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

[alsa-devel] [PATCH 0/2] ASoC: hdmi-codec: Allow driver to restrict sample size
by srinivas.kandagatla＠linaro.org 27 Jul '17

27 Jul '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patch adds option to allow hdmi-codec drivers to restrict the audio sample size based on the values that are suppored in hardware. First patch is adding that formats option to hdmi_codec_pdata and second patch is a fix in adv7511 codec driver which only supports 16 and 24 bit samples. Srinivas Kandagatla (2): ASoC: hdmi-codec: Allow drivers to restrict sample sizes. drm/bridge: adv7511: restrict audio sample sizes drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 2 ++ include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 3 +++ 3 files changed, 6 insertions(+) -- 2.11.0

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[alsa-devel] [PATCHv2 0/6] Motorola Droid 4 Audio Support
by Sebastian Reichel 27 Jul '17

27 Jul '17

Hi, This adds audio support to Motorola Droid 4. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (actually untested, but probably needs some quirks) Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: sound: add motorola,cpcap-audio-codec ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio ARM: dts: omap4-droid4: add soundcard Tony Lindgren (1): ALSA: pcm: Export soc_dpcm_runtime_update .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 7 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1607 ++++++++++++++++++++ sound/soc/soc-pcm.c | 2 + 7 files changed, 1726 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

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Re: [alsa-devel] [PATCH v1 5/6] ASoC: codecs: msm8916-wcd-analog: add MBHC support
by Damien Riegel 27 Jul '17

27 Jul '17

[Sorry for breaking the thread, but I don't have the initial message to reply to.] I tried your patchset and faced some issues regarding removal detection: the driver detects headphone and microphone removal as soon as I press a button on the headset. evtest logs: [Headset insertion] Event: time 10207.841157, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 10207.841157, -------------- SYN_REPORT ------------ Event: time 10208.039105, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 0 Event: time 10208.039105, -------------- SYN_REPORT ------------ Event: time 10208.075519, type 5 (EV_SW), code 2 (SW_HEADPHONE_INSERT), value 1 Event: time 10208.075519, type 5 (EV_SW), code 4 (SW_MICROPHONE_INSERT), value 1 Event: time 10208.075519, -------------- SYN_REPORT ------------ [Pressing Volume up Button] Event: time 10210.590403, type 1 (EV_KEY), code 115 (KEY_VOLUMEUP), value 1 Event: time 10210.590403, type 5 (EV_SW), code 2 (SW_HEADPHONE_INSERT), value 0 Event: time 10210.590403, type 5 (EV_SW), code 4 (SW_MICROPHONE_INSERT), value 0 Event: time 10210.590403, -------------- SYN_REPORT ------------ Event: time 10210.719560, type 1 (EV_KEY), code 115 (KEY_VOLUMEUP), value 0 Event: time 10210.719560, -------------- SYN_REPORT ------------ [Unplugging] Event: time 10221.926156, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 10221.926156, -------------- SYN_REPORT ------------ Event: time 10222.026466, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 0 Event: time 10222.026466, -------------- SYN_REPORT ------------ Also, the KEY_MEDIA is wrongly reported but I don't see that every time, I think it depends on how fast I plug in the headset. Anyway, I think that can easily be filtered out by testing the presence of the microphone. Regards, -- Damien

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[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

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[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

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[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

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[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

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[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

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[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

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