From c62db3d5abf89ca5502c7fe1f869c2862e48336d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 12 Jan 2016 15:53:46 +0800 Subject: ASoC: rt5659: Staticise rt5659_i2c_shutdown Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5659.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 820d8fa62b5e..47c717f4964a 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4184,7 +4184,7 @@ static int rt5659_i2c_remove(struct i2c_client *i2c) return 0; } -void rt5659_i2c_shutdown(struct i2c_client *client) +static void rt5659_i2c_shutdown(struct i2c_client *client) { struct rt5659_priv *rt5659 = i2c_get_clientdata(client); -- cgit v1.2.3 From 712a8038cc24dba668afe82f0413714ca87184e0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 27 Jan 2016 14:26:18 +0100 Subject: ASoC: ssm4567: Reset device before regcache_sync() When the ssm4567 is powered up the driver calles regcache_sync() to restore the register map content. regcache_sync() assumes that the device is in its power-on reset state. Make sure that this is the case by explicitly resetting the ssm4567 register map before calling regcache_sync() otherwise we might end up with a incorrect register map which leads to undefined behaviour. One such undefined behaviour was observed when returning from system suspend while a playback stream is active, in that case the ssm4567 was kept muted after resume. Fixes: 1ee44ce03011 ("ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier") Reported-by: Harsha Priya Tested-by: Fang, Yang A Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index e619d5651b09..080c78e88e10 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -352,6 +352,11 @@ static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) regcache_cache_only(ssm4567->regmap, !enable); if (enable) { + ret = regmap_write(ssm4567->regmap, SSM4567_REG_SOFT_RESET, + 0x00); + if (ret) + return ret; + ret = regmap_update_bits(ssm4567->regmap, SSM4567_REG_POWER_CTRL, SSM4567_POWER_SPWDN, 0x00); -- cgit v1.2.3 From ba4bc32eaa39ba7687f0958ae90eec94da613b46 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 25 Jan 2016 18:07:33 +0100 Subject: ASoC: s3c24xx: use const snd_soc_component_driver pointer An older patch to convert the API in the s3c i2s driver ended up passing a const pointer into a function that takes a non-const pointer, so we now get a warning: sound/soc/samsung/s3c2412-i2s.c: In function 's3c2412_iis_dev_probe': sound/soc/samsung/s3c2412-i2s.c:172:9: error: passing argument 3 of 's3c_i2sv2_register_component' discards 'const' qualifier from pointer target type [-Werror=discarded-qualifiers] However, the s3c_i2sv2_register_component() function again passes the pointer into another function taking a const, so we just need to change its prototype. Fixes: eca3b01d0885 ("ASoC: switch over to use snd_soc_register_component() on s3c i2s") Signed-off-by: Arnd Bergmann Reviewed-by: Krzysztof Kozlowski Signed-off-by: Mark Brown --- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c-i2s-v2.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index df65c5b494b1..b6ab3fc5789e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -709,7 +709,7 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) #endif int s3c_i2sv2_register_component(struct device *dev, int id, - struct snd_soc_component_driver *cmp_drv, + const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index 90abab364b49..d0684145ed1f 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -101,7 +101,7 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, * soc core. */ extern int s3c_i2sv2_register_component(struct device *dev, int id, - struct snd_soc_component_driver *cmp_drv, + const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv); #endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ -- cgit v1.2.3 From 6049af00fc2fac8d27f8bd064ff68b16991a80f7 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Mon, 22 Feb 2016 15:56:55 +0800 Subject: ASoC: rt5640: add master clock handling for rt5640 enable/disable master clock when codec is active or not. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5640.txt | 3 +++ sound/soc/codecs/rt5640.c | 31 ++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ 3 files changed, 36 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 9e62f6eb348f..57fe64643050 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -12,6 +12,9 @@ Required properties: Optional properties: +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + - realtek,in1-differential - realtek,in2-differential - realtek,in3-differential diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 11d032cdc658..6cd84fb2196a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1949,7 +1949,33 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* + * SND_SOC_BIAS_PREPARE is called while preparing for a + * transition to ON or away from ON. If current bias_level + * is SND_SOC_BIAS_ON, then it is preparing for a transition + * away from ON. Disable the clock in that case, otherwise + * enable it. + */ + if (IS_ERR(rt5640->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5640->mclk); + } else { + ret = clk_prepare_enable(rt5640->mclk); + if (ret) + return ret; + } + break; + case SND_SOC_BIAS_STANDBY: if (SND_SOC_BIAS_OFF == snd_soc_codec_get_bias_level(codec)) { snd_soc_update_bits(codec, RT5640_PWR_ANLG1, @@ -2088,6 +2114,11 @@ static int rt5640_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* Check if MCLK provided */ + rt5640->mclk = devm_clk_get(codec->dev, "mclk"); + if (PTR_ERR(rt5640->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + rt5640->codec = codec; snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 83a7150ddc24..1761c3a98b76 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -12,6 +12,7 @@ #ifndef _RT5640_H #define _RT5640_H +#include #include /* Info */ @@ -2097,6 +2098,7 @@ struct rt5640_priv { struct snd_soc_codec *codec; struct rt5640_platform_data pdata; struct regmap *regmap; + struct clk *mclk; int sysclk; int sysclk_src; -- cgit v1.2.3 From c467fc0e010b66069c0d5bb3e8e869adf267115f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Mar 2016 15:09:37 +0800 Subject: ASoC: rt5640: Set PLL src according to source rt5640_set_dai_pll set pll source according to given source and dai id. However, the pll source should be set according to given source only. Signed-off-by: Jack Yu Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6cd84fb2196a..863c190c5076 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1902,21 +1902,12 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_MCLK); break; case RT5640_PLL1_S_BCLK1: + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); + break; case RT5640_PLL1_S_BCLK2: - dai_sel = get_sdp_info(codec, dai->id); - if (dai_sel < 0) { - dev_err(codec->dev, - "Failed to get sdp info: %d\n", dai_sel); - return -EINVAL; - } - if (dai_sel & RT5640_U_IF1) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); - } - if (dai_sel & RT5640_U_IF2) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); - } + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); break; default: dev_err(codec->dev, "Unknown PLL source %d\n", source); -- cgit v1.2.3 From 57586fb76471c5fc521b2e9c3a4eae259b06d479 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Mar 2016 15:09:38 +0800 Subject: ASoC: rt5640: add supplys for dac power The DAC1/2 power is for both DACs and related mixer/mux. Add SUPPLY type widgets to support it. Signed-off-by: Jack Yu Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 39 ++++++++++++++++++++++++--------------- 1 file changed, 24 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 863c190c5076..af9b5f18f0e0 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1217,11 +1217,14 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, rt5640_dig_r_mix, ARRAY_SIZE(rt5640_dig_r_mix)), /* DACs */ - SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R1_BIT, 0), - + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R1_BIT, 0, NULL, 0), /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), @@ -1298,9 +1301,9 @@ static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + SND_SOC_DAPM_DAC("DAC R2", NULL, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + SND_SOC_DAPM_DAC("DAC L2", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, @@ -1317,6 +1320,10 @@ static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC L2 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), @@ -1328,11 +1335,6 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), - SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, @@ -1493,8 +1495,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXL", "Stereo ADC Switch", "Stereo ADC MIXL"}, {"DAC MIXL", "INF1 Switch", "IF1 DAC L"}, + {"DAC MIXL", NULL, "DAC L1 Power"}, {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, + {"DAC MIXR", NULL, "DAC R1 Power"}, {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, @@ -1507,8 +1511,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC L1", NULL, "DAC L1 Power"}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R1", NULL, "DAC R1 Power"}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, @@ -1595,8 +1601,9 @@ static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { {"DAC L2 Mux", "IF2", "IF2 DAC L"}, {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - + {"DAC L2 Mux", NULL, "DAC L2 Power"}, {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + {"DAC R2 Mux", NULL, "DAC R2 Power"}, {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"Stereo DAC MIXL", "ANC Switch", "ANC"}, @@ -1614,8 +1621,10 @@ static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { {"DAC L2", NULL, "Mono DAC MIXL"}, {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC L2", NULL, "DAC L2 Power"}, {"DAC R2", NULL, "Mono DAC MIXR"}, {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "DAC R2 Power"}, {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, @@ -1656,8 +1665,8 @@ static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, - {"IF2 DAC L", NULL, "DAC L2 Filter"}, - {"IF2 DAC R", NULL, "DAC R2 Filter"}, + {"IF2 DAC L", NULL, "DAC L2 Power"}, + {"IF2 DAC R", NULL, "DAC R2 Power"}, }; static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) -- cgit v1.2.3 From c77dd678a7659d58625bd3b3a36d1329d9e7d44c Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Tue, 8 Mar 2016 13:35:31 +0530 Subject: ASoC: rt5640: remove unused variable We are getting build warning about: sound/soc/codecs/rt5640.c:1892:11: warning: unused variable 'dai_sel' The use of the variable was removed but the variable itself was not removed. Fixes: c467fc0e010b ("ASoC: rt5640: Set PLL src according to source") Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index af9b5f18f0e0..e8b5ba04417a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1889,7 +1889,7 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); struct rl6231_pll_code pll_code; - int ret, dai_sel; + int ret; if (source == rt5640->pll_src && freq_in == rt5640->pll_in && freq_out == rt5640->pll_out) -- cgit v1.2.3