep '21

09 Sep '21

From: Xiaotan Luo <lxt(a)rock-chips.com> [ Upstream commit 1bf56843e664eef2525bdbfae6a561e98910f676 ] - DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC - DSP_B: PCM no delay mode, L data MSB during FRM LRC Signed-off-by: Xiaotan Luo <lxt(a)rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950562-14281-3-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 93a4829f80cc..8d1a7114f6c2 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -235,12 +235,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; @@ -259,12 +259,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 53/59] ASoC: rockchip: i2s: Fix regmap_ops hang
by Sasha Levin 09 Sep '21

09 Sep '21

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> [ Upstream commit 53ca9b9777b95cdd689181d7c547e38dc79adad0 ] API 'set_fmt' maybe called when PD is off, in the situation, any register access will hang the system. so, enable PD before r/w register. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950520-14190-4-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0e07e3dea7de..93a4829f80cc 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -188,7 +188,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -201,7 +203,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -215,7 +218,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -238,7 +242,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -261,12 +266,16 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 30/59] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4a76b099a508..e389ecf06e63 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -226,9 +226,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -236,19 +233,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -290,6 +291,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -297,6 +299,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 69/74] ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
by Sasha Levin 09 Sep '21

09 Sep '21

From: Xiaotan Luo <lxt(a)rock-chips.com> [ Upstream commit 1bf56843e664eef2525bdbfae6a561e98910f676 ] - DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC - DSP_B: PCM no delay mode, L data MSB during FRM LRC Signed-off-by: Xiaotan Luo <lxt(a)rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950562-14281-3-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index cab381c9dea1..c6d11a59310f 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -236,12 +236,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; @@ -260,12 +260,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 68/74] ASoC: rockchip: i2s: Fix regmap_ops hang
by Sasha Levin 09 Sep '21

09 Sep '21

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> [ Upstream commit 53ca9b9777b95cdd689181d7c547e38dc79adad0 ] API 'set_fmt' maybe called when PD is off, in the situation, any register access will hang the system. so, enable PD before r/w register. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950520-14190-4-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b86f76c3598c..cab381c9dea1 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -189,7 +189,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -202,7 +204,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -216,7 +219,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -239,7 +243,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -262,12 +267,16 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 34/74] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 186c0ee059da..c4d19b88d17d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -293,9 +293,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -303,19 +300,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -357,6 +358,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -364,6 +366,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 098/109] ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
by Sasha Levin 09 Sep '21

09 Sep '21

From: Xiaotan Luo <lxt(a)rock-chips.com> [ Upstream commit 1bf56843e664eef2525bdbfae6a561e98910f676 ] - DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC - DSP_B: PCM no delay mode, L data MSB during FRM LRC Signed-off-by: Xiaotan Luo <lxt(a)rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950562-14281-3-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index f48b146cd96a..086c90e09577 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -233,12 +233,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; @@ -257,12 +257,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 097/109] ASoC: rockchip: i2s: Fix regmap_ops hang
by Sasha Levin 09 Sep '21

09 Sep '21

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> [ Upstream commit 53ca9b9777b95cdd689181d7c547e38dc79adad0 ] API 'set_fmt' maybe called when PD is off, in the situation, any register access will hang the system. so, enable PD before r/w register. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950520-14190-4-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 61c984f10d8e..f48b146cd96a 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -186,7 +186,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -199,7 +201,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -213,7 +216,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -236,7 +240,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -259,12 +264,16 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 086/109] ASoC: Intel: Skylake: Fix passing loadable flag for module
by Sasha Levin 09 Sep '21

09 Sep '21

From: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> [ Upstream commit c5ed9c547cba1dc1238c6e8a0c290fd62ee6e127 ] skl_get_module_info() tries to set mconfig->module->loadable before mconfig->module has been assigned thus flag was always set to false and driver did not try to load module binaries. Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 7f287424af9b..439dd4ba690c 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1333,21 +1333,6 @@ static int skl_get_module_info(struct skl_dev *skl, return -EIO; } - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - if (mconfig->module) - mconfig->module->loadable = module->is_loadable; - ret = 0; - break; - } - } - - if (ret) - return ret; - - uuid_mod = &module->uuid; - ret = -EIO; for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; @@ -1357,10 +1342,18 @@ static int skl_get_module_info(struct skl_dev *skl, break; } } + if (skl->nr_modules && ret) return ret; + ret = -EIO; list_for_each_entry(module, &skl->uuid_list, list) { + if (guid_equal(uuid_mod, &module->uuid)) { + mconfig->id.module_id = module->id; + mconfig->module->loadable = module->is_loadable; + ret = 0; + } + for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; if (guid_equal(&pin_id->mod_uuid, &module->uuid)) @@ -1374,7 +1367,7 @@ static int skl_get_module_info(struct skl_dev *skl, } } - return 0; + return ret; } static int skl_populate_modules(struct skl_dev *skl) -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 085/109] ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER
by Sasha Levin 09 Sep '21

09 Sep '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e4e0633bcadc950b4b4af06c7f1bb7f7e3e86321 ] KeyPhrasebuffer, Mixin and Mixout modules configuration is described by firmware's basic module configuration structure. There are no extended parameters required. Update functions taking part in building INIT_INSTANCE IPC payload to reflect that. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 047/109] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c67b86e2d0c0..7830d014d924 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -284,9 +284,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -294,19 +291,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -348,6 +349,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -355,6 +357,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 002/109] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Sasha Levin 09 Sep '21

09 Sep '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 6c5c659dfe3f02e08054a6c20019e3886618b512 ] On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Link: https://lore.kernel.org/r/20210707214752.3831-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 71f2d42188c4..51e75b781968 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 132/252] ALSA: pci: cs46xx: Fix set up buffer type properly
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4d9e9153f1c64d91a125c6967bc0bfb0bb653ea0 ] CS46xx driver switches the buffer depending on the number of periods, and in some cases it switches to the own buffer without updating the buffer type properly. This may cause a problem with the mmap on exotic architectures that require the own mmap call for the coherent DMA buffer. This patch addresses the potential breakage by replacing the buffer setup with the proper macro. It also simplifies the source code, too. Link: https://lore.kernel.org/r/20210809071829.22238-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 30 ++++++++---------------------- 1 file changed, 8 insertions(+), 22 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1e1eb17f8e07..d43927dcd61e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1121,9 +1121,7 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, if (params_periods(hw_params) == CS46XX_FRAGS) { if (runtime->dma_area != cpcm->hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = cpcm->hw_buf.area; - runtime->dma_addr = cpcm->hw_buf.addr; - runtime->dma_bytes = cpcm->hw_buf.bytes; + snd_pcm_set_runtime_buffer(substream, &cpcm->hw_buf); #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -1143,11 +1141,8 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, #endif } else { - if (runtime->dma_area == cpcm->hw_buf.area) { - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; - } + if (runtime->dma_area == cpcm->hw_buf.area) + snd_pcm_set_runtime_buffer(substream, NULL); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) { #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -1196,9 +1191,7 @@ static int snd_cs46xx_playback_hw_free(struct snd_pcm_substream *substream) if (runtime->dma_area != cpcm->hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; + snd_pcm_set_runtime_buffer(substream, NULL); return 0; } @@ -1287,16 +1280,11 @@ static int snd_cs46xx_capture_hw_params(struct snd_pcm_substream *substream, if (runtime->periods == CS46XX_FRAGS) { if (runtime->dma_area != chip->capt.hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = chip->capt.hw_buf.area; - runtime->dma_addr = chip->capt.hw_buf.addr; - runtime->dma_bytes = chip->capt.hw_buf.bytes; + snd_pcm_set_runtime_buffer(substream, &chip->capt.hw_buf); substream->ops = &snd_cs46xx_capture_ops; } else { - if (runtime->dma_area == chip->capt.hw_buf.area) { - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; - } + if (runtime->dma_area == chip->capt.hw_buf.area) + snd_pcm_set_runtime_buffer(substream, NULL); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -1313,9 +1301,7 @@ static int snd_cs46xx_capture_hw_free(struct snd_pcm_substream *substream) if (runtime->dma_area != chip->capt.hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; + snd_pcm_set_runtime_buffer(substream, NULL); return 0; } -- 2.30.2

2 1

[PATCH AUTOSEL 5.14 198/252] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 905c7965f653..5db2f4865bbb 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

2 1

[PATCH AUTOSEL 5.10 157/176] ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
by Sasha Levin 09 Sep '21

09 Sep '21

From: Xiaotan Luo <lxt(a)rock-chips.com> [ Upstream commit 1bf56843e664eef2525bdbfae6a561e98910f676 ] - DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC - DSP_B: PCM no delay mode, L data MSB during FRM LRC Signed-off-by: Xiaotan Luo <lxt(a)rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950562-14281-3-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 43522d7701e9..fa84ec695b52 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -233,12 +233,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; @@ -257,12 +257,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 156/176] ASoC: rockchip: i2s: Fix regmap_ops hang
by Sasha Levin 09 Sep '21

09 Sep '21

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> [ Upstream commit 53ca9b9777b95cdd689181d7c547e38dc79adad0 ] API 'set_fmt' maybe called when PD is off, in the situation, any register access will hang the system. so, enable PD before r/w register. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950520-14190-4-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 593299675b8c..43522d7701e9 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -186,7 +186,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -199,7 +201,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -213,7 +216,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -236,7 +240,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -259,12 +264,16 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 144/176] ASoC: Intel: Skylake: Fix passing loadable flag for module
by Sasha Levin 09 Sep '21

09 Sep '21

From: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> [ Upstream commit c5ed9c547cba1dc1238c6e8a0c290fd62ee6e127 ] skl_get_module_info() tries to set mconfig->module->loadable before mconfig->module has been assigned thus flag was always set to false and driver did not try to load module binaries. Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index bbe8d782e0af..b1897a057397 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1318,21 +1318,6 @@ static int skl_get_module_info(struct skl_dev *skl, return -EIO; } - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - if (mconfig->module) - mconfig->module->loadable = module->is_loadable; - ret = 0; - break; - } - } - - if (ret) - return ret; - - uuid_mod = &module->uuid; - ret = -EIO; for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; @@ -1342,10 +1327,18 @@ static int skl_get_module_info(struct skl_dev *skl, break; } } + if (skl->nr_modules && ret) return ret; + ret = -EIO; list_for_each_entry(module, &skl->uuid_list, list) { + if (guid_equal(uuid_mod, &module->uuid)) { + mconfig->id.module_id = module->id; + mconfig->module->loadable = module->is_loadable; + ret = 0; + } + for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; if (guid_equal(&pin_id->mod_uuid, &module->uuid)) @@ -1359,7 +1352,7 @@ static int skl_get_module_info(struct skl_dev *skl, } } - return 0; + return ret; } static int skl_populate_modules(struct skl_dev *skl) -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 143/176] ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER
by Sasha Levin 09 Sep '21

09 Sep '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e4e0633bcadc950b4b4af06c7f1bb7f7e3e86321 ] KeyPhrasebuffer, Mixin and Mixout modules configuration is described by firmware's basic module configuration structure. There are no extended parameters required. Update functions taking part in building INIT_INSTANCE IPC payload to reflect that. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 142/176] soundwire: intel: fix potential race condition during power down
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ea6942dad4b2a7e1735aa0f10f3d0b04b847750f ] The power down sequence sets the link_up flag as false outside of the mutex_lock. This is potentially unsafe. In additional the flow in that sequence can be improved by first testing if the link was powered, setting the link_up flag as false and proceeding with the power down. In case the CPA bits cannot be cleared, we only flag an error since we cannot deal with interrupts any longer. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20210818024954.16873-2-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 6a1e862b16c3..dad4326a2a71 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -537,12 +537,14 @@ static int intel_link_power_down(struct sdw_intel *sdw) mutex_lock(sdw->link_res->shim_lock); - intel_shim_master_ip_to_glue(sdw); - if (!(*shim_mask & BIT(link_id))) dev_err(sdw->cdns.dev, "%s: Unbalanced power-up/down calls\n", __func__); + sdw->cdns.link_up = false; + + intel_shim_master_ip_to_glue(sdw); + *shim_mask &= ~BIT(link_id); if (!*shim_mask) { @@ -559,20 +561,21 @@ static int intel_link_power_down(struct sdw_intel *sdw) link_control &= spa_mask; ret = intel_clear_bit(shim, SDW_SHIM_LCTL, link_control, cpa_mask); + if (ret < 0) { + dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); + + /* + * we leave the sdw->cdns.link_up flag as false since we've disabled + * the link at this point and cannot handle interrupts any longer. + */ + } } link_control = intel_readl(shim, SDW_SHIM_LCTL); mutex_unlock(sdw->link_res->shim_lock); - if (ret < 0) { - dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); - - return ret; - } - - sdw->cdns.link_up = false; - return 0; + return ret; } static void intel_shim_sync_arm(struct sdw_intel *sdw) -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 166/252] ALSA: hda: Drop workaround for a hang at shutdown again
by Sasha Levin 09 Sep '21

09 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8fc8e903156f42c66245838441d03607e9067381 ] The commit 0165c4e19f6e ("ALSA: hda: Fix hang during shutdown due to link reset") modified the shutdown callback of the HD-audio controller for working around a hang. Meanwhile, the actual culprit of the hang was identified to be the leftover active codecs that may interfere with the powered down controller somehow, but we took a minimal fix approach for 5.14, and that was the commit above. Now, since the codec drivers go runtime-suspend at shutdown for 5.15, we can revert the change and make sure that the full runtime-suspend is performed at shutdown of HD-audio controller again. This patch essentially reverts the commit above to restore the behavior. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214045 Link: https://lore.kernel.org/r/20210817075630.7115-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0062c18b646a..0322b289505e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,11 +883,10 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) +static void azx_shutdown_chip(struct azx *chip) { azx_stop_chip(chip); - if (!skip_link_reset) - azx_enter_link_reset(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -896,11 +895,6 @@ static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); -static void azx_shutdown_chip(struct azx *chip) -{ - __azx_shutdown_chip(chip, false); -} - static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2391,7 +2385,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - __azx_shutdown_chip(chip, true); + azx_shutdown_chip(chip); } /* PCI IDs */ -- 2.30.2

2 1

[PATCH AUTOSEL 5.10 072/176] ASoC: Intel: update sof_pcm512x quirks
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 22414cade8dfec25ab94df52b3a4f7aa8edb6120 ] The default SOF topology enables SSP capture and DMICs, even though both of these hardware capabilities are not always available in hardware (specific versions of HiFiberry and DMIC kit needed). For the SSP capture, this leads to annoying "SP5-Codec: ASoC: no backend capture" and "streamSSP5-Codec: ASoC: no users capture at close - state 0" errors. Update the quirks to match what the topology needs, which also allows for the ability to remove SSP capture and DMIC support. BugLink: https://github.com/thesofproject/linux/issues/3061 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802152151.15832-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_pcm512x.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index d2b0456236c7..bdd671f07659 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -26,11 +26,16 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define SOF_PCM512X_ENABLE_SSP_CAPTURE BIT(4) +#define SOF_PCM512X_ENABLE_DMIC BIT(5) #define IDISP_CODEC_MASK 0x4 /* Default: SSP5 */ -static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); +static unsigned long sof_pcm512x_quirk = + SOF_PCM512X_SSP_CODEC(5) | + SOF_PCM512X_ENABLE_SSP_CAPTURE | + SOF_PCM512X_ENABLE_DMIC; static bool is_legacy_cpu; @@ -245,8 +250,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].dpcm_playback = 1; /* * capture only supported with specific versions of the Hifiberry DAC+ - * links[id].dpcm_capture = 1; */ + if (sof_pcm512x_quirk & SOF_PCM512X_ENABLE_SSP_CAPTURE) + links[id].dpcm_capture = 1; links[id].no_pcm = 1; links[id].cpus = &cpus[id]; links[id].num_cpus = 1; @@ -381,6 +387,9 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK; + if (!(sof_pcm512x_quirk & SOF_PCM512X_ENABLE_DMIC)) + dmic_be_num = 0; + /* compute number of dai links */ sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 070/176] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ca14730232ba..43ee3d095a1b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -286,9 +286,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -296,19 +293,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -350,6 +351,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -357,6 +359,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 003/176] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Sasha Levin 09 Sep '21

09 Sep '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 6c5c659dfe3f02e08054a6c20019e3886618b512 ] On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Link: https://lore.kernel.org/r/20210707214752.3831-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 142373ec411a..89210048e6c2 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 197/219] ASoC: soc-pcm: protect BE dailink state changes in trigger
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 0c75fc7193387776c10f7c7b440d93496e3d5e21 ] When more than one FE is connected to a BE, e.g. in a mixing use case, the BE can be triggered multiple times when the FE are opened/started concurrently. This race condition is problematic in the case of SoundWire BE dailinks, and this is not desirable in a general case. The code carefully checks when the BE can be stopped or hw_free'ed, but the trigger code does not use any mutual exclusion. Fix by using the same spinlock already used to check FE states, and set the state before the trigger. In case of errors, the initial state will be restored. This patch does not change how the triggers are handled, it only makes sure the states are handled in critical sections. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Message-Id: <20210817164054.250028-2-pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 103 ++++++++++++++++++++++++++++++++++++-------- 1 file changed, 85 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d1c570ca21ea..b944f56a469a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2001,6 +2001,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_pcm_runtime *be; struct snd_soc_dpcm *dpcm; int ret = 0; + unsigned long flags; + enum snd_soc_dpcm_state state; for_each_dpcm_be(fe, stream, dpcm) { struct snd_pcm_substream *be_substream; @@ -2017,76 +2019,141 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + spin_lock_irqsave(&fe->card->dpcm_lock, flags); if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_RESUME: - if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: + spin_lock_irqsave(&fe->card->dpcm_lock, flags); if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } } -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 196/219] ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
by Sasha Levin 09 Sep '21

09 Sep '21

From: Xiaotan Luo <lxt(a)rock-chips.com> [ Upstream commit 1bf56843e664eef2525bdbfae6a561e98910f676 ] - DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC - DSP_B: PCM no delay mode, L data MSB during FRM LRC Signed-off-by: Xiaotan Luo <lxt(a)rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950562-14281-3-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index d020cb2bdbc4..ac9980ed266e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -233,12 +233,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; @@ -257,12 +257,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 195/219] ASoC: rockchip: i2s: Fix regmap_ops hang
by Sasha Levin 09 Sep '21

09 Sep '21

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> [ Upstream commit 53ca9b9777b95cdd689181d7c547e38dc79adad0 ] API 'set_fmt' maybe called when PD is off, in the situation, any register access will hang the system. so, enable PD before r/w register. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950520-14190-4-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0740764e7f71..d020cb2bdbc4 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -186,7 +186,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -199,7 +201,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -213,7 +216,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -236,7 +240,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -259,12 +264,16 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 179/219] ASoC: Intel: Skylake: Fix passing loadable flag for module
by Sasha Levin 09 Sep '21

09 Sep '21

From: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> [ Upstream commit c5ed9c547cba1dc1238c6e8a0c290fd62ee6e127 ] skl_get_module_info() tries to set mconfig->module->loadable before mconfig->module has been assigned thus flag was always set to false and driver did not try to load module binaries. Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b1ca64d2f7ea..031d5dc7e660 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1317,21 +1317,6 @@ static int skl_get_module_info(struct skl_dev *skl, return -EIO; } - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - if (mconfig->module) - mconfig->module->loadable = module->is_loadable; - ret = 0; - break; - } - } - - if (ret) - return ret; - - uuid_mod = &module->uuid; - ret = -EIO; for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; @@ -1341,10 +1326,18 @@ static int skl_get_module_info(struct skl_dev *skl, break; } } + if (skl->nr_modules && ret) return ret; + ret = -EIO; list_for_each_entry(module, &skl->uuid_list, list) { + if (guid_equal(uuid_mod, &module->uuid)) { + mconfig->id.module_id = module->id; + mconfig->module->loadable = module->is_loadable; + ret = 0; + } + for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; if (guid_equal(&pin_id->mod_uuid, &module->uuid)) @@ -1358,7 +1351,7 @@ static int skl_get_module_info(struct skl_dev *skl, } } - return 0; + return ret; } static int skl_populate_modules(struct skl_dev *skl) -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 178/219] ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER
by Sasha Levin 09 Sep '21

09 Sep '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e4e0633bcadc950b4b4af06c7f1bb7f7e3e86321 ] KeyPhrasebuffer, Mixin and Mixout modules configuration is described by firmware's basic module configuration structure. There are no extended parameters required. Update functions taking part in building INIT_INSTANCE IPC payload to reflect that. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 177/219] soundwire: intel: fix potential race condition during power down
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ea6942dad4b2a7e1735aa0f10f3d0b04b847750f ] The power down sequence sets the link_up flag as false outside of the mutex_lock. This is potentially unsafe. In additional the flow in that sequence can be improved by first testing if the link was powered, setting the link_up flag as false and proceeding with the power down. In case the CPA bits cannot be cleared, we only flag an error since we cannot deal with interrupts any longer. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20210818024954.16873-2-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fd95f94630b1..c03d51ad40bf 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -537,12 +537,14 @@ static int intel_link_power_down(struct sdw_intel *sdw) mutex_lock(sdw->link_res->shim_lock); - intel_shim_master_ip_to_glue(sdw); - if (!(*shim_mask & BIT(link_id))) dev_err(sdw->cdns.dev, "%s: Unbalanced power-up/down calls\n", __func__); + sdw->cdns.link_up = false; + + intel_shim_master_ip_to_glue(sdw); + *shim_mask &= ~BIT(link_id); if (!*shim_mask) { @@ -559,18 +561,19 @@ static int intel_link_power_down(struct sdw_intel *sdw) link_control &= spa_mask; ret = intel_clear_bit(shim, SDW_SHIM_LCTL, link_control, cpa_mask); + if (ret < 0) { + dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); + + /* + * we leave the sdw->cdns.link_up flag as false since we've disabled + * the link at this point and cannot handle interrupts any longer. + */ + } } mutex_unlock(sdw->link_res->shim_lock); - if (ret < 0) { - dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); - - return ret; - } - - sdw->cdns.link_up = false; - return 0; + return ret; } static void intel_shim_sync_arm(struct sdw_intel *sdw) -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 088/219] ASoC: Intel: update sof_pcm512x quirks
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 22414cade8dfec25ab94df52b3a4f7aa8edb6120 ] The default SOF topology enables SSP capture and DMICs, even though both of these hardware capabilities are not always available in hardware (specific versions of HiFiberry and DMIC kit needed). For the SSP capture, this leads to annoying "SP5-Codec: ASoC: no backend capture" and "streamSSP5-Codec: ASoC: no users capture at close - state 0" errors. Update the quirks to match what the topology needs, which also allows for the ability to remove SSP capture and DMIC support. BugLink: https://github.com/thesofproject/linux/issues/3061 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802152151.15832-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_pcm512x.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 8620d4f38493..335c212c1961 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -26,11 +26,16 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define SOF_PCM512X_ENABLE_SSP_CAPTURE BIT(4) +#define SOF_PCM512X_ENABLE_DMIC BIT(5) #define IDISP_CODEC_MASK 0x4 /* Default: SSP5 */ -static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); +static unsigned long sof_pcm512x_quirk = + SOF_PCM512X_SSP_CODEC(5) | + SOF_PCM512X_ENABLE_SSP_CAPTURE | + SOF_PCM512X_ENABLE_DMIC; static bool is_legacy_cpu; @@ -245,8 +250,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].dpcm_playback = 1; /* * capture only supported with specific versions of the Hifiberry DAC+ - * links[id].dpcm_capture = 1; */ + if (sof_pcm512x_quirk & SOF_PCM512X_ENABLE_SSP_CAPTURE) + links[id].dpcm_capture = 1; links[id].no_pcm = 1; links[id].cpus = &cpus[id]; links[id].num_cpus = 1; @@ -381,6 +387,9 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK; + if (!(sof_pcm512x_quirk & SOF_PCM512X_ENABLE_DMIC)) + dmic_be_num = 0; + /* compute number of dai links */ sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 086/219] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 22dbd9d93c1e..4bddb969176f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -290,9 +290,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -300,19 +297,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -354,6 +355,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -361,6 +363,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 007/219] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Sasha Levin 09 Sep '21

09 Sep '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 6c5c659dfe3f02e08054a6c20019e3886618b512 ] On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Link: https://lore.kernel.org/r/20210707214752.3831-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index ec04e3386bc0..8617793ed955 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 006/219] ASoC: ti: davinci-mcasp: Fix DIT mode support
by Sasha Levin 09 Sep '21

09 Sep '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit bbdd3f4dbe81e19b9123bc54e23ed54517615524 ] The DIT mode support has not been tested due to lack of platform where it can be tested. To be able to use the McASP on OMAP4/5 (only supporting DIT mode) we need to have DIT mode working in the McASP driver on a know platform. After hacking around (on BBW, mcasp1.axr1 can be routed out for this) it appeared that DIT mode is broken. This patch fixes it up and 16/24 bit audio works along with passthrough, but I have only tested with DTS example and test files. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210705194249.2385-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 150 ++++++++++++++++++++++++++++++----- 1 file changed, 129 insertions(+), 21 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..41d7cb132198 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -83,6 +83,8 @@ struct davinci_mcasp { struct snd_pcm_substream *substreams[2]; unsigned int dai_fmt; + u32 iec958_status; + /* Audio can not be enabled due to missing parameter(s) */ bool missing_audio_param; @@ -757,6 +759,9 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + return 0; + dev_dbg(mcasp->dev, "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n", __func__, tx_mask, rx_mask, slots, slot_width); @@ -827,6 +832,20 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask); + } else { + /* + * according to the TRM it should be TXROT=0, this one works: + * 16 bit to 23-8 (TXROT=6, rotate 24 bits) + * 24 bit to 23-0 (TXROT=0, rotate 0 bits) + * + * TXROT = 0 only works with 24bit samples + */ + tx_rotate = (sample_width / 4 + 2) & 0x7; + + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), + TXROT(7)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(15), + TXSSZ(0x0F)); } mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask); @@ -842,10 +861,16 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 tx_ser = 0; u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; - u8 max_active_serializers = (channels + slots - 1) / slots; - u8 max_rx_serializers, max_tx_serializers; + u8 max_active_serializers, max_rx_serializers, max_tx_serializers; int active_serializers, numevt; u32 reg; + + /* In DIT mode we only allow maximum of one serializers for now */ + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + max_active_serializers = 1; + else + max_active_serializers = (channels + slots - 1) / slots; + /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -1031,16 +1056,18 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, unsigned int rate) { - u32 cs_value = 0; - u8 *cs_bytes = (u8*) &cs_value; + u8 *cs_bytes = (u8 *)&mcasp->iec958_status; - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 - and LSB first */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); + if (!mcasp->dat_port) + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, 0xFFFF); + /* Set the TX tdm : for all the slots */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); @@ -1049,16 +1076,8 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); - /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); - - /* Enable the DIT */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); - /* Set S/PDIF channel status bits */ - cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; - cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; - + cs_bytes[3] &= ~IEC958_AES3_CON_FS; switch (rate) { case 22050: cs_bytes[3] |= IEC958_AES3_CON_FS_22050; @@ -1088,12 +1107,15 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, cs_bytes[3] |= IEC958_AES3_CON_FS_192000; break; default: - printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + dev_err(mcasp->dev, "unsupported sampling rate: %d\n", rate); return -EINVAL; } - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, mcasp->iec958_status); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, mcasp->iec958_status); + + /* Enable the DIT */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); return 0; } @@ -1237,12 +1259,18 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); + unsigned int bclk_target; if (mcasp->slot_width) sbits = mcasp->slot_width; + if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) + bclk_target = rate * sbits * slots; + else + bclk_target = rate * 128; + davinci_mcasp_calc_clk_div(mcasp, mcasp->sysclk_freq, - rate * sbits * slots, true); + bclk_target, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1598,6 +1626,77 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_tdm_slot = davinci_mcasp_set_tdm_slot, }; +static int davinci_mcasp_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int davinci_mcasp_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(uctl->value.iec958.status, &mcasp->iec958_status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(&mcasp->iec958_status, uctl->value.iec958.status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_con_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memset(ucontrol->value.iec958.status, 0xff, sizeof(mcasp->iec958_status)); + return 0; +} + +static const struct snd_kcontrol_new davinci_mcasp_iec958_ctls[] = { + { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_get, + .put = davinci_mcasp_iec958_put, + }, { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_con_mask_get, + }, +}; + +static void davinci_mcasp_init_iec958_status(struct davinci_mcasp *mcasp) +{ + unsigned char *cs = (u8 *)&mcasp->iec958_status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_PCM_CODER; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM; +} + static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -1605,6 +1704,12 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) { + davinci_mcasp_init_iec958_status(mcasp); + snd_soc_add_dai_controls(dai, davinci_mcasp_iec958_ctls, + ARRAY_SIZE(davinci_mcasp_iec958_ctls)); + } + return 0; } @@ -1651,7 +1756,8 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = DAVINCI_MCASP_PCM_FMTS, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, }, .ops = &davinci_mcasp_dai_ops, }, @@ -1871,6 +1977,8 @@ static int davinci_mcasp_get_config(struct davinci_mcasp *mcasp, } else { mcasp->tdm_slots = pdata->tdm_slots; } + } else { + mcasp->tdm_slots = 32; } mcasp->num_serializer = pdata->num_serializer; -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 224/252] ASoC: soc-pcm: protect BE dailink state changes in trigger
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 0c75fc7193387776c10f7c7b440d93496e3d5e21 ] When more than one FE is connected to a BE, e.g. in a mixing use case, the BE can be triggered multiple times when the FE are opened/started concurrently. This race condition is problematic in the case of SoundWire BE dailinks, and this is not desirable in a general case. The code carefully checks when the BE can be stopped or hw_free'ed, but the trigger code does not use any mutual exclusion. Fix by using the same spinlock already used to check FE states, and set the state before the trigger. In case of errors, the initial state will be restored. This patch does not change how the triggers are handled, it only makes sure the states are handled in critical sections. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Message-Id: <20210817164054.250028-2-pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 103 ++++++++++++++++++++++++++++++++++++-------- 1 file changed, 85 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d1c570ca21ea..b944f56a469a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2001,6 +2001,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_pcm_runtime *be; struct snd_soc_dpcm *dpcm; int ret = 0; + unsigned long flags; + enum snd_soc_dpcm_state state; for_each_dpcm_be(fe, stream, dpcm) { struct snd_pcm_substream *be_substream; @@ -2017,76 +2019,141 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + spin_lock_irqsave(&fe->card->dpcm_lock, flags); if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_RESUME: - if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: + spin_lock_irqsave(&fe->card->dpcm_lock, flags); if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) { + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); continue; + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + state = be->dpcm[stream].state; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + ret = soc_pcm_trigger(be_substream, cmd); - if (ret) + if (ret) { + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + be->dpcm[stream].state = state; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); goto end; + } - be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } } -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 223/252] ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B
by Sasha Levin 09 Sep '21

09 Sep '21

From: Xiaotan Luo <lxt(a)rock-chips.com> [ Upstream commit 1bf56843e664eef2525bdbfae6a561e98910f676 ] - DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC - DSP_B: PCM no delay mode, L data MSB during FRM LRC Signed-off-by: Xiaotan Luo <lxt(a)rock-chips.com> Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950562-14281-3-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 1fd47e56c77f..b65dfbc3545b 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -233,12 +233,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; @@ -257,12 +257,12 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 222/252] ASoC: rockchip: i2s: Fix regmap_ops hang
by Sasha Levin 09 Sep '21

09 Sep '21

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> [ Upstream commit 53ca9b9777b95cdd689181d7c547e38dc79adad0 ] API 'set_fmt' maybe called when PD is off, in the situation, any register access will hang the system. so, enable PD before r/w register. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Link: https://lore.kernel.org/r/1629950520-14190-4-git-send-email-sugar.zhang@roc… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index c7dc3509bceb..1fd47e56c77f 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -186,7 +186,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -199,7 +201,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -213,7 +216,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -236,7 +240,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -259,12 +264,16 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 201/252] ASoC: Intel: Skylake: Fix passing loadable flag for module
by Sasha Levin 09 Sep '21

09 Sep '21

From: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> [ Upstream commit c5ed9c547cba1dc1238c6e8a0c290fd62ee6e127 ] skl_get_module_info() tries to set mconfig->module->loadable before mconfig->module has been assigned thus flag was always set to false and driver did not try to load module binaries. Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b1ca64d2f7ea..031d5dc7e660 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1317,21 +1317,6 @@ static int skl_get_module_info(struct skl_dev *skl, return -EIO; } - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - if (mconfig->module) - mconfig->module->loadable = module->is_loadable; - ret = 0; - break; - } - } - - if (ret) - return ret; - - uuid_mod = &module->uuid; - ret = -EIO; for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; @@ -1341,10 +1326,18 @@ static int skl_get_module_info(struct skl_dev *skl, break; } } + if (skl->nr_modules && ret) return ret; + ret = -EIO; list_for_each_entry(module, &skl->uuid_list, list) { + if (guid_equal(uuid_mod, &module->uuid)) { + mconfig->id.module_id = module->id; + mconfig->module->loadable = module->is_loadable; + ret = 0; + } + for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; if (guid_equal(&pin_id->mod_uuid, &module->uuid)) @@ -1358,7 +1351,7 @@ static int skl_get_module_info(struct skl_dev *skl, } } - return 0; + return ret; } static int skl_populate_modules(struct skl_dev *skl) -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 200/252] ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER
by Sasha Levin 09 Sep '21

09 Sep '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e4e0633bcadc950b4b4af06c7f1bb7f7e3e86321 ] KeyPhrasebuffer, Mixin and Mixout modules configuration is described by firmware's basic module configuration structure. There are no extended parameters required. Update functions taking part in building INIT_INSTANCE IPC payload to reflect that. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 199/252] soundwire: intel: fix potential race condition during power down
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ea6942dad4b2a7e1735aa0f10f3d0b04b847750f ] The power down sequence sets the link_up flag as false outside of the mutex_lock. This is potentially unsafe. In additional the flow in that sequence can be improved by first testing if the link was powered, setting the link_up flag as false and proceeding with the power down. In case the CPA bits cannot be cleared, we only flag an error since we cannot deal with interrupts any longer. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20210818024954.16873-2-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index c11e3d8cd308..f156de765c68 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -537,12 +537,14 @@ static int intel_link_power_down(struct sdw_intel *sdw) mutex_lock(sdw->link_res->shim_lock); - intel_shim_master_ip_to_glue(sdw); - if (!(*shim_mask & BIT(link_id))) dev_err(sdw->cdns.dev, "%s: Unbalanced power-up/down calls\n", __func__); + sdw->cdns.link_up = false; + + intel_shim_master_ip_to_glue(sdw); + *shim_mask &= ~BIT(link_id); if (!*shim_mask) { @@ -559,18 +561,19 @@ static int intel_link_power_down(struct sdw_intel *sdw) link_control &= spa_mask; ret = intel_clear_bit(shim, SDW_SHIM_LCTL, link_control, cpa_mask); + if (ret < 0) { + dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); + + /* + * we leave the sdw->cdns.link_up flag as false since we've disabled + * the link at this point and cannot handle interrupts any longer. + */ + } } mutex_unlock(sdw->link_res->shim_lock); - if (ret < 0) { - dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); - - return ret; - } - - sdw->cdns.link_up = false; - return 0; + return ret; } static void intel_shim_sync_arm(struct sdw_intel *sdw) -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 179/252] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Sasha Levin 09 Sep '21

09 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit cc64c390b215b404524725a94857d6fb58d9a62a ] This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87v940wyf9.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/adg.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..5f1e72edfee0 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -391,9 +391,9 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, struct clk *clk; clk = clk_register_fixed_rate(dev, name, parent, 0, 0); - if (IS_ERR(clk)) { + if (IS_ERR_OR_NULL(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; @@ -582,7 +582,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clkout[CLKOUT] = clk; @@ -596,7 +596,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, req_rate[0]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clkout[i] = clk; -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 102/252] ASoC: Intel: update sof_pcm512x quirks
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 22414cade8dfec25ab94df52b3a4f7aa8edb6120 ] The default SOF topology enables SSP capture and DMICs, even though both of these hardware capabilities are not always available in hardware (specific versions of HiFiberry and DMIC kit needed). For the SSP capture, this leads to annoying "SP5-Codec: ASoC: no backend capture" and "streamSSP5-Codec: ASoC: no users capture at close - state 0" errors. Update the quirks to match what the topology needs, which also allows for the ability to remove SSP capture and DMIC support. BugLink: https://github.com/thesofproject/linux/issues/3061 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802152151.15832-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_pcm512x.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 2ec9c62366e2..6815204e58d5 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -26,11 +26,16 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define SOF_PCM512X_ENABLE_SSP_CAPTURE BIT(4) +#define SOF_PCM512X_ENABLE_DMIC BIT(5) #define IDISP_CODEC_MASK 0x4 /* Default: SSP5 */ -static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); +static unsigned long sof_pcm512x_quirk = + SOF_PCM512X_SSP_CODEC(5) | + SOF_PCM512X_ENABLE_SSP_CAPTURE | + SOF_PCM512X_ENABLE_DMIC; static bool is_legacy_cpu; @@ -244,8 +249,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].dpcm_playback = 1; /* * capture only supported with specific versions of the Hifiberry DAC+ - * links[id].dpcm_capture = 1; */ + if (sof_pcm512x_quirk & SOF_PCM512X_ENABLE_SSP_CAPTURE) + links[id].dpcm_capture = 1; links[id].no_pcm = 1; links[id].cpus = &cpus[id]; links[id].num_cpus = 1; @@ -380,6 +386,9 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK; + if (!(sof_pcm512x_quirk & SOF_PCM512X_ENABLE_DMIC)) + dmic_be_num = 0; + /* compute number of dai links */ sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 100/252] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 91a6d712eb58..c403fb672594 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -290,9 +290,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -300,19 +297,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -354,6 +355,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -361,6 +363,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 013/252] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Sasha Levin 09 Sep '21

09 Sep '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 6c5c659dfe3f02e08054a6c20019e3886618b512 ] On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Link: https://lore.kernel.org/r/20210707214752.3831-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index ec04e3386bc0..8617793ed955 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 012/252] ASoC: ti: davinci-mcasp: Fix DIT mode support
by Sasha Levin 09 Sep '21

09 Sep '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit bbdd3f4dbe81e19b9123bc54e23ed54517615524 ] The DIT mode support has not been tested due to lack of platform where it can be tested. To be able to use the McASP on OMAP4/5 (only supporting DIT mode) we need to have DIT mode working in the McASP driver on a know platform. After hacking around (on BBW, mcasp1.axr1 can be routed out for this) it appeared that DIT mode is broken. This patch fixes it up and 16/24 bit audio works along with passthrough, but I have only tested with DTS example and test files. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210705194249.2385-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 150 ++++++++++++++++++++++++++++++----- 1 file changed, 129 insertions(+), 21 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 017a5a5e56cd..64ec6d485834 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -83,6 +83,8 @@ struct davinci_mcasp { struct snd_pcm_substream *substreams[2]; unsigned int dai_fmt; + u32 iec958_status; + /* Audio can not be enabled due to missing parameter(s) */ bool missing_audio_param; @@ -757,6 +759,9 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + return 0; + dev_dbg(mcasp->dev, "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n", __func__, tx_mask, rx_mask, slots, slot_width); @@ -827,6 +832,20 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask); + } else { + /* + * according to the TRM it should be TXROT=0, this one works: + * 16 bit to 23-8 (TXROT=6, rotate 24 bits) + * 24 bit to 23-0 (TXROT=0, rotate 0 bits) + * + * TXROT = 0 only works with 24bit samples + */ + tx_rotate = (sample_width / 4 + 2) & 0x7; + + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), + TXROT(7)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(15), + TXSSZ(0x0F)); } mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask); @@ -842,10 +861,16 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 tx_ser = 0; u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; - u8 max_active_serializers = (channels + slots - 1) / slots; - u8 max_rx_serializers, max_tx_serializers; + u8 max_active_serializers, max_rx_serializers, max_tx_serializers; int active_serializers, numevt; u32 reg; + + /* In DIT mode we only allow maximum of one serializers for now */ + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + max_active_serializers = 1; + else + max_active_serializers = (channels + slots - 1) / slots; + /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -1031,16 +1056,18 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, unsigned int rate) { - u32 cs_value = 0; - u8 *cs_bytes = (u8*) &cs_value; + u8 *cs_bytes = (u8 *)&mcasp->iec958_status; - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 - and LSB first */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); + if (!mcasp->dat_port) + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, 0xFFFF); + /* Set the TX tdm : for all the slots */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); @@ -1049,16 +1076,8 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); - /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); - - /* Enable the DIT */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); - /* Set S/PDIF channel status bits */ - cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; - cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; - + cs_bytes[3] &= ~IEC958_AES3_CON_FS; switch (rate) { case 22050: cs_bytes[3] |= IEC958_AES3_CON_FS_22050; @@ -1088,12 +1107,15 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, cs_bytes[3] |= IEC958_AES3_CON_FS_192000; break; default: - printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + dev_err(mcasp->dev, "unsupported sampling rate: %d\n", rate); return -EINVAL; } - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, mcasp->iec958_status); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, mcasp->iec958_status); + + /* Enable the DIT */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); return 0; } @@ -1237,12 +1259,18 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); + unsigned int bclk_target; if (mcasp->slot_width) sbits = mcasp->slot_width; + if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) + bclk_target = rate * sbits * slots; + else + bclk_target = rate * 128; + davinci_mcasp_calc_clk_div(mcasp, mcasp->sysclk_freq, - rate * sbits * slots, true); + bclk_target, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1598,6 +1626,77 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_tdm_slot = davinci_mcasp_set_tdm_slot, }; +static int davinci_mcasp_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int davinci_mcasp_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(uctl->value.iec958.status, &mcasp->iec958_status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(&mcasp->iec958_status, uctl->value.iec958.status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_con_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memset(ucontrol->value.iec958.status, 0xff, sizeof(mcasp->iec958_status)); + return 0; +} + +static const struct snd_kcontrol_new davinci_mcasp_iec958_ctls[] = { + { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_get, + .put = davinci_mcasp_iec958_put, + }, { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_con_mask_get, + }, +}; + +static void davinci_mcasp_init_iec958_status(struct davinci_mcasp *mcasp) +{ + unsigned char *cs = (u8 *)&mcasp->iec958_status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_PCM_CODER; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM; +} + static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -1605,6 +1704,12 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) { + davinci_mcasp_init_iec958_status(mcasp); + snd_soc_add_dai_controls(dai, davinci_mcasp_iec958_ctls, + ARRAY_SIZE(davinci_mcasp_iec958_ctls)); + } + return 0; } @@ -1651,7 +1756,8 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = DAVINCI_MCASP_PCM_FMTS, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, }, .ops = &davinci_mcasp_dai_ops, }, @@ -1871,6 +1977,8 @@ static int davinci_mcasp_get_config(struct davinci_mcasp *mcasp, } else { mcasp->tdm_slots = pdata->tdm_slots; } + } else { + mcasp->tdm_slots = 32; } mcasp->num_serializer = pdata->num_serializer; -- 2.30.2

1 0

[PATCH] ASoC: SOF: fix a typo in put operations for kcontrol
by Pierre-Louis Bossart 08 Sep '21

08 Sep '21

From: Rander Wang <rander.wang(a)intel.com> SOF_CTRL_TYPE_VALUE_CHAN_SET should be used for put operations for consistency. The current use of _GET is obviously incorrect but _GET and _SET result in the same action so there is no functional change introduced by this patch. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index a5dd728c580a..504500dd4d43 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -108,7 +108,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_VOLUME, true); return change; @@ -179,7 +179,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_SWITCH, true); @@ -226,7 +226,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_ENUM, true); -- 2.25.1

1 1

[GIT PULL] ASoC fixes for v5.15-rc1
by Mark Brown 08 Sep '21

08 Sep '21

The following changes since commit 38b7673000949ca784fcb8a9feb70d2a802befa6: Merge remote-tracking branch 'asoc/for-5.15' into asoc-linus (2021-08-30 12:30:33 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc1 for you to fetch changes up to fcb958ee8e832e9cdf43408535207e15f14af755: ASoC: rockchip: i2s: Fix concurrency between tx/rx (2021-09-03 16:19:01 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A collection of fixes that came in during the merge window, nothing too remarkable but a reasonably large number of fixes. ---------------------------------------------------------------- Colin Ian King (1): ASoC: samsung: s3c24xx_simtec: fix spelling mistake "devicec" -> "device" Geert Uytterhoeven (1): ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK Kuninori Morimoto (1): ASoC: audio-graph: respawn Platform Support Mark Brown (1): Merge tag 'asoc-v5.15' into asoc-5.15 Nathan Chancellor (1): ASoC: Intel: boards: Fix CONFIG_SND_SOC_SDW_MOCKUP select Shengjiu Wang (1): ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX8ULP Shuming Fan (1): ASoC: rt5682: fix headset background noise when S3 state Sugar Zhang (1): ASoC: rockchip: i2s: Fix concurrency between tx/rx Trevor Wu (3): ASoC: mediatek: mt8195: add MTK_PMIC_WRAP dependency ASoC: dt-bindings: mt8195: remove dependent headers in the example ASoC: mt8195: correct the dts parsing logic about DPTX and HDMITX .../devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + .../devicetree/bindings/sound/mt8195-afe-pcm.yaml | 40 ++++++++++------------ sound/soc/codecs/rt5682.c | 3 -- sound/soc/generic/audio-graph-card.c | 6 ++++ sound/soc/intel/boards/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 3 +- .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 22 ++++++------ sound/soc/rockchip/rockchip_i2s.c | 7 ++++ sound/soc/samsung/s3c24xx_simtec.c | 2 +- 9 files changed, 47 insertions(+), 39 deletions(-)

2 1

[PATCH v5 00/21] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 08 Sep '21

08 Sep '21

Many thanks for reviewing v4. This version addresses all the comments raised as part of v4 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ Thanks, srini Changes since v4: only apr dt-binding changes. - deprecated qcom,apr-domain dt-bindings to make it more reusable - added more protection-domain list that have been missed during yaml conversion. - remove "qcom,apr" compatible that was added by mistake during yaml conversion. Srinivas Kandagatla (21): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 169 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 50 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 74 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 202 +++ drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 289 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 203 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1170 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 692 ++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 504 +++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 266 ++++ sound/soc/qcom/qdsp6/q6apm.c | 916 +++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 154 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 256 ++++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1118 ++++++++++++++++ 31 files changed, 7406 insertions(+), 1414 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

5 53

[PATCH] ASoC: qcom: lpass-platform: Reset irq clear reg post handling interrupts
by Srinivasa Rao Mandadapu 07 Sep '21

07 Sep '21

Update interrupt clear register with reset value after addressing all interrupts. This is to fix playback or capture hanging issue in simultaneous playback and capture usecase. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f9df76d37858..1a0a4b0b1a03 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -749,6 +749,12 @@ static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data) } } + rv = regmap_write(drvdata->lpaif_map, LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST), 0x0); + if (rv) { + pr_err("error writing to irqstat reg: %d\n", rv); + return IRQ_NONE; + } + return IRQ_HANDLED; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH v6 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 07 Sep '21

07 Sep '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v6 changes: Fix header comment style Remove some unsafe mixer controls Remove some fields in regmap readable_reg Remove unneccesssary regmap setup in component_probe Remove extra whitespace Reorder in probe: handle_pdata before regulator_enable Set initial analog gain to 0dB David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 162 ++ include/sound/cs35l41.h | 35 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 114 ++ sound/soc/codecs/cs35l41-spi.c | 143 ++ sound/soc/codecs/cs35l41-tables.c | 597 +++++++ sound/soc/codecs/cs35l41.c | 1561 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 775 ++++++++ 9 files changed, 3403 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

2 4

[PATCH] ASoC: wm8960: Fix clock configuration on slave mode
by Shengjiu Wang 07 Sep '21

07 Sep '21

There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392..9c6af76a60fd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,7 +742,7 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { dev_dbg(component->dev, "Codec is slave mode, no need to configure clock\n"); return 0; -- 2.17.1

2 1

[PATCH 1/2] ALSA: gus: Fix repeated probes of snd_gus_create()
by Takashi Iwai 07 Sep '21

07 Sep '21

GUS card object may be repeatedly probed for the legacy ISA devices, and the behavior doesn't fit with the devres resource management. Revert partially back to the classical way for the snd_gus_card object, so that the repeated calls of snd_gus_create() are allowed. Fixes: 5b88da3c800f ("ALSA: gus: Allocate resources with device-managed APIs") Reported-by: kernel test robot <oliver.sang(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/gus/gus_main.c | 44 ++++++++++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 13 deletions(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index ae1e2542ee4a..3b46490271fe 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -87,10 +87,24 @@ static void snd_gus_init_control(struct snd_gus_card *gus) static int snd_gus_free(struct snd_gus_card *gus) { - if (gus->gf1.res_port2) { - snd_gf1_stop(gus); - snd_gus_init_dma_irq(gus, 0); + if (gus->gf1.res_port2 == NULL) + goto __hw_end; + snd_gf1_stop(gus); + snd_gus_init_dma_irq(gus, 0); + __hw_end: + release_and_free_resource(gus->gf1.res_port1); + release_and_free_resource(gus->gf1.res_port2); + if (gus->gf1.irq >= 0) + free_irq(gus->gf1.irq, (void *) gus); + if (gus->gf1.dma1 >= 0) { + disable_dma(gus->gf1.dma1); + free_dma(gus->gf1.dma1); } + if (!gus->equal_dma && gus->gf1.dma2 >= 0) { + disable_dma(gus->gf1.dma2); + free_dma(gus->gf1.dma2); + } + kfree(gus); return 0; } @@ -116,7 +130,7 @@ int snd_gus_create(struct snd_card *card, }; *rgus = NULL; - gus = devm_kzalloc(card->dev, sizeof(*gus), GFP_KERNEL); + gus = kzalloc(sizeof(*gus), GFP_KERNEL); if (gus == NULL) return -ENOMEM; spin_lock_init(&gus->reg_lock); @@ -142,33 +156,35 @@ int snd_gus_create(struct snd_card *card, gus->gf1.reg_timerctrl = GUSP(gus, TIMERCNTRL); gus->gf1.reg_timerdata = GUSP(gus, TIMERDATA); /* allocate resources */ - gus->gf1.res_port1 = devm_request_region(card->dev, port, 16, - "GUS GF1 (Adlib/SB)"); + gus->gf1.res_port1 = request_region(port, 16, "GUS GF1 (Adlib/SB)"); if (!gus->gf1.res_port1) { snd_printk(KERN_ERR "gus: can't grab SB port 0x%lx\n", port); + snd_gus_free(gus); return -EBUSY; } - gus->gf1.res_port2 = devm_request_region(card->dev, port + 0x100, 12, - "GUS GF1 (Synth)"); + gus->gf1.res_port2 = request_region(port + 0x100, 12, "GUS GF1 (Synth)"); if (!gus->gf1.res_port2) { snd_printk(KERN_ERR "gus: can't grab synth port 0x%lx\n", port + 0x100); + snd_gus_free(gus); return -EBUSY; } - if (irq >= 0 && devm_request_irq(card->dev, irq, snd_gus_interrupt, 0, - "GUS GF1", (void *) gus)) { + if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) { snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq); + snd_gus_free(gus); return -EBUSY; } gus->gf1.irq = irq; card->sync_irq = irq; - if (snd_devm_request_dma(card->dev, dma1, "GUS - 1")) { + if (request_dma(dma1, "GUS - 1")) { snd_printk(KERN_ERR "gus: can't grab DMA1 %d\n", dma1); + snd_gus_free(gus); return -EBUSY; } gus->gf1.dma1 = dma1; if (dma2 >= 0 && dma1 != dma2) { - if (snd_devm_request_dma(card->dev, dma2, "GUS - 2")) { + if (request_dma(dma2, "GUS - 2")) { snd_printk(KERN_ERR "gus: can't grab DMA2 %d\n", dma2); + snd_gus_free(gus); return -EBUSY; } gus->gf1.dma2 = dma2; @@ -193,8 +209,10 @@ int snd_gus_create(struct snd_card *card, gus->gf1.volume_ramp = 25; gus->gf1.smooth_pan = 1; err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, gus, &ops); - if (err < 0) + if (err < 0) { + snd_gus_free(gus); return err; + } *rgus = gus; return 0; } -- 2.26.2

1 1

[PATCH v1] ALSA: vx222: fix null-ptr-deref
by Tong Zhang 07 Sep '21

07 Sep '21

a recent refactor created a null pointer vx in snd_vx222_probe(). The vx pointer should have been populated in snd_vx222_create() as suggested in earlier version, otherwise vx->core.ibl.size will throw an error. [ 1.298398] BUG: kernel NULL pointer dereference, address: 00000000000001d8 [ 1.316799] RIP: 0010:snd_vx222_probe+0x155/0x290 [snd_vx222] Fixes: 3bde3359aa16 ("ALSA: vx222: Allocate resources with device-managed APIs") Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> --- sound/pci/vx222/vx222.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index f48cc20b9e8a..468a6a20dc1e 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -137,6 +137,7 @@ static int snd_vx222_create(struct snd_card *card, struct pci_dev *pci, } chip->irq = pci->irq; card->sync_irq = chip->irq; + *rchip = vx; return 0; } -- 2.25.1

2 1

ucm2: Add UCM support for rockchip_es8316 on Pinebook Pro
by GitHub pull_request - opened 07 Sep '21

07 Sep '21

alsa-project/alsa-ucm-conf pull request #112 was opened from starnight: Add UCM config files for Pinebook Pro's audio. The config files are transformed and tuned from Manjaro ARM's packages: * pinebookpro-post-install's asound.state [1] * pinebookpro-audio's audio_jack_plugged_in.sh [2] By the way, with these config files, the Speaker and Headphones will be switched according to the (un)plugged state automatically. [1] https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-post-… [2] https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-audio… Fixes: https://github.com/alsa-project/alsa-ucm-conf/issues/110 Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/112 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/112.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: dpcm: fix race condition to dpcm links in dpcm_be_dai_trigger
by Gyeongtaek Lee 07 Sep '21

07 Sep '21

If routing change and underrun stop is run at the same time, data abort can be occurred by the following sequence. CPU0: Processing underrun CPU1: Processing routing change dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock is needed during accessing the lists for dpcm links. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 53 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 48f71bb81a2f..df2cd4c0dabe 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1993,17 +1993,63 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +struct dpcm_be_list { + unsigned int num; + struct snd_soc_pcm_runtime *be[]; +}; + +static int dpcm_create_be_list(struct snd_soc_pcm_runtime *fe, int stream, + struct dpcm_be_list **be_list) +{ + struct snd_soc_dpcm *dpcm; + struct dpcm_be_list *be; + int size = 0; + int ret = 0; + unsigned long flags; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + + for_each_dpcm_be(fe, stream, dpcm) + size++; + + be = kzalloc(struct_size(be, be, size), GFP_ATOMIC); + if (!be) { + ret = -ENOMEM; + } else { + unsigned int i = 0; + + for_each_dpcm_be(fe, stream, dpcm) + be->be[i++] = dpcm->be; + + *be_list = be; + } + + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} + +static void dpcm_free_be_list(struct dpcm_be_list *be_list) +{ + kfree(be_list); +} + int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_pcm_runtime *be; - struct snd_soc_dpcm *dpcm; + struct dpcm_be_list *be_list; int ret = 0; + int i; - for_each_dpcm_be(fe, stream, dpcm) { + ret = dpcm_create_be_list(fe, stream, &be_list); + if (ret < 0) + return ret; + + for(i = 0; i < be_list->num; i++) { struct snd_pcm_substream *be_substream; - be = dpcm->be; + be = be_list->be[i]; be_substream = snd_soc_dpcm_get_substream(be, stream); /* is this op for this BE ? */ @@ -2092,6 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (ret < 0) dev_err(fe->dev, "ASoC: %s() failed at %s (%d)\n", __func__, be->dai_link->name, ret); + dpcm_free_be_list(be_list); return ret; } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); base-commit: 4ac6d90867a4de2e12117e755dbd76e08d88697f -- 2.21.0

1 0

[PATCH] dt-bindings: Use 'enum' instead of 'oneOf' plus 'const' entries
by Rob Herring 06 Sep '21

06 Sep '21

'enum' is equivalent to 'oneOf' with a list of 'const' entries, but 'enum' is more concise and yields better error messages. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Maxime Ripard <mripard(a)kernel.org> Cc: Vignesh R <vigneshr(a)ti.com> Cc: Marc Zyngier <maz(a)kernel.org> Cc: Sakari Ailus <sakari.ailus(a)linux.intel.com> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Lee Jones <lee.jones(a)linaro.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Kishon Vijay Abraham I <kishon(a)ti.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Lezcano <daniel.lezcano(a)linaro.org> Cc: dmaengine(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-pci(a)vger.kernel.org Cc: linux-phy(a)lists.infradead.org Cc: linux-serial(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-spi(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/display/msm/dsi-phy-10nm.yaml | 6 +++--- .../bindings/display/msm/dsi-phy-14nm.yaml | 6 +++--- .../bindings/display/msm/dsi-phy-28nm.yaml | 8 ++++---- .../bindings/dma/allwinner,sun6i-a31-dma.yaml | 12 ++++++------ .../devicetree/bindings/firmware/arm,scpi.yaml | 6 +++--- .../devicetree/bindings/i2c/ti,omap4-i2c.yaml | 10 +++++----- .../interrupt-controller/loongson,liointc.yaml | 8 ++++---- .../devicetree/bindings/media/i2c/mipi-ccs.yaml | 8 ++++---- .../devicetree/bindings/mfd/ti,lp87565-q1.yaml | 6 +++--- .../devicetree/bindings/net/realtek-bluetooth.yaml | 8 ++++---- .../bindings/net/ti,k3-am654-cpsw-nuss.yaml | 8 ++++---- .../devicetree/bindings/net/ti,k3-am654-cpts.yaml | 6 +++--- Documentation/devicetree/bindings/pci/loongson.yaml | 8 ++++---- .../devicetree/bindings/phy/intel,lgm-emmc-phy.yaml | 6 +++--- .../devicetree/bindings/serial/8250_omap.yaml | 9 +++++---- .../devicetree/bindings/sound/qcom,sm8250.yaml | 6 +++--- .../devicetree/bindings/sound/tlv320adcx140.yaml | 8 ++++---- .../devicetree/bindings/spi/realtek,rtl-spi.yaml | 12 ++++++------ .../devicetree/bindings/timer/arm,sp804.yaml | 6 +++--- 19 files changed, 74 insertions(+), 73 deletions(-) diff --git a/Documentation/devicetree/bindings/display/msm/dsi-phy-10nm.yaml b/Documentation/devicetree/bindings/display/msm/dsi-phy-10nm.yaml index 4a26bef19360..4399715953e1 100644 --- a/Documentation/devicetree/bindings/display/msm/dsi-phy-10nm.yaml +++ b/Documentation/devicetree/bindings/display/msm/dsi-phy-10nm.yaml @@ -14,9 +14,9 @@ allOf: properties: compatible: - oneOf: - - const: qcom,dsi-phy-10nm - - const: qcom,dsi-phy-10nm-8998 + enum: + - qcom,dsi-phy-10nm + - qcom,dsi-phy-10nm-8998 reg: items: diff --git a/Documentation/devicetree/bindings/display/msm/dsi-phy-14nm.yaml b/Documentation/devicetree/bindings/display/msm/dsi-phy-14nm.yaml index 72a00cce0147..064df50e21a5 100644 --- a/Documentation/devicetree/bindings/display/msm/dsi-phy-14nm.yaml +++ b/Documentation/devicetree/bindings/display/msm/dsi-phy-14nm.yaml @@ -14,9 +14,9 @@ allOf: properties: compatible: - oneOf: - - const: qcom,dsi-phy-14nm - - const: qcom,dsi-phy-14nm-660 + enum: + - qcom,dsi-phy-14nm + - qcom,dsi-phy-14nm-660 reg: items: diff --git a/Documentation/devicetree/bindings/display/msm/dsi-phy-28nm.yaml b/Documentation/devicetree/bindings/display/msm/dsi-phy-28nm.yaml index b106007116b4..69eecaa64b18 100644 --- a/Documentation/devicetree/bindings/display/msm/dsi-phy-28nm.yaml +++ b/Documentation/devicetree/bindings/display/msm/dsi-phy-28nm.yaml @@ -14,10 +14,10 @@ allOf: properties: compatible: - oneOf: - - const: qcom,dsi-phy-28nm-hpm - - const: qcom,dsi-phy-28nm-lp - - const: qcom,dsi-phy-28nm-8960 + enum: + - qcom,dsi-phy-28nm-hpm + - qcom,dsi-phy-28nm-lp + - qcom,dsi-phy-28nm-8960 reg: items: diff --git a/Documentation/devicetree/bindings/dma/allwinner,sun6i-a31-dma.yaml b/Documentation/devicetree/bindings/dma/allwinner,sun6i-a31-dma.yaml index c1676b96daac..a6df6f8b54db 100644 --- a/Documentation/devicetree/bindings/dma/allwinner,sun6i-a31-dma.yaml +++ b/Documentation/devicetree/bindings/dma/allwinner,sun6i-a31-dma.yaml @@ -19,12 +19,12 @@ properties: description: The cell is the request line number. compatible: - oneOf: - - const: allwinner,sun6i-a31-dma - - const: allwinner,sun8i-a23-dma - - const: allwinner,sun8i-a83t-dma - - const: allwinner,sun8i-h3-dma - - const: allwinner,sun8i-v3s-dma + enum: + - allwinner,sun6i-a31-dma + - allwinner,sun8i-a23-dma + - allwinner,sun8i-a83t-dma + - allwinner,sun8i-h3-dma + - allwinner,sun8i-v3s-dma reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/firmware/arm,scpi.yaml b/Documentation/devicetree/bindings/firmware/arm,scpi.yaml index d7113b06454b..23b346bd1252 100644 --- a/Documentation/devicetree/bindings/firmware/arm,scpi.yaml +++ b/Documentation/devicetree/bindings/firmware/arm,scpi.yaml @@ -131,9 +131,9 @@ properties: properties: compatible: - oneOf: - - const: arm,scpi-dvfs-clocks - - const: arm,scpi-variable-clocks + enum: + - arm,scpi-dvfs-clocks + - arm,scpi-variable-clocks '#clock-cells': const: 1 diff --git a/Documentation/devicetree/bindings/i2c/ti,omap4-i2c.yaml b/Documentation/devicetree/bindings/i2c/ti,omap4-i2c.yaml index ff165ad1bee8..db0843be91c5 100644 --- a/Documentation/devicetree/bindings/i2c/ti,omap4-i2c.yaml +++ b/Documentation/devicetree/bindings/i2c/ti,omap4-i2c.yaml @@ -72,11 +72,11 @@ additionalProperties: false if: properties: compatible: - oneOf: - - const: ti,omap2420-i2c - - const: ti,omap2430-i2c - - const: ti,omap3-i2c - - const: ti,omap4-i2c + enum: + - ti,omap2420-i2c + - ti,omap2430-i2c + - ti,omap3-i2c + - ti,omap4-i2c then: properties: diff --git a/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml b/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml index edf26452dc72..750cc44628e9 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml @@ -19,10 +19,10 @@ allOf: properties: compatible: - oneOf: - - const: loongson,liointc-1.0 - - const: loongson,liointc-1.0a - - const: loongson,liointc-2.0 + enum: + - loongson,liointc-1.0 + - loongson,liointc-1.0a + - loongson,liointc-2.0 reg: minItems: 1 diff --git a/Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml b/Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml index 701f4e0d138f..39395ea8c318 100644 --- a/Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml +++ b/Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml @@ -83,10 +83,10 @@ properties: link-frequencies: true data-lanes: true bus-type: - oneOf: - - const: 1 # CSI-2 C-PHY - - const: 3 # CCP2 - - const: 4 # CSI-2 D-PHY + enum: + - 1 # CSI-2 C-PHY + - 3 # CCP2 + - 4 # CSI-2 D-PHY required: - link-frequencies diff --git a/Documentation/devicetree/bindings/mfd/ti,lp87565-q1.yaml b/Documentation/devicetree/bindings/mfd/ti,lp87565-q1.yaml index 48d4d53c25f9..012d25111054 100644 --- a/Documentation/devicetree/bindings/mfd/ti,lp87565-q1.yaml +++ b/Documentation/devicetree/bindings/mfd/ti,lp87565-q1.yaml @@ -11,9 +11,9 @@ maintainers: properties: compatible: - oneOf: - - const: ti,lp87565 - - const: ti,lp87565-q1 + enum: + - ti,lp87565 + - ti,lp87565-q1 reg: description: I2C slave address diff --git a/Documentation/devicetree/bindings/net/realtek-bluetooth.yaml b/Documentation/devicetree/bindings/net/realtek-bluetooth.yaml index 4f485df69ac3..0634e69dd9a6 100644 --- a/Documentation/devicetree/bindings/net/realtek-bluetooth.yaml +++ b/Documentation/devicetree/bindings/net/realtek-bluetooth.yaml @@ -17,10 +17,10 @@ description: properties: compatible: - oneOf: - - const: "realtek,rtl8723bs-bt" - - const: "realtek,rtl8723cs-bt" - - const: "realtek,rtl8822cs-bt" + enum: + - realtek,rtl8723bs-bt + - realtek,rtl8723cs-bt + - realtek,rtl8822cs-bt device-wake-gpios: maxItems: 1 diff --git a/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml b/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml index 783b9e32cf66..4b97a0f1175b 100644 --- a/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml +++ b/Documentation/devicetree/bindings/net/ti,k3-am654-cpsw-nuss.yaml @@ -53,10 +53,10 @@ properties: "#size-cells": true compatible: - oneOf: - - const: ti,am654-cpsw-nuss - - const: ti,j721e-cpsw-nuss - - const: ti,am642-cpsw-nuss + enum: + - ti,am654-cpsw-nuss + - ti,j721e-cpsw-nuss + - ti,am642-cpsw-nuss reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml b/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml index 4317eba503ca..1a81bf70c88c 100644 --- a/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml +++ b/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml @@ -45,9 +45,9 @@ properties: pattern: "^cpts@[0-9a-f]+$" compatible: - oneOf: - - const: ti,am65-cpts - - const: ti,j721e-cpts + enum: + - ti,am65-cpts + - ti,j721e-cpts reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/pci/loongson.yaml b/Documentation/devicetree/bindings/pci/loongson.yaml index 82bc6c486ca3..a8324a9bd002 100644 --- a/Documentation/devicetree/bindings/pci/loongson.yaml +++ b/Documentation/devicetree/bindings/pci/loongson.yaml @@ -17,10 +17,10 @@ allOf: properties: compatible: - oneOf: - - const: loongson,ls2k-pci - - const: loongson,ls7a-pci - - const: loongson,rs780e-pci + enum: + - loongson,ls2k-pci + - loongson,ls7a-pci + - loongson,rs780e-pci reg: minItems: 1 diff --git a/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml b/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml index edd9d70a672a..954e67571dfd 100644 --- a/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml +++ b/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml @@ -23,9 +23,9 @@ description: |+ properties: compatible: - oneOf: - - const: intel,lgm-emmc-phy - - const: intel,keembay-emmc-phy + enum: + - intel,lgm-emmc-phy + - intel,keembay-emmc-phy "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/serial/8250_omap.yaml b/Documentation/devicetree/bindings/serial/8250_omap.yaml index 1c826fcf5828..c987fb648c3c 100644 --- a/Documentation/devicetree/bindings/serial/8250_omap.yaml +++ b/Documentation/devicetree/bindings/serial/8250_omap.yaml @@ -90,10 +90,11 @@ additionalProperties: false if: properties: compatible: - oneOf: - - const: ti,omap2-uart - - const: ti,omap3-uart - - const: ti,omap4-uart + contains: + enum: + - ti,omap2-uart + - ti,omap3-uart + - ti,omap4-uart then: properties: diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 72ad9ab91832..7d57eb91657a 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -15,9 +15,9 @@ description: properties: compatible: - oneOf: - - const: qcom,sm8250-sndcard - - const: qcom,qrb5165-rb5-sndcard + enum: + - qcom,sm8250-sndcard + - qcom,qrb5165-rb5-sndcard audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 54d64785aad2..d77c8283526d 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -24,10 +24,10 @@ description: | properties: compatible: - oneOf: - - const: ti,tlv320adc3140 - - const: ti,tlv320adc5140 - - const: ti,tlv320adc6140 + enum: + - ti,tlv320adc3140 + - ti,tlv320adc5140 + - ti,tlv320adc6140 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/spi/realtek,rtl-spi.yaml b/Documentation/devicetree/bindings/spi/realtek,rtl-spi.yaml index 30a62a211984..2f938c293f70 100644 --- a/Documentation/devicetree/bindings/spi/realtek,rtl-spi.yaml +++ b/Documentation/devicetree/bindings/spi/realtek,rtl-spi.yaml @@ -15,12 +15,12 @@ allOf: properties: compatible: - oneOf: - - const: realtek,rtl8380-spi - - const: realtek,rtl8382-spi - - const: realtek,rtl8391-spi - - const: realtek,rtl8392-spi - - const: realtek,rtl8393-spi + enum: + - realtek,rtl8380-spi + - realtek,rtl8382-spi + - realtek,rtl8391-spi + - realtek,rtl8392-spi + - realtek,rtl8393-spi reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/timer/arm,sp804.yaml b/Documentation/devicetree/bindings/timer/arm,sp804.yaml index 960e2bd66a97..41be7cdab2ec 100644 --- a/Documentation/devicetree/bindings/timer/arm,sp804.yaml +++ b/Documentation/devicetree/bindings/timer/arm,sp804.yaml @@ -23,9 +23,9 @@ select: properties: compatible: contains: - oneOf: - - const: arm,sp804 - - const: hisilicon,sp804 + enum: + - arm,sp804 + - hisilicon,sp804 required: - compatible -- 2.30.2

9 8

hdajackretask doesn't check for script's exit code
by GitHub issues - edited 06 Sep '21

06 Sep '21

alsa-project/alsa-tools issue #8 was edited from kroq-gar78: `hdajackretask` creates a script in `/tmp` before executing it. When `/tmp` is mounted with `noexec`, running that script will fail, but the GUI gives no indication that it failed. I'm on Ubuntu 18.04.5, with `alsa-tools-gui` package version `1.1.3-1`. The GUI should check for the exit code of the script, and notify the user appropriately if the script failed to execute. Also, if this isn't the appropriate place to report a bug, please let me know! Issue URL : https://github.com/alsa-project/alsa-tools/issues/8 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdajackretask doesn't check for script's exit code
by GitHub issues - edited 06 Sep '21

06 Sep '21

alsa-project/alsa-tools issue #8 was edited from kroq-gar78: `hdajackretask` creates a script in `/tmp` before executing it. When `/tmp` is mounted with `noexec`, running that script will fail, but the GUI gives no indication that it failed. I'm on Ubuntu 18.04.5 with `alsa-tools-gui` package version `1.1.3-1`. The GUI should check for the exit code of the script, and notify the user appropriately if the script failed to execute. Also, if this isn't the appropriate place to report a bug, please let me know! Issue URL : https://github.com/alsa-project/alsa-tools/issues/8 Repository URL: https://github.com/alsa-project/alsa-tools

1 0