July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: samsung: s3c2412: cleanups / fixes for preparation of clocks.
by Arvind Yadav 27 Jul '17

27 Jul '17

-Use devm_clk_get() to make cleanup paths more simple. -clk_prepare_enable() can fail here and we must check its return value. -Add s3c_i2sv2_remove cleanup function of s3c_i2sv2_probe. -No need to iounmap. Here, mapping done by devm_ioremap_resource. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/samsung/s3c-i2s-v2.c | 19 ++++++++++++++++--- sound/soc/samsung/s3c-i2s-v2.h | 3 +++ sound/soc/samsung/s3c2412-i2s.c | 10 ++++++++-- 3 files changed, 27 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 8f42dea..ab4899f 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -625,20 +625,24 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, { struct device *dev = dai->dev; unsigned int iismod; + int ret; i2s->dev = dev; /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->iis_pclk = clk_get(dev, "iis"); + i2s->iis_pclk = devm_clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); - iounmap(i2s->regs); return -ENOENT; } - clk_enable(i2s->iis_pclk); + ret = clk_prepare_enable(i2s->iis_pclk); + if (ret) { + dev_err(dev, "failed to prepare iis_clock\n"); + return ret; + } /* Mark ourselves as in TXRX mode so we can run through our cleanup * process without warnings. */ @@ -652,6 +656,15 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); +int s3c_i2sv2_remove(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + clk_disable_unprepare(i2s->iis_pclk); + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_remove); + #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) { diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index 182d805..d6844b9 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -91,6 +91,9 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, struct s3c_i2sv2_info *i2s, unsigned long base); + +extern int s3c_i2sv2_remove(struct snd_soc_dai *dai); + /** * s3c_i2sv2_register_component - register component and dai with soc core * @dev: DAI device diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 0a47182..adfbd52d 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -65,13 +65,18 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); + s3c_i2sv2_remove(dai); return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_prepare_enable(s3c2412_i2s.iis_cclk); + clk_set_parent(s3c2412_i2s.iis_cclk, devm_clk_get(dai->dev, "mpll")); + ret = clk_prepare_enable(s3c2412_i2s.iis_cclk); + if (ret) { + s3c_i2sv2_remove(dai); + return ret; + } s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; @@ -85,6 +90,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable_unprepare(s3c2412_i2s.iis_cclk); + s3c_i2sv2_remove(dai); return 0; } -- 2.7.4

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[alsa-devel] [PATCH][resend] ASoC: rsnd: merge snd_pcm_ops::open and snd_soc_dai_ops::startup
by Kuninori Morimoto 27 Jul '17

27 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd driver has snd_pcm_ops::open and snd_soc_dai_ops::startup, but, these are called from soc_pcm_open in same time. This patch merge these into rsnd_soc_dai_startup() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 80 +++++++++++++++++++----------------------------- 1 file changed, 31 insertions(+), 49 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8dcf1a8..6fbbcbf 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -844,12 +844,28 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, ir, &ic); } -static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime, - struct snd_soc_dai *dai) +static struct snd_pcm_hardware rsnd_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint; + struct snd_pcm_runtime *runtime = substream->runtime; unsigned int max_channels = rsnd_rdai_channels_get(rdai); + int ret; int i; /* @@ -866,34 +882,26 @@ static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime, constraint->count = i + 1; } + snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, constraint); + snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + /* * Sampling Rate / Channel Limitation * It depends on Clock Master Mode */ - if (!rsnd_rdai_is_clk_master(rdai)) - return; - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rsnd_soc_hw_rule_rate, dai, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rsnd_soc_hw_rule_channels, dai, - SNDRV_PCM_HW_PARAM_RATE, -1); -} - -static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); - int ret; - - /* rsnd_io_to_runtime() is not yet enabled here */ - rsnd_soc_hw_constraint(substream->runtime, dai); + if (rsnd_rdai_is_clk_master(rdai)) { + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + rsnd_soc_hw_rule_rate, dai, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + rsnd_soc_hw_rule_channels, dai, + SNDRV_PCM_HW_PARAM_RATE, -1); + } /* * call rsnd_dai_call without spinlock @@ -1105,31 +1113,6 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) /* * pcm ops */ -static struct snd_pcm_hardware rsnd_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 64 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 32, - .fifo_size = 256, -}; - -static int rsnd_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - - return ret; -} - static int rsnd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -1159,7 +1142,6 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) } static struct snd_pcm_ops rsnd_pcm_ops = { - .open = rsnd_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = rsnd_hw_params, .hw_free = snd_pcm_lib_free_pages, -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: Fix Kabylake machine driver for pops and clicks
by Harsha Priya 26 Jul '17

26 Jul '17

This patch adds fixes for the following issues in Kabylake machine driver 1. crackling noise in rt5663 headset codec 2. enabling 4 slot IV feedback for max98927 speaker amp codec 3. pop noise while recording from rt5514 dmic codec Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 28 ++++++++++++---------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 3fe4a08..4c51965 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -319,7 +319,9 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); ret = snd_soc_dai_set_sysclk(codec_dai, RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); @@ -349,27 +351,27 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, 0, - RT5514_PLL1_S_BCLK, RT5514_AIF1_BCLK_FREQ, - RT5514_AIF1_SYSCLK_FREQ); + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { - dev_err(rtd->dev, "set bclk err: %d\n", ret); + dev_err(rtd->dev, "set sysclk err: %d\n", ret); return ret; } + } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_PLL1, RT5514_AIF1_SYSCLK_FREQ, - SND_SOC_CLOCK_IN); + if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { - dev_err(rtd->dev, "set sclk err: %d\n", ret); + dev_err(rtd->dev, "DEV0 TDM slot err:%d\n", ret); return ret; } } - if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME) || - !strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF0, 3, 8, 16); + + if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + dev_err(rtd->dev, "DEV1 TDM slot err:%d\n", ret); return ret; } } -- 1.9.1

4 3

[alsa-devel] intel sound device not detected
by fulgor fulgor 26 Jul '17

26 Jul '17

Hello I read a post from you in a mailling list I use manjaro with latest kernel but sound device is not detected could you help ? thanks fulgorit

1 0

[alsa-devel] [PATCH 00/11] Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Arvind Yadav (11): [PATCH 01/11] ASoC: samsung: s3c2412: Handle return value of clk_prepare_enable. [PATCH 02/11] ASoC: samsung: s3c24xx: Handle return value of clk_prepare_enable. [PATCH 03/11] ASoC: samsung: pcm: Handle return value of clk_prepare_enable. [PATCH 04/11] ASoC: samsung: i2s: Handle return value of clk_prepare_enable. [PATCH 05/11] ASoC: samsung: spdif: Handle return value of clk_prepare_enable. [PATCH 06/11] ASoC: mxs-saif: Handle return value of clk_prepare_enable/clk_prepare. [PATCH 07/11] ASoC: jz4740: Handle return value of clk_prepare_enable. [PATCH 08/11] ASoC: sun4i-spdif: Handle return value of clk_prepare_enable. [PATCH 09/11] ASoC: atmel: ac97c: Handle return value of clk_prepare_enable. [PATCH 10/11] ASoC: pxa: Handle return value of clk_prepare_enable. [PATCH 11/11] ASoC: pxa-ssp: Handle return value of clk_prepare_enable. sound/atmel/ac97c.c | 8 +++++--- sound/soc/jz4740/jz4740-i2s.c | 21 +++++++++++++++++---- sound/soc/mxs/mxs-saif.c | 13 ++++++++++--- sound/soc/pxa/pxa-ssp.c | 23 +++++++++++++++++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +++++- sound/soc/samsung/i2s.c | 19 +++++++++++++++---- sound/soc/samsung/pcm.c | 8 ++++++-- sound/soc/samsung/s3c2412-i2s.c | 4 +++- sound/soc/samsung/s3c24xx-i2s.c | 11 +++++++++-- sound/soc/samsung/spdif.c | 8 ++++++-- sound/soc/sunxi/sun4i-spdif.c | 11 ++++++++--- 11 files changed, 101 insertions(+), 31 deletions(-) -- 1.9.1

3 22

[alsa-devel] [PATCH v2 01/11] ASoC: samsung: s3c2412: Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- Chnage in v2 : Error handling for things done in s3c_i2sv2_probe(). sound/soc/samsung/s3c2412-i2s.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 0a47182..0b96927 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -65,13 +65,16 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - return PTR_ERR(s3c2412_i2s.iis_cclk); + ret = PTR_ERR(s3c2412_i2s.iis_cclk); + goto err; } /* Set MPLL as the source for IIS CLK */ clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_prepare_enable(s3c2412_i2s.iis_cclk); + ret = clk_prepare_enable(s3c2412_i2s.iis_cclk); + if (ret) + goto err; s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; @@ -80,6 +83,11 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) S3C_GPIO_PULL_NONE); return 0; + +err: + clk_disable(s3c2412_i2s.iis_pclk); + clk_put(s3c2412_i2s.iis_pclk); + return ret; } static int s3c2412_i2s_remove(struct snd_soc_dai *dai) -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: rt5665: add MONOVOL Playback Volume control
by Bard Liao 26 Jul '17

26 Jul '17

MONOVOL Playback Volume is a playback volume control on "MONOVOL MIX". Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 7e64b25..54ace3b 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1402,6 +1402,9 @@ static const struct snd_kcontrol_new rt5665_snd_controls[] = { RT5665_L_VOL_SFT, 15, 1, snd_soc_get_volsw, rt5665_mono_vol_put, mono_vol_tlv), + SOC_SINGLE_TLV("MONOVOL Playback Volume", RT5665_MONO_OUT, + RT5665_L_VOL_SFT, 39, 1, out_vol_tlv), + /* Output Volume */ SOC_DOUBLE_TLV("OUT Playback Volume", RT5665_LOUT, RT5665_L_VOL_SFT, RT5665_R_VOL_SFT, 39, 1, out_vol_tlv), -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: sh: hac: add missing "int ret"
by Kuninori Morimoto 26 Jul '17

26 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit b047e1cce8 ("ASoC: ac97: Support multi-platform AC'97") modified hac_soc_platform_probe(), but "int ret" was missed. This patch adds missing "int ret", otherwise, we will get linux/sound/soc/sh/hac.c: In function 'hac_soc_platform_probe': linux/sound/soc/sh/hac.c:318: error: 'ret' undeclared (first use in this function) linux/sound/soc/sh/hac.c:318: error: (Each undeclared identifier is reported only once linux/sound/soc/sh/hac.c:318: error: for each function it appears in.) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/hac.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 84c5103..624aaf5 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -315,6 +315,8 @@ static int hac_hw_params(struct snd_pcm_substream *substream, static int hac_soc_platform_probe(struct platform_device *pdev) { + int ret; + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); if (ret != 0) return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: don't use private_value on rsnd_kctrl_new()
by Kuninori Morimoto 26 Jul '17

26 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd kctrl is using both .private_value (for rsnd_kctrl_cfg) and .private_data (for rsnd_mod) on snd_kcontrol. But only 1 private data (= rsnd_kctrl_cfg) can be enough if it has rsnd_mod pointer. This patch doesn't use private_value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 14 ++++++-------- sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index d3381f3..c345623 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1151,11 +1151,10 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) /* * snd_kcontrol */ -#define kcontrol_to_cfg(kctrl) ((struct rsnd_kctrl_cfg *)kctrl->private_value) static int rsnd_kctrl_info(struct snd_kcontrol *kctrl, struct snd_ctl_elem_info *uinfo) { - struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl); if (cfg->texts) { uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -1181,7 +1180,7 @@ static int rsnd_kctrl_info(struct snd_kcontrol *kctrl, static int rsnd_kctrl_get(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *uc) { - struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl); int i; for (i = 0; i < cfg->size; i++) @@ -1196,8 +1195,7 @@ static int rsnd_kctrl_get(struct snd_kcontrol *kctrl, static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *uc) { - struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl); int i, change = 0; if (!cfg->accept(cfg->io)) @@ -1214,7 +1212,7 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, } if (change && cfg->update) - cfg->update(cfg->io, mod); + cfg->update(cfg->io, cfg->mod); return change; } @@ -1266,14 +1264,13 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, .index = rtd->num, .get = rsnd_kctrl_get, .put = rsnd_kctrl_put, - .private_value = (unsigned long)cfg, }; int ret; if (size > RSND_MAX_CHANNELS) return -EINVAL; - kctrl = snd_ctl_new1(&knew, mod); + kctrl = snd_ctl_new1(&knew, cfg); if (!kctrl) return -ENOMEM; @@ -1289,6 +1286,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->card = card; cfg->kctrl = kctrl; cfg->io = io; + cfg->mod = mod; return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 99c5761..c5de71f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -614,6 +614,7 @@ struct rsnd_kctrl_cfg { struct rsnd_dai_stream *io; struct snd_card *card; struct snd_kcontrol *kctrl; + struct rsnd_mod *mod; }; #define RSND_MAX_CHANNELS 8 -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Add the I2S ASRC support
by Oder Chiou 26 Jul '17

26 Jul '17

The patch adds the I2S ASRC support. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5514.h | 7 +++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a105c84..882fdc4 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -63,6 +63,7 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_SRC_CTRL, 0x44000eee}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, + {RT5514_ASRC_IN_CTRL1, 0x00000003}, }; static const struct reg_default rt5514_reg[] = { @@ -471,6 +472,15 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int rt5514_i2s_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + return (rt5514->sysclk > rt5514->lrck * 384); +} + static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -541,6 +551,10 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { RT5514_POW_PLL1_LDO_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL1", RT5514_PWR_ANA2, RT5514_POW_PLL1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ASRC AD1", 1, RT5514_CLK_CTRL2, + RT5514_CLK_AD0_ASRC_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ASRC AD2", 1, RT5514_CLK_CTRL2, + RT5514_CLK_AD1_ASRC_EN_BIT, 0, NULL, 0), /* ADC Mux */ SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, @@ -638,6 +652,7 @@ static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, { "Stereo1 ADC MIX", NULL, "adc stereo1 filter" }, { "adc stereo1 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll }, + { "adc stereo1 filter", NULL, "ASRC AD1", rt5514_i2s_use_asrc }, { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, @@ -654,6 +669,7 @@ static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { { "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXR" }, { "Stereo2 ADC MIX", NULL, "adc stereo2 filter" }, { "adc stereo2 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll }, + { "adc stereo2 filter", NULL, "ASRC AD2", rt5514_i2s_use_asrc }, { "AIF1TX", NULL, "Stereo1 ADC MIX"}, { "AIF1TX", NULL, "Stereo2 ADC MIX"}, @@ -706,6 +722,9 @@ static int rt5514_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_I2S_DL_MASK, val_len); + regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL1, + RT5514_CLK_AD_ANA1_SEL_MASK, + (pre_div + 1) << RT5514_CLK_AD_ANA1_SEL_SFT); regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL2, RT5514_CLK_SYS_DIV_OUT_MASK | RT5514_SEL_ADC_OSR_MASK, pre_div << RT5514_CLK_SYS_DIV_OUT_SFT | diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 75024a4..2dc40e6 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -37,6 +37,7 @@ #define RT5514_PLL3_CALIB_CTRL5 0x2124 #define RT5514_DELAY_BUF_CTRL1 0x2140 #define RT5514_DELAY_BUF_CTRL3 0x2148 +#define RT5514_ASRC_IN_CTRL1 0x2180 #define RT5514_DOWNFILTER0_CTRL1 0x2190 #define RT5514_DOWNFILTER0_CTRL2 0x2194 #define RT5514_DOWNFILTER0_CTRL3 0x2198 @@ -197,8 +198,14 @@ #define RT5514_CLK_AD0_EN_BIT 23 #define RT5514_CLK_DMIC_OUT_SEL_MASK (0x7 << 8) #define RT5514_CLK_DMIC_OUT_SEL_SFT 8 +#define RT5514_CLK_AD_ANA1_SEL_MASK (0xf << 0) +#define RT5514_CLK_AD_ANA1_SEL_SFT 0 /* RT5514_CLK_CTRL2 (0x2108) */ +#define RT5514_CLK_AD1_ASRC_EN (0x1 << 17) +#define RT5514_CLK_AD1_ASRC_EN_BIT 17 +#define RT5514_CLK_AD0_ASRC_EN (0x1 << 16) +#define RT5514_CLK_AD0_ASRC_EN_BIT 16 #define RT5514_CLK_SYS_DIV_OUT_MASK (0x7 << 8) #define RT5514_CLK_SYS_DIV_OUT_SFT 8 #define RT5514_SEL_ADC_OSR_MASK (0x7 << 4) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] constify codecs acpi_device_id.
by Arvind Yadav 26 Jul '17

26 Jul '17

acpi_device_id are not supposed to change at runtime. All functions working with acpi_device_id provided by <acpi/acpi_bus.h> work with const acpi_device_id. So mark the non-const structs as const. Arvind Yadav (4): [PATCH 1/4] ASoC: rt5514: constify acpi_device_id. [PATCH 2/4] ASoC: rt5663: constify acpi_device_id. [PATCH 3/4] ASoC: rt5659: constify acpi_device_id. [PATCH 4/4] ASoC: rt5665: constify acpi_device_id. sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH] ASoC: rt5514: Use the IS_ENABLED to supports the module build
by Oder Chiou 26 Jul '17

26 Jul '17

The patch uses the macro IS_ENABLED to make sure that the SPI function also supports the module build. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 882fdc4..7fffa7c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -31,7 +31,7 @@ #include "rl6231.h" #include "rt5514.h" -#if defined(CONFIG_SND_SOC_RT5514_SPI) +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) #include "rt5514-spi.h" #endif @@ -312,7 +312,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); if (fw) { -#if defined(CONFIG_SND_SOC_RT5514_SPI) +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_write(0x4ff60000, fw->data, ((fw->size/8)+1)*8); #else @@ -325,7 +325,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); if (fw) { -#if defined(CONFIG_SND_SOC_RT5514_SPI) +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_write(0x4ffc0000, fw->data, ((fw->size/8)+1)*8); #else -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5670: fix wrong name of sto2 adc 2 mux
by Bard Liao 26 Jul '17

26 Jul '17

The kcontrol name of Stereo2 ADC 2 Mux is not matched with what we used in dapm widget. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0376667..2866f9b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1175,7 +1175,7 @@ static const struct snd_kcontrol_new rt5670_sto_adc_2_mux = static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l_mux = +static const struct snd_kcontrol_new rt5670_sto2_adc_2_mux = SOC_DAPM_ENUM("Stereo2 ADC 2 Mux", rt5670_stereo2_adc2_enum); -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: remove cache_bypass from snd_soc_codec
by Kuninori Morimoto 26 Jul '17

26 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_codec .cache_bypass related operation code has been removed. Let's remove remaining code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 - sound/soc/codecs/max98927.c | 1 - 2 files changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dfc6dae..436fa0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -900,7 +900,6 @@ struct snd_soc_codec { struct list_head list; /* runtime */ - unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int cache_init:1; /* codec cache has been initialized */ /* codec IO */ diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..b0380b5 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -577,7 +577,6 @@ static int max98927_probe(struct snd_soc_codec *codec) max98927->codec = codec; codec->control_data = max98927->regmap; - codec->cache_bypass = 1; /* Software Reset */ regmap_write(max98927->regmap, -- 1.9.1

2 1

[alsa-devel] [PATCH 08/11] ASoC: sun4i-spdif: Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/sunxi/sun4i-spdif.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index eaefd07..4a316c0 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -458,11 +458,16 @@ static int sun4i_spdif_runtime_suspend(struct device *dev) static int sun4i_spdif_runtime_resume(struct device *dev) { struct sun4i_spdif_dev *host = dev_get_drvdata(dev); + int ret; - clk_prepare_enable(host->spdif_clk); - clk_prepare_enable(host->apb_clk); + ret = clk_prepare_enable(host->spdif_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->apb_clk); + if (ret) + clk_disable_unprepare(host->spdif_clk); - return 0; + return ret; } static int sun4i_spdif_probe(struct platform_device *pdev) -- 1.9.1

2 1

[alsa-devel] [PATCH 09/11] ASoC: atmel: ac97c: Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/atmel/ac97c.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 9d2c9d9..52b0522 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -783,7 +783,9 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_prepare_enable(pclk); + retval = clk_prepare_enable(pclk); + if (retval) + return retval; retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -897,9 +899,9 @@ static int atmel_ac97c_resume(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; + int ret = clk_prepare_enable(chip->pclk); - clk_prepare_enable(chip->pclk); - return 0; + return ret; } static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix missing sentinels in sst_acpi_mach
by Vinod Koul 26 Jul '17

26 Jul '17

Couple of instances of sst_acpi_mach were having missing sentinels so add them up Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..9e3f8c04dd32 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -941,6 +941,7 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = { .machine_quirk = sst_acpi_codec_list, .quirk_data = &bxt_codecs, }, + {} }; static struct sst_acpi_mach sst_kbl_devdata[] = { @@ -991,6 +992,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = { .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", }, + {} }; /* PCI IDs */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: board: Fix missing sentinel for bxt_board_id
by Pradeep Tewani 26 Jul '17

26 Jul '17

Without a sentinel in the array of platform devices, we can get a panic so add a sentinel. Full stack dump below: [ 231.564705] BUG: KASAN: global-out-of-bounds in platform_match+0xb0/0x146 [ 231.565500] Read of size 1 at addr ffffffff8380a1a0 by task swapper/0/1 [ 231.566280] [ 231.566594] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.12.0-00289-g152771f #1 [ 231.567526] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.9.3-20161025_171302-gandalf 04/01/2014 [ 231.568730] Call Trace: [ 231.569128] dump_stack+0xf8/0x14a [ 231.569615] print_address_description+0x57/0x1e4 [ 231.570216] ? driver_probe_device+0x814/0x814 [ 231.570798] kasan_report+0x1cb/0x1eb [ 231.571302] ? platform_match+0xb0/0x146 [ 231.571833] __asan_load1+0x45/0x47 [ 231.572324] platform_match+0xb0/0x146 [ 231.572843] ? platform_uevent+0x73/0x73 [ 231.573370] __driver_attach+0x47/0x16a [ 231.573890] bus_for_each_dev+0x11a/0x15c [ 231.574422] ? bus_remove_file+0x57/0x57 [ 231.574951] ? do_raw_spin_unlock+0xc3/0xe0 [ 231.575501] driver_attach+0x2b/0x2e [ 231.576002] bus_add_driver+0x25e/0x3c5 [ 231.576520] ? set_debug_rodata+0x20/0x20 [ 231.584776] driver_register+0x13e/0x19d [ 231.585305] __platform_driver_register+0x6c/0x71 [ 231.585908] ? skl_driver_init+0x20/0x20 [ 231.586436] broxton_audio_init+0x17/0x19 [ 231.586976] do_one_initcall+0x11a/0x221 [ 231.587499] ? start_kernel+0x525/0x525 [ 231.588020] ? lock_downgrade+0x281/0x2ec [ 231.588557] ? trace_hardirqs_on_caller+0x223/0x267 [ 231.589169] ? _raw_spin_unlock_irqrestore+0x49/0x55 [ 231.589791] ? set_debug_rodata+0x20/0x20 [ 231.590324] kernel_init_freeable+0x1fd/0x2b0 [ 231.590891] ? rest_init+0x17c/0x17c [ 231.591384] kernel_init+0x11/0x157 [ 231.591876] ? rest_init+0x17c/0x17c [ 231.592371] ret_from_fork+0x2a/0x40 [ 231.592865] [ 231.593166] The buggy address belongs to the variable: [ 231.593806] bxt_board_ids+0x40/0x9e0 Fixes: 152771fbc3 ("ASoC: Intel: board: Add Geminilake platform support") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Pradeep Tewani <pradeep.d.tewani(a)intel.com> --- sound/soc/intel/boards/bxt_rt298.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index a4b5bcb..7843104 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -595,6 +595,7 @@ static const struct platform_device_id bxt_board_ids[] = { (unsigned long)&broxton_rt298 }, { .name = "glk_alc298s_i2s", .driver_data = (unsigned long)&geminilake_rt298 }, + {} }; static struct platform_driver broxton_audio = { -- 2.7.4

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5670: merge ADC L/R Mux
by Bard Liao 26 Jul '17

26 Jul '17

Some ADC L/R Muxes share the some control bit. So, merge them. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 42 ++++++++++++++++-------------------------- 1 file changed, 16 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0ec7985..2866f9b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1151,20 +1151,15 @@ static const char * const rt5670_stereo_adc1_src[] = { static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); -static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = - SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); - -static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = - SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); +static const struct snd_kcontrol_new rt5670_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo1 ADC 1 Mux", rt5670_stereo1_adc1_enum); static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = - SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); +static const struct snd_kcontrol_new rt5670_sto2_adc_1_mux = + SOC_DAPM_ENUM("Stereo2 ADC 1 Mux", rt5670_stereo2_adc1_enum); -static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = - SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); /* MX-27 MX-26 [11] */ static const char * const rt5670_stereo_adc2_src[] = { @@ -1174,20 +1169,15 @@ static const char * const rt5670_stereo_adc2_src[] = { static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = - SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); - -static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = - SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); +static const struct snd_kcontrol_new rt5670_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo1 ADC 2 Mux", rt5670_stereo1_adc2_enum); static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = - SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); +static const struct snd_kcontrol_new rt5670_sto2_adc_2_mux = + SOC_DAPM_ENUM("Stereo2 ADC 2 Mux", rt5670_stereo2_adc2_enum); -static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = - SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); /* MX-27 MX26 [10] */ static const char * const rt5670_stereo_adc_src[] = { @@ -1642,23 +1632,23 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto1_dmic_mux), SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_l2_mux), + &rt5670_sto_adc_2_mux), SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_r2_mux), + &rt5670_sto_adc_2_mux), SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_l1_mux), + &rt5670_sto_adc_1_mux), SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_r1_mux), + &rt5670_sto_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto2_dmic_mux), SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_l2_mux), + &rt5670_sto2_adc_2_mux), SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_r2_mux), + &rt5670_sto2_adc_2_mux), SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_l1_mux), + &rt5670_sto2_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_r1_mux), + &rt5670_sto2_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto2_adc_lr_mux), SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, -- 2.7.4

2 2

[alsa-devel] [PATCH v4 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 26 Jul '17

26 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. v4: - initialized the clk_init_data structure, - switched to of_clk_parent_fill instead of manually setting parent_names, - removed include of clkdev.h in favor of slab.h, - removed unnecessary second cast in operation, - used clamp instead of two conditions, - removed dependency on clock-output-names, the name is gotten from DT node, - merged all three drivers for audio PLLs (FRAC, PMC, PAD) into one, v3: - added a flag per generated clock to know which one is allowed to modify audio_pll rate, - set the flag in setup function depending on the compatible and the clock ID, v2: - split big patch containing dt-binding, pll drivers and classd modifications, - removed unused AUDIO_PLL_*FOUT* defines from clk-audio-pll-pmc, - added conditions for audio pll rate setting restriction for SSC and I2S, - renamed all mentions of ACLK to GCLK in classD driver, Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop dt-bindings: clk: at91: add audio plls to the compatible list clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock Documentation/devicetree/bindings/clock/at91-clock.txt | 10 +- Documentation/devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 +- arch/arm/boot/dts/sama5d2.dtsi | 39 +- arch/arm/mach-at91/Kconfig | 4 +- drivers/clk/at91/Makefile | 1 +- drivers/clk/at91/clk-audio-pll.c | 531 ++++++++- drivers/clk/at91/clk-generated.c | 101 +- include/linux/clk/at91_pmc.h | 25 +- sound/soc/atmel/atmel-classd.c | 47 +- 10 files changed, 724 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll.c base-commit: 047a0f692a9354d730dad30f25e1ebd8437b32b1 -- git-series 0.9.1

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[alsa-devel] [PATCH v3 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 26 Jul '17

26 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop clk: at91: add audio plls to the compatible list in DT binding clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock .../devicetree/bindings/clock/at91-clock.txt | 10 + .../devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 ++ arch/arm/boot/dts/sama5d2.dtsi | 39 +++- arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 2 + drivers/clk/at91/clk-audio-pll-pad.c | 206 ++++++++++++++++++ drivers/clk/at91/clk-audio-pll-pmc.c | 174 +++++++++++++++ drivers/clk/at91/clk-audio-pll.c | 239 +++++++++++++++++++++ drivers/clk/at91/clk-generated.c | 101 +++++++-- include/linux/clk/at91_pmc.h | 25 +++ sound/soc/atmel/atmel-classd.c | 47 ++-- 12 files changed, 813 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll-pad.c create mode 100644 drivers/clk/at91/clk-audio-pll-pmc.c create mode 100644 drivers/clk/at91/clk-audio-pll.c -- 2.11.0

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[alsa-devel] [PATCH] ALSA: hda - Add mute led support for HP ProBook 440 G4
by Kai-Heng Feng 26 Jul '17

26 Jul '17

Mic mute led does not work on HP ProBook 440 G4. We can use CXT_FIXUP_MUTE_LED_GPIO fixup to support it. BugLink: https://bugs.launchpad.net/bugs/1705586 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e8253737c47a..615a69a32970 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -931,6 +931,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.13.3

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[alsa-devel] [PATCH v2 0/2] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 25 Jul '17

25 Jul '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (2): ASoC: sun4i-i2s: Add more quirks for newer SoCs ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 469 ++++++++++++++++++--- 2 files changed, 407 insertions(+), 64 deletions(-) -- 2.13.3

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[alsa-devel] [ANNOUNCE] Audio Mini Summit 2017 at Prague, Oct 27
by Takashi Iwai 25 Jul '17

25 Jul '17

Hi, this is an announcement of Audio Mini Summit 2017. It's an annual meeting we regularly have along with major Linux conferences, and this year, we'll hold the meeting at Prague (Czech Republic) on Friday, October 27, right after GStreamer Summit, ELCE and Open Source Summit Europe. We planned a whole-day meeting, and will likely have an evening dinner event, too. Please note that the meeting will be held in a conference room of SUSE Prague office, not in Hilton hotel where ELCE/OSSE is held. It's located near Hilton, just a couple of blocks away. The address of SUSE office is: Corso II, Křižíkova 148/34, 18600 Praha 8 https://goo.gl/maps/anET1yh3XXu Also, since the conference room is hosted by SUSE, it's an open event at this time; that is, you can attend only this meeting without registration for other Linux Foundation events in that week, too. If you are interested in the event, please fill up the registration form in Google doc spreadsheet below: https://docs.google.com/spreadsheets/d/1a-RaYtTVVtgh_cyRN5GwATiixppGCMEwqw6… Just add yourself in the attendee list from the web page. And, now we are gathering the topics to discuss, too. If you have any interesting topics to be discussed in the meeting, feel free to post to alsa-devel ML. I'm going to write up later. thanks, Takashi

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Re: [alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Hsin-yu Chao 25 Jul '17

25 Jul '17

The original cause seems to be in spi driver. I tested setting enable_dma flag to false in spi-pxa2xx driver and that fixes the problem. Will investigate in spi driver side. Thanks! Hsin-yu On Mon, Jul 24, 2017 at 11:06 AM, Oder Chiou <oder_chiou(a)realtek.com> wrote: > > On Fri, Jul 21, 2017 at 04:00:47PM +0800, Oder Chiou wrote: > > > The patch uses the local variable to get the data from SPI device to > > > prevent to get all zero data. > > > > How do you end up with all zero data from reading directly into rxbuf? > > > While the "runtime->dma_area" put into the rxbuf directly on the Intel > platform, it seemed that had the limitation to let the memory cannot be > written the data using the driver of the SPI controller. We verified the > original code on the platforms of Nvidia and Rockchip, and these are > working fine. >

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