From 03ba791df98d15d07ea74075122af71e35c7611c Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 11 May 2017 13:44:38 +0200 Subject: ASoC: rt5514: fix gcc-7 warning gcc-7 warns that there is a duplicate 'const' specifier in some variables that are declared using the SOC_ENUM_SINGLE_DECL macro: sound/soc/codecs/rt5514.c:398:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( sound/soc/codecs/rt5514.c:405:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( This removes one to fix the warning. Fixes: 4a6180ea7399 ("ASoC: rt5514: add rt5514 codec driver") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b1ddf0..28ab9e2bb051 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -395,14 +395,14 @@ static const char * const rt5514_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); static const struct snd_kcontrol_new rt5514_sto1_dmic_mux = SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); -- cgit v1.2.3 From 27a655c4bd8d9851c0f2ef9ec0d3793d068acbe9 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 11 May 2017 13:44:39 +0200 Subject: ASoC: rt5665: fix gcc-7 warning gcc-7 warns that there is a duplicate 'const' specifier in some variables that are declared using the SOC_ENUM_SINGLE_DECL macro: sound/soc/codecs/rt5665.c:915:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_01_adc_enum, sound/soc/codecs/rt5665.c:918:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_23_adc_enum, sound/soc/codecs/rt5665.c:921:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_45_adc_enum, sound/soc/codecs/rt5665.c:924:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_67_adc_enum, ... This removes one to fix the 68 warnings in this file Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 136 +++++++++++++++++++++++----------------------- 1 file changed, 68 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd22307f5b6..14b0cf89edf5 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -912,46 +912,46 @@ static const char * const rt5665_data_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_01_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_01_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT01_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_23_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_23_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT23_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_45_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_45_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT45_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_67_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_67_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT67_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_01_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_01_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT01_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_23_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_23_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT23_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_45_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_45_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT45_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_67_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_67_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT67_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_1_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_1_dac_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_1_DAC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_1_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_1_adc_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_1_ADC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_2_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_2_dac_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_2_DAC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_2_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_2_adc_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_2_ADC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if3_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if3_dac_enum, RT5665_DIG_INF3_DATA, RT5665_IF3_DAC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if3_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if3_adc_enum, RT5665_DIG_INF3_DATA, RT5665_IF3_ADC_SEL_SFT, rt5665_data_select); static const struct snd_kcontrol_new rt5665_if1_1_01_adc_swap_mux = @@ -1819,14 +1819,14 @@ static const char * const rt5665_dac2_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF2_2 DAC", "IF3 DAC", "Mono ADC MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_l2_enum, RT5665_DAC2_CTRL, RT5665_DAC_L2_SEL_SFT, rt5665_dac2_src); static const struct snd_kcontrol_new rt5665_dac_l2_mux = SOC_DAPM_ENUM("Digital DAC L2 Source", rt5665_dac_l2_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_r2_enum, RT5665_DAC2_CTRL, RT5665_DAC_R2_SEL_SFT, rt5665_dac2_src); @@ -1839,14 +1839,14 @@ static const char * const rt5665_dac3_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF2_2 DAC", "IF3 DAC", "STO2 ADC MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_l3_enum, RT5665_DAC3_CTRL, RT5665_DAC_L3_SEL_SFT, rt5665_dac3_src); static const struct snd_kcontrol_new rt5665_dac_l3_mux = SOC_DAPM_ENUM("Digital DAC L3 Source", rt5665_dac_l3_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_r3_enum, RT5665_DAC3_CTRL, RT5665_DAC_R3_SEL_SFT, rt5665_dac3_src); @@ -1859,14 +1859,14 @@ static const char * const rt5665_sto1_adc1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc1l_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC1L_SRC_SFT, rt5665_sto1_adc1_src); static const struct snd_kcontrol_new rt5665_sto1_adc1l_mux = SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5665_sto1_adc1l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc1r_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC1R_SRC_SFT, rt5665_sto1_adc1_src); @@ -1879,14 +1879,14 @@ static const char * const rt5665_sto1_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L", "ADC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adcl_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADCL_SRC_SFT, rt5665_sto1_adc_src); static const struct snd_kcontrol_new rt5665_sto1_adcl_mux = SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5665_sto1_adcl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adcr_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADCR_SRC_SFT, rt5665_sto1_adc_src); @@ -1899,14 +1899,14 @@ static const char * const rt5665_sto1_adc2_src[] = { "DAC MIX", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc2l_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC2L_SRC_SFT, rt5665_sto1_adc2_src); static const struct snd_kcontrol_new rt5665_sto1_adc2l_mux = SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5665_sto1_adc2l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc2r_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC2R_SRC_SFT, rt5665_sto1_adc2_src); @@ -1919,7 +1919,7 @@ static const char * const rt5665_sto1_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_dmic_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_DMIC_SRC_SFT, rt5665_sto1_dmic_src); @@ -1931,7 +1931,7 @@ static const char * const rt5665_sto1_dd_l_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_dd_l_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_DD_L_SRC_SFT, rt5665_sto1_dd_l_src); @@ -1943,7 +1943,7 @@ static const char * const rt5665_sto1_dd_r_src[] = { "STO2 DAC", "MONO DAC", "AEC REF" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_dd_r_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_DD_R_SRC_SFT, rt5665_sto1_dd_r_src); @@ -1956,7 +1956,7 @@ static const char * const rt5665_mono_adc_l2_src[] = { "DAC MIXL", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_l2_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_L2_SRC_SFT, rt5665_mono_adc_l2_src); @@ -1970,7 +1970,7 @@ static const char * const rt5665_mono_adc_l1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_l1_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_L1_SRC_SFT, rt5665_mono_adc_l1_src); @@ -1982,14 +1982,14 @@ static const char * const rt5665_mono_dd_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dd_l_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DD_L_SRC_SFT, rt5665_mono_dd_src); static const struct snd_kcontrol_new rt5665_mono_dd_l_mux = SOC_DAPM_ENUM("Mono DD L Source", rt5665_mono_dd_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dd_r_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DD_R_SRC_SFT, rt5665_mono_dd_src); @@ -2002,14 +2002,14 @@ static const char * const rt5665_mono_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L", "ADC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_l_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_L_SRC_SFT, rt5665_mono_adc_src); static const struct snd_kcontrol_new rt5665_mono_adc_l_mux = SOC_DAPM_ENUM("Mono ADC L Source", rt5665_mono_adc_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adcr_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_R_SRC_SFT, rt5665_mono_adc_src); @@ -2022,7 +2022,7 @@ static const char * const rt5665_mono_dmic_l_src[] = { "DMIC1 L", "DMIC2 L" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dmic_l_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DMIC_L_SRC_SFT, rt5665_mono_dmic_l_src); @@ -2035,7 +2035,7 @@ static const char * const rt5665_mono_adc_r2_src[] = { "DAC MIXR", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_r2_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_R2_SRC_SFT, rt5665_mono_adc_r2_src); @@ -2048,7 +2048,7 @@ static const char * const rt5665_mono_adc_r1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_r1_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_R1_SRC_SFT, rt5665_mono_adc_r1_src); @@ -2061,7 +2061,7 @@ static const char * const rt5665_mono_dmic_r_src[] = { "DMIC1 R", "DMIC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dmic_r_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DMIC_R_SRC_SFT, rt5665_mono_dmic_r_src); @@ -2075,14 +2075,14 @@ static const char * const rt5665_sto2_adc1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc1l_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC1L_SRC_SFT, rt5665_sto2_adc1_src); static const struct snd_kcontrol_new rt5665_sto2_adc1l_mux = SOC_DAPM_ENUM("Stereo2 ADC1L Source", rt5665_sto2_adc1l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc1r_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC1R_SRC_SFT, rt5665_sto2_adc1_src); @@ -2095,14 +2095,14 @@ static const char * const rt5665_sto2_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adcl_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADCL_SRC_SFT, rt5665_sto2_adc_src); static const struct snd_kcontrol_new rt5665_sto2_adcl_mux = SOC_DAPM_ENUM("Stereo2 ADCL Source", rt5665_sto2_adcl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adcr_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADCR_SRC_SFT, rt5665_sto2_adc_src); @@ -2115,14 +2115,14 @@ static const char * const rt5665_sto2_adc2_src[] = { "DAC MIX", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc2l_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC2L_SRC_SFT, rt5665_sto2_adc2_src); static const struct snd_kcontrol_new rt5665_sto2_adc2l_mux = SOC_DAPM_ENUM("Stereo2 ADC2L Source", rt5665_sto2_adc2l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc2r_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC2R_SRC_SFT, rt5665_sto2_adc2_src); @@ -2135,7 +2135,7 @@ static const char * const rt5665_sto2_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_dmic_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_DMIC_SRC_SFT, rt5665_sto2_dmic_src); @@ -2147,7 +2147,7 @@ static const char * const rt5665_sto2_dd_l_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_dd_l_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_DD_L_SRC_SFT, rt5665_sto2_dd_l_src); @@ -2159,7 +2159,7 @@ static const char * const rt5665_sto2_dd_r_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_dd_r_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_DD_R_SRC_SFT, rt5665_sto2_dd_r_src); @@ -2172,14 +2172,14 @@ static const char * const rt5665_dac1_src[] = { "IF1 DAC1", "IF2_1 DAC", "IF2_2 DAC", "IF3 DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_r1_enum, RT5665_AD_DA_MIXER, RT5665_DAC1_R_SEL_SFT, rt5665_dac1_src); static const struct snd_kcontrol_new rt5665_dac_r1_mux = SOC_DAPM_ENUM("DAC R1 Source", rt5665_dac_r1_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_l1_enum, RT5665_AD_DA_MIXER, RT5665_DAC1_L_SEL_SFT, rt5665_dac1_src); @@ -2192,14 +2192,14 @@ static const char * const rt5665_dig_dac_mix_src[] = { "Stereo1 DAC Mixer", "Stereo2 DAC Mixer", "Mono DAC Mixer" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dig_dac_mixl_enum, RT5665_A_DAC1_MUX, RT5665_DAC_MIX_L_SFT, rt5665_dig_dac_mix_src); static const struct snd_kcontrol_new rt5665_dig_dac_mixl_mux = SOC_DAPM_ENUM("DAC Digital Mixer L Source", rt5665_dig_dac_mixl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dig_dac_mixr_enum, RT5665_A_DAC1_MUX, RT5665_DAC_MIX_R_SFT, rt5665_dig_dac_mix_src); @@ -2212,14 +2212,14 @@ static const char * const rt5665_alg_dac1_src[] = { "Stereo1 DAC Mixer", "DAC1", "DMIC1" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_l1_enum, RT5665_A_DAC1_MUX, RT5665_A_DACL1_SFT, rt5665_alg_dac1_src); static const struct snd_kcontrol_new rt5665_alg_dac_l1_mux = SOC_DAPM_ENUM("Analog DAC L1 Source", rt5665_alg_dac_l1_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_r1_enum, RT5665_A_DAC1_MUX, RT5665_A_DACR1_SFT, rt5665_alg_dac1_src); @@ -2232,14 +2232,14 @@ static const char * const rt5665_alg_dac2_src[] = { "Mono DAC Mixer", "DAC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_l2_enum, RT5665_A_DAC2_MUX, RT5665_A_DACL2_SFT, rt5665_alg_dac2_src); static const struct snd_kcontrol_new rt5665_alg_dac_l2_mux = SOC_DAPM_ENUM("Analog DAC L2 Source", rt5665_alg_dac_l2_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_r2_enum, RT5665_A_DAC2_MUX, RT5665_A_DACR2_SFT, rt5665_alg_dac2_src); @@ -2253,7 +2253,7 @@ static const char * const rt5665_if2_1_adc_in_src[] = { "IF1 DAC2", "IF2_2 DAC", "IF3 DAC", "DAC1 MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if2_1_adc_in_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_1_ADC_IN_SFT, rt5665_if2_1_adc_in_src); @@ -2266,7 +2266,7 @@ static const char * const rt5665_if2_2_adc_in_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF3 DAC", "DAC1 MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if2_2_adc_in_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_2_ADC_IN_SFT, rt5665_if2_2_adc_in_src); @@ -2280,7 +2280,7 @@ static const char * const rt5665_if3_adc_in_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF2_2 DAC", "DAC1 MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if3_adc_in_enum, RT5665_DIG_INF3_DATA, RT5665_IF3_ADC_IN_SFT, rt5665_if3_adc_in_src); @@ -2293,14 +2293,14 @@ static const char * const rt5665_pdm_src[] = { "Stereo1 DAC", "Stereo2 DAC", "Mono DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_pdm_l_enum, RT5665_PDM_OUT_CTRL, RT5665_PDM1_L_SFT, rt5665_pdm_src); static const struct snd_kcontrol_new rt5665_pdm_l_mux = SOC_DAPM_ENUM("PDM L Source", rt5665_pdm_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_pdm_r_enum, RT5665_PDM_OUT_CTRL, RT5665_PDM1_R_SFT, rt5665_pdm_src); @@ -2314,7 +2314,7 @@ static const char * const rt5665_if1_1_adc1_data_src[] = { "STO1 ADC", "IF2_1 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_1_adc1_data_enum, RT5665_TDM_CTRL_3, RT5665_IF1_ADC1_SEL_SFT, rt5665_if1_1_adc1_data_src); @@ -2326,7 +2326,7 @@ static const char * const rt5665_if1_1_adc2_data_src[] = { "STO2 ADC", "IF2_2 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_1_adc2_data_enum, RT5665_TDM_CTRL_3, RT5665_IF1_ADC2_SEL_SFT, rt5665_if1_1_adc2_data_src); @@ -2338,7 +2338,7 @@ static const char * const rt5665_if1_1_adc3_data_src[] = { "MONO ADC", "IF3 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_1_adc3_data_enum, RT5665_TDM_CTRL_3, RT5665_IF1_ADC3_SEL_SFT, rt5665_if1_1_adc3_data_src); @@ -2350,7 +2350,7 @@ static const char * const rt5665_if1_2_adc1_data_src[] = { "STO1 ADC", "IF1 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc1_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC1_SEL_SFT, rt5665_if1_2_adc1_data_src); @@ -2362,7 +2362,7 @@ static const char * const rt5665_if1_2_adc2_data_src[] = { "STO2 ADC", "IF2_1 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc2_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC2_SEL_SFT, rt5665_if1_2_adc2_data_src); @@ -2374,7 +2374,7 @@ static const char * const rt5665_if1_2_adc3_data_src[] = { "MONO ADC", "IF2_2 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc3_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC3_SEL_SFT, rt5665_if1_2_adc3_data_src); @@ -2386,7 +2386,7 @@ static const char * const rt5665_if1_2_adc4_data_src[] = { "DAC1", "IF3 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc4_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC4_SEL_SFT, rt5665_if1_2_adc4_data_src); @@ -2401,14 +2401,14 @@ static const char * const rt5665_tdm_adc_data_src[] = { "4123", "4132", "4213", "4231", "4312", "4321" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_tdm1_adc_data_enum, RT5665_TDM_CTRL_3, RT5665_TDM_ADC_SEL_SFT, rt5665_tdm_adc_data_src); static const struct snd_kcontrol_new rt5665_tdm1_adc_mux = SOC_DAPM_ENUM("TDM1 ADC Mux", rt5665_tdm1_adc_data_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_tdm2_adc_data_enum, RT5665_TDM_CTRL_4, RT5665_TDM_ADC_SEL_SFT, rt5665_tdm_adc_data_src); -- cgit v1.2.3 From 9f3b777f1de9ff5d17f7259b8f7da5e9d4303e87 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 2 May 2017 22:33:01 +0900 Subject: ASoC: codecs: msm8916: fix invalid cast to bool type A function snd_soc_update_bits() is an application of regmap_update_bits_base(). This function takes some arguments for bitmask and new value, thus the arguments should be a type which has width. However bool is used to variable for the argument. This brings truncation and results in invalid operation. This commit fixes this bug by using unsigned int type, instead of bool. This bug is detected by sparse: smsm8916-wcd-analog.c:809:43: warning: odd constant _Bool cast (40 becomes 1) smsm8916-wcd-analog.c:814:43: warning: odd constant _Bool cast (40 becomes 1) Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Takashi Sakamoto Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-analog.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index d8e8590746af..a78802920c3c 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -223,8 +223,8 @@ struct pm8916_wcd_analog_priv { u16 codec_version; struct clk *mclk; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; - bool micbias1_cap_mode; - bool micbias2_cap_mode; + unsigned int micbias1_cap_mode; + unsigned int micbias2_cap_mode; }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; @@ -285,7 +285,7 @@ static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec) static int pm8916_wcd_analog_enable_micbias_ext(struct snd_soc_codec *codec, int event, - int reg, u32 cap_mode) + int reg, unsigned int cap_mode) { switch (event) { case SND_SOC_DAPM_POST_PMU: -- cgit v1.2.3 From 2a54e845f6e5069666e1749bd952abdc0413910d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 2 May 2017 22:33:02 +0900 Subject: ASoC: hisilicon: localize functions without external linkage A driver for hi6210 sound interface on hi6220 boards includes some functions which has no external linkage. These functions should have static qualifier. This commit adds the qualifier to localize the functions. This issue is detected by sparse: hi6210-i2s.c:100:5: warning: symbol 'hi6210_i2s_startup' was not declared. Should it be static? hi6210-i2s.c:178:6: warning: symbol 'hi6210_i2s_shutdown' was not declared. Should it be static? hi6210-i2s.c:527:27: warning: symbol 'hi6210_i2s_dai_init' was not declared. Should it be static? Signed-off-by: Takashi Sakamoto Signed-off-by: Mark Brown --- sound/soc/hisilicon/hi6210-i2s.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 45163e5202f5..b193d3beb253 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -97,8 +97,8 @@ static inline u32 hi6210_read_reg(struct hi6210_i2s *i2s, int reg) return readl(i2s->base + reg); } -int hi6210_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int hi6210_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); int ret, n; @@ -175,8 +175,9 @@ int hi6210_i2s_startup(struct snd_pcm_substream *substream, return 0; } -void hi6210_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) + +static void hi6210_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); int n; @@ -524,7 +525,7 @@ static struct snd_soc_dai_ops hi6210_i2s_dai_ops = { .shutdown = hi6210_i2s_shutdown, }; -struct snd_soc_dai_driver hi6210_i2s_dai_init = { +static const struct snd_soc_dai_driver hi6210_i2s_dai_init = { .probe = hi6210_i2s_dai_probe, .playback = { .channels_min = 2, -- cgit v1.2.3 From 51827c41c9ce07293b094691673e6ec23dfdc5e8 Mon Sep 17 00:00:00 2001 From: Tomas Vilda Date: Sat, 13 May 2017 00:29:37 +0300 Subject: ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec Signed-off-by: Tomas Vilda Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8a90ba8cd71..d7d03c92cb8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { static struct snd_soc_dai_driver dac31xx_dai_driver[] = { { - .name = "tlv32dac31xx-hifi", + .name = "tlv320dac31xx-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, -- cgit v1.2.3 From fa1014302791a1e436387e93a90f38717d7f9b03 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Tue, 2 May 2017 09:42:58 +0800 Subject: ASoC: nau8824: TDM support Support TDM format for NAU88L24. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 12 +++++++++++ 2 files changed, 64 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cca974d26136..3a309b18035e 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1124,6 +1124,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +/** + * nau8824_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: Bitmask representing active TX slots. Ex. + * 0xf for normal 4 channel TDM. + * 0xf0 for shifted 4 channel TDM + * @rx_mask: Bitmask [0:1] representing active DACR RX slots. + * Bitmask [2:3] representing active DACL RX slots. + * 00=CH0,01=CH1,10=CH2,11=CH3. Ex. + * 0xf for DACL/R selecting TDM CH3. + * 0xf0 for DACL/R selecting shifted TDM CH3. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Only support 4 slots TDM. + */ +static int nau8824_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int tslot_l = 0, ctrl_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf0) && (rx_mask & 0xf)) || + ((rx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf) && (tx_mask & 0xf0))) + return -EINVAL; + + ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + tslot_l = 4 * slot_width; + ctrl_val |= (tx_mask >> 4); + } else { + ctrl_val |= tx_mask; + } + if (rx_mask & 0xf0) + ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT); + else + ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT); + + regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL, + NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN | + NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK | + NAU8824_TDM_TX_MASK, ctrl_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT, + NAU8824_TSLOT_L_MASK, tslot_l); + + return 0; +} + /** * nau8824_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. @@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = { static const struct snd_soc_dai_ops nau8824_dai_ops = { .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, + .set_tdm_slot = nau8824_set_tdm_slot, }; #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 87ac9a382aed..21eae2431c83 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -258,6 +258,18 @@ #define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) #define NAU8824_I2S_BLK_DIV_MASK 0x7 +/* PORT0_LEFT_TIME_SLOT (0x1E) */ +#define NAU8824_TSLOT_L_MASK 0x3ff + +/* TDM_CTRL (0x20) */ +#define NAU8824_TDM_MODE (0x1 << 15) +#define NAU8824_TDM_OFFSET_EN (0x1 << 14) +#define NAU8824_TDM_DACL_RX_SFT 6 +#define NAU8824_TDM_DACL_RX_MASK (0x3 << NAU8824_TDM_DACL_RX_SFT) +#define NAU8824_TDM_DACR_RX_SFT 4 +#define NAU8824_TDM_DACR_RX_MASK (0x3 << NAU8824_TDM_DACR_RX_SFT) +#define NAU8824_TDM_TX_MASK 0xf + /* ADC_FILTER_CTRL (0x24) */ #define NAU8824_ADC_SYNC_DOWN_MASK 0x3 #define NAU8824_ADC_SYNC_DOWN_32 0 -- cgit v1.2.3 From d60bc8d6c6d7e5f9765852b0be57de639ba65808 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 2 May 2017 10:42:56 +0800 Subject: ASoC: rt5514: Add more width and channels support in the TDM mode This patch adds more width and channels support in the TDM mode. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 23 +++++++++++++++++++++-- sound/soc/codecs/rt5514.h | 6 ++++++ 2 files changed, 27 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b1ddf0..ff97360c03db 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -906,9 +906,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; - if (slots == 4) + switch (slots) { + case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + break; + + case 6: + val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH; + break; + + case 8: + val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH; + break; + case 2: + default: + break; + } switch (slot_width) { case 20: @@ -919,6 +933,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; break; + case 25: + val |= RT5514_TDM_MODE2; + break; + case 32: val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; break; @@ -930,7 +948,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | - RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | + RT5514_TDM_MODE2, val); return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 5d343fb6d125..02bc212a86d9 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -117,6 +117,8 @@ #define RT5514_POW_ADCFEDL_BIT 0 /* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE2 (0x1 << 30) +#define RT5514_TDM_MODE2_SFT 30 #define RT5514_TDM_MODE (0x1 << 28) #define RT5514_TDM_MODE_SFT 28 #define RT5514_I2S_LR_MASK (0x1 << 26) @@ -136,6 +138,8 @@ #define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) #define RT5514_TDMSLOT_SEL_RX_SFT 10 #define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_TDMSLOT_SEL_RX_6CH (0x2 << 10) +#define RT5514_TDMSLOT_SEL_RX_8CH (0x3 << 10) #define RT5514_CH_LEN_RX_MASK (0x3 << 8) #define RT5514_CH_LEN_RX_SFT 8 #define RT5514_CH_LEN_RX_16 (0x0 << 8) @@ -145,6 +149,8 @@ #define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) #define RT5514_TDMSLOT_SEL_TX_SFT 6 #define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_TDMSLOT_SEL_TX_6CH (0x2 << 6) +#define RT5514_TDMSLOT_SEL_TX_8CH (0x3 << 6) #define RT5514_CH_LEN_TX_MASK (0x3 << 4) #define RT5514_CH_LEN_TX_SFT 4 #define RT5514_CH_LEN_TX_16 (0x0 << 4) -- cgit v1.2.3 From 30b7d88de034a2b5c4f20c0cd05c792d9b619d70 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 2 May 2017 11:00:39 +0800 Subject: ASoC: rt5665: add ADC STO2 ASRC support "ADC Stereo2 Filter" is with ASRC supported. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 14b0cf89edf5..26bf157ca293 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2684,6 +2684,8 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { RT5665_DAC_MONO_R_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5665_ASRC_1, RT5665_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5665_ASRC_1, + RT5665_ADC_STO2_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono L ASRC", 1, RT5665_ASRC_1, RT5665_ADC_MONO_L_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono R ASRC", 1, RT5665_ASRC_1, @@ -3227,6 +3229,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc}, {"ADC Mono Left Filter", NULL, "ADC Mono L ASRC", is_using_asrc}, {"ADC Mono Right Filter", NULL, "ADC Mono R ASRC", is_using_asrc}, {"DAC Mono Left Filter", NULL, "DAC Mono L ASRC", is_using_asrc}, -- cgit v1.2.3 From b98ae9ad559fea64dee5fcc8e3ba4bf936ceb5e6 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 15 May 2017 10:33:59 +0200 Subject: ASoC: rt5665: Fix uninitialized warning in rt5665_i2s_pin_event() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With gcc 4.1.2: sound/soc/codecs/rt5665.c: In function ‘rt5665_i2s_pin_event’: sound/soc/codecs/rt5665.c:2610: warning: ‘mask1’ may be used uninitialized in this function sound/soc/codecs/rt5665.c:2610: warning: ‘val2’ may be used uninitialized in this function sound/soc/codecs/rt5665.c:2610: warning: ‘val1’ may be used uninitialized in this function The first one is currently a false positive, as rt5665_i2s_pin_event() is never called with snd_soc_dapm_widget.shift set to a value not handled by the switch() statement. But that may change, so preinitialize mask1 to fix this, like is already done for mask2. The last two are false-positives, the compiler is just not smart enough to notice the mask and val variables are always used together. Fixes: 9b5d3865b3b410d2 ("ASoC: rt5665: set i2s pin share configuration") Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 26bf157ca293..c0f36d85ee4d 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2607,7 +2607,7 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int val1, val2, mask1, mask2 = 0; + unsigned int val1, val2, mask1 = 0, mask2 = 0; switch (w->shift) { case RT5665_PWR_I2S2_1_BIT: @@ -2635,13 +2635,17 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, } switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, val1); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, val1); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, val2); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, 0); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, 0); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, 0); -- cgit v1.2.3 From 0c343a35bfecdf26c7041781815f3b639a45d93a Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sun, 14 May 2017 17:53:21 +0100 Subject: ASoC: hdmi-codec: fix spelling mistake: "deteced" -> "detected" Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..a3f15149afcf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -819,7 +819,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) mutex_unlock(&hdmi_mutex); if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) { - dev_err(dev, "too many hdmi codec are deteced\n"); + dev_err(dev, "too many hdmi codec are detected\n"); return -EINVAL; } -- cgit v1.2.3 From 18fe7869764c0b86e8ce6539bbb6e528f1d9928f Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 16 May 2017 16:24:56 +0000 Subject: ASoC: rt5665: make local symbol rt5665_i2c_driver static Fixes the following sparse warnings: sound/soc/codecs/rt5665.c:4928:19: warning: symbol 'rt5665_i2c_driver' was not declared. Should it be static? Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index c0f36d85ee4d..7420010fd8e9 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4929,7 +4929,7 @@ static struct acpi_device_id rt5665_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5665_acpi_match); #endif -struct i2c_driver rt5665_i2c_driver = { +static struct i2c_driver rt5665_i2c_driver = { .driver = { .name = "rt5665", .of_match_table = of_match_ptr(rt5665_of_match), -- cgit v1.2.3 From 3048e76c93bccf875a49025870de08aed86c4692 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Tue, 16 May 2017 19:20:00 +0200 Subject: ASoC: fsi: Move inline fsi_stream_is_play() before use MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With gcc 4.1.2: sound/soc/sh/fsi.c:304: warning: ‘fsi_stream_is_play’ declared inline after being called sound/soc/sh/fsi.c:304: warning: previous declaration of ‘fsi_stream_is_play’ was here Move fsi_stream_is_play() up to fix this, removing the need for a forward declaration as well. Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ead520182e26..7c4bdd82bb95 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -301,7 +301,12 @@ struct fsi_master { spinlock_t lock; }; -static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io); +static inline int fsi_stream_is_play(struct fsi_priv *fsi, + struct fsi_stream *io) +{ + return &fsi->playback == io; +} + /* * basic read write function @@ -489,12 +494,6 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi) /* * fsi_stream_xx() function */ -static inline int fsi_stream_is_play(struct fsi_priv *fsi, - struct fsi_stream *io) -{ - return &fsi->playback == io; -} - static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, struct snd_pcm_substream *substream) { -- cgit v1.2.3 From 6d3edf866ffa7a9348cfc30d9f58270e4f8d068e Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 15 May 2017 19:02:07 +0800 Subject: ASoC: rt5514: Add ACPI match ID This patch adds the ACPI match ID for rt5514 codec. Signed-off-by: Hsin-Yu Chao Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index ff97360c03db..5c30c4d64ebe 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include #include #include #include @@ -1095,6 +1096,14 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5514_acpi_match[] = { + { "10EC5514", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match); +#endif + static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", @@ -1198,6 +1207,7 @@ static const struct dev_pm_ops rt5514_i2_pm_ops = { static struct i2c_driver rt5514_i2c_driver = { .driver = { .name = "rt5514", + .acpi_match_table = ACPI_PTR(rt5514_acpi_match), .of_match_table = of_match_ptr(rt5514_of_match), .pm = &rt5514_i2_pm_ops, }, -- cgit v1.2.3 From e4e6ec7b127c97fbfa92161d0fc69f526f603e97 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Thu, 18 May 2017 17:19:52 +0200 Subject: ASoC: stm32: Add I2S driver Add I2S ASoC driver for STM32. This version of the driver supports only exclusive playback and capture interface. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/Kconfig | 2 +- sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_i2s.c | 941 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 946 insertions(+), 1 deletion(-) create mode 100644 sound/soc/stm/stm32_i2s.c (limited to 'sound') diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 972970f0890a..a6372de54042 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -5,4 +5,4 @@ menuconfig SND_SOC_STM32 select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help - Say Y if you want to enable ASoC-support for STM32 + Say Y if you want to enable ASoC support for STM32 diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile index e466a4759698..82519313c0b4 100644 --- a/sound/soc/stm/Makefile +++ b/sound/soc/stm/Makefile @@ -4,3 +4,7 @@ obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai-sub.o snd-soc-stm32-sai-objs := stm32_sai.o obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai.o + +# I2S +snd-soc-stm32-i2s-objs := stm32_i2s.o +obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-i2s.o diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c new file mode 100644 index 000000000000..22152a1ca733 --- /dev/null +++ b/sound/soc/stm/stm32_i2s.c @@ -0,0 +1,941 @@ +/* + * STM32 ALSA SoC Digital Audio Interface (I2S) driver. + * + * Copyright (C) 2017, STMicroelectronics - All Rights Reserved + * Author(s): Olivier Moysan for STMicroelectronics. + * + * License terms: GPL V2.0. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more + * details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#define STM32_I2S_CR1_REG 0x0 +#define STM32_I2S_CFG1_REG 0x08 +#define STM32_I2S_CFG2_REG 0x0C +#define STM32_I2S_IER_REG 0x10 +#define STM32_I2S_SR_REG 0x14 +#define STM32_I2S_IFCR_REG 0x18 +#define STM32_I2S_TXDR_REG 0X20 +#define STM32_I2S_RXDR_REG 0x30 +#define STM32_I2S_CGFR_REG 0X50 + +/* Bit definition for SPI2S_CR1 register */ +#define I2S_CR1_SPE BIT(0) +#define I2S_CR1_CSTART BIT(9) +#define I2S_CR1_CSUSP BIT(10) +#define I2S_CR1_HDDIR BIT(11) +#define I2S_CR1_SSI BIT(12) +#define I2S_CR1_CRC33_17 BIT(13) +#define I2S_CR1_RCRCI BIT(14) +#define I2S_CR1_TCRCI BIT(15) + +/* Bit definition for SPI_CFG2 register */ +#define I2S_CFG2_IOSWP_SHIFT 15 +#define I2S_CFG2_IOSWP BIT(I2S_CFG2_IOSWP_SHIFT) +#define I2S_CFG2_LSBFRST BIT(23) +#define I2S_CFG2_AFCNTR BIT(31) + +/* Bit definition for SPI_CFG1 register */ +#define I2S_CFG1_FTHVL_SHIFT 5 +#define I2S_CFG1_FTHVL_MASK GENMASK(8, I2S_CFG1_FTHVL_SHIFT) +#define I2S_CFG1_FTHVL_SET(x) ((x) << I2S_CFG1_FTHVL_SHIFT) + +#define I2S_CFG1_TXDMAEN BIT(15) +#define I2S_CFG1_RXDMAEN BIT(14) + +/* Bit definition for SPI2S_IER register */ +#define I2S_IER_RXPIE BIT(0) +#define I2S_IER_TXPIE BIT(1) +#define I2S_IER_DPXPIE BIT(2) +#define I2S_IER_EOTIE BIT(3) +#define I2S_IER_TXTFIE BIT(4) +#define I2S_IER_UDRIE BIT(5) +#define I2S_IER_OVRIE BIT(6) +#define I2S_IER_CRCEIE BIT(7) +#define I2S_IER_TIFREIE BIT(8) +#define I2S_IER_MODFIE BIT(9) +#define I2S_IER_TSERFIE BIT(10) + +/* Bit definition for SPI2S_SR register */ +#define I2S_SR_RXP BIT(0) +#define I2S_SR_TXP BIT(1) +#define I2S_SR_DPXP BIT(2) +#define I2S_SR_EOT BIT(3) +#define I2S_SR_TXTF BIT(4) +#define I2S_SR_UDR BIT(5) +#define I2S_SR_OVR BIT(6) +#define I2S_SR_CRCERR BIT(7) +#define I2S_SR_TIFRE BIT(8) +#define I2S_SR_MODF BIT(9) +#define I2S_SR_TSERF BIT(10) +#define I2S_SR_SUSP BIT(11) +#define I2S_SR_TXC BIT(12) +#define I2S_SR_RXPLVL GENMASK(14, 13) +#define I2S_SR_RXWNE BIT(15) + +#define I2S_SR_MASK GENMASK(15, 0) + +/* Bit definition for SPI_IFCR register */ +#define I2S_IFCR_EOTC BIT(3) +#define I2S_IFCR_TXTFC BIT(4) +#define I2S_IFCR_UDRC BIT(5) +#define I2S_IFCR_OVRC BIT(6) +#define I2S_IFCR_CRCEC BIT(7) +#define I2S_IFCR_TIFREC BIT(8) +#define I2S_IFCR_MODFC BIT(9) +#define I2S_IFCR_TSERFC BIT(10) +#define I2S_IFCR_SUSPC BIT(11) + +#define I2S_IFCR_MASK GENMASK(11, 3) + +/* Bit definition for SPI_I2SCGFR register */ +#define I2S_CGFR_I2SMOD BIT(0) + +#define I2S_CGFR_I2SCFG_SHIFT 1 +#define I2S_CGFR_I2SCFG_MASK GENMASK(3, I2S_CGFR_I2SCFG_SHIFT) +#define I2S_CGFR_I2SCFG_SET(x) ((x) << I2S_CGFR_I2SCFG_SHIFT) + +#define I2S_CGFR_I2SSTD_SHIFT 4 +#define I2S_CGFR_I2SSTD_MASK GENMASK(5, I2S_CGFR_I2SSTD_SHIFT) +#define I2S_CGFR_I2SSTD_SET(x) ((x) << I2S_CGFR_I2SSTD_SHIFT) + +#define I2S_CGFR_PCMSYNC BIT(7) + +#define I2S_CGFR_DATLEN_SHIFT 8 +#define I2S_CGFR_DATLEN_MASK GENMASK(9, I2S_CGFR_DATLEN_SHIFT) +#define I2S_CGFR_DATLEN_SET(x) ((x) << I2S_CGFR_DATLEN_SHIFT) + +#define I2S_CGFR_CHLEN_SHIFT 10 +#define I2S_CGFR_CHLEN BIT(I2S_CGFR_CHLEN_SHIFT) +#define I2S_CGFR_CKPOL BIT(11) +#define I2S_CGFR_FIXCH BIT(12) +#define I2S_CGFR_WSINV BIT(13) +#define I2S_CGFR_DATFMT BIT(14) + +#define I2S_CGFR_I2SDIV_SHIFT 16 +#define I2S_CGFR_I2SDIV_BIT_H 23 +#define I2S_CGFR_I2SDIV_MASK GENMASK(I2S_CGFR_I2SDIV_BIT_H,\ + I2S_CGFR_I2SDIV_SHIFT) +#define I2S_CGFR_I2SDIV_SET(x) ((x) << I2S_CGFR_I2SDIV_SHIFT) +#define I2S_CGFR_I2SDIV_MAX ((1 << (I2S_CGFR_I2SDIV_BIT_H -\ + I2S_CGFR_I2SDIV_SHIFT)) - 1) + +#define I2S_CGFR_ODD_SHIFT 24 +#define I2S_CGFR_ODD BIT(I2S_CGFR_ODD_SHIFT) +#define I2S_CGFR_MCKOE BIT(25) + +enum i2s_master_mode { + I2S_MS_NOT_SET, + I2S_MS_MASTER, + I2S_MS_SLAVE, +}; + +enum i2s_mode { + I2S_I2SMOD_TX_SLAVE, + I2S_I2SMOD_RX_SLAVE, + I2S_I2SMOD_TX_MASTER, + I2S_I2SMOD_RX_MASTER, + I2S_I2SMOD_FD_SLAVE, + I2S_I2SMOD_FD_MASTER, +}; + +enum i2s_fifo_th { + I2S_FIFO_TH_NONE, + I2S_FIFO_TH_ONE_QUARTER, + I2S_FIFO_TH_HALF, + I2S_FIFO_TH_THREE_QUARTER, + I2S_FIFO_TH_FULL, +}; + +enum i2s_std { + I2S_STD_I2S, + I2S_STD_LEFT_J, + I2S_STD_RIGHT_J, + I2S_STD_DSP, +}; + +enum i2s_datlen { + I2S_I2SMOD_DATLEN_16, + I2S_I2SMOD_DATLEN_24, + I2S_I2SMOD_DATLEN_32, +}; + +#define STM32_I2S_DAI_NAME_SIZE 20 +#define STM32_I2S_FIFO_SIZE 16 + +#define STM32_I2S_IS_MASTER(x) ((x)->ms_flg == I2S_MS_MASTER) +#define STM32_I2S_IS_SLAVE(x) ((x)->ms_flg == I2S_MS_SLAVE) + +/** + * @regmap_conf: I2S register map configuration pointer + * @egmap: I2S register map pointer + * @pdev: device data pointer + * @dai_drv: DAI driver pointer + * @dma_data_tx: dma configuration data for tx channel + * @dma_data_rx: dma configuration data for tx channel + * @substream: PCM substream data pointer + * @i2sclk: kernel clock feeding the I2S clock generator + * @pclk: peripheral clock driving bus interface + * @x8kclk: I2S parent clock for sampling frequencies multiple of 8kHz + * @x11kclk: I2S parent clock for sampling frequencies multiple of 11kHz + * @base: mmio register base virtual address + * @phys_addr: I2S registers physical base address + * @lock_fd: lock to manage race conditions in full duplex mode + * @dais_name: DAI name + * @mclk_rate: master clock frequency (Hz) + * @fmt: DAI protocol + * @refcount: keep count of opened streams on I2S + * @ms_flg: master mode flag. + */ +struct stm32_i2s_data { + const struct regmap_config *regmap_conf; + struct regmap *regmap; + struct platform_device *pdev; + struct snd_soc_dai_driver *dai_drv; + struct snd_dmaengine_dai_dma_data dma_data_tx; + struct snd_dmaengine_dai_dma_data dma_data_rx; + struct snd_pcm_substream *substream; + struct clk *i2sclk; + struct clk *pclk; + struct clk *x8kclk; + struct clk *x11kclk; + void __iomem *base; + dma_addr_t phys_addr; + spinlock_t lock_fd; /* Manage race conditions for full duplex */ + char dais_name[STM32_I2S_DAI_NAME_SIZE]; + unsigned int mclk_rate; + unsigned int fmt; + int refcount; + int ms_flg; +}; + +static irqreturn_t stm32_i2s_isr(int irq, void *devid) +{ + struct stm32_i2s_data *i2s = (struct stm32_i2s_data *)devid; + struct platform_device *pdev = i2s->pdev; + u32 sr, ier; + unsigned long flags; + int err = 0; + + regmap_read(i2s->regmap, STM32_I2S_SR_REG, &sr); + regmap_read(i2s->regmap, STM32_I2S_IER_REG, &ier); + + flags = sr & ier; + if (!flags) { + dev_dbg(&pdev->dev, "Spurious IRQ sr=0x%08x, ier=0x%08x\n", + sr, ier); + return IRQ_NONE; + } + + regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, flags); + + if (flags & I2S_SR_OVR) { + dev_dbg(&pdev->dev, "Overrun\n"); + err = 1; + } + + if (flags & I2S_SR_UDR) { + dev_dbg(&pdev->dev, "Underrun\n"); + err = 1; + } + + if (flags & I2S_SR_TIFRE) + dev_dbg(&pdev->dev, "Frame error\n"); + + if (err) + snd_pcm_stop_xrun(i2s->substream); + + return IRQ_HANDLED; +} + +static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case STM32_I2S_CR1_REG: + case STM32_I2S_CFG1_REG: + case STM32_I2S_CFG2_REG: + case STM32_I2S_IER_REG: + case STM32_I2S_SR_REG: + case STM32_I2S_IFCR_REG: + case STM32_I2S_TXDR_REG: + case STM32_I2S_RXDR_REG: + case STM32_I2S_CGFR_REG: + return true; + default: + return false; + } +} + +static bool stm32_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case STM32_I2S_TXDR_REG: + case STM32_I2S_RXDR_REG: + return true; + default: + return false; + } +} + +static bool stm32_i2s_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case STM32_I2S_CR1_REG: + case STM32_I2S_CFG1_REG: + case STM32_I2S_CFG2_REG: + case STM32_I2S_IER_REG: + case STM32_I2S_IFCR_REG: + case STM32_I2S_TXDR_REG: + case STM32_I2S_CGFR_REG: + return true; + default: + return false; + } +} + +static int stm32_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + u32 cgfr; + u32 cgfr_mask = I2S_CGFR_I2SSTD_MASK | I2S_CGFR_CKPOL | + I2S_CGFR_WSINV | I2S_CGFR_I2SCFG_MASK; + + dev_dbg(cpu_dai->dev, "fmt %x\n", fmt); + + /* + * winv = 0 : default behavior (high/low) for all standards + * ckpol = 0 for all standards. + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cgfr = I2S_CGFR_I2SSTD_SET(I2S_STD_I2S); + break; + case SND_SOC_DAIFMT_MSB: + cgfr = I2S_CGFR_I2SSTD_SET(I2S_STD_LEFT_J); + break; + case SND_SOC_DAIFMT_LSB: + cgfr = I2S_CGFR_I2SSTD_SET(I2S_STD_RIGHT_J); + break; + case SND_SOC_DAIFMT_DSP_A: + cgfr = I2S_CGFR_I2SSTD_SET(I2S_STD_DSP); + break; + /* DSP_B not mapped on I2S PCM long format. 1 bit offset does not fit */ + default: + dev_err(cpu_dai->dev, "Unsupported protocol %#x\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* DAI clock strobing */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + cgfr |= I2S_CGFR_CKPOL; + break; + case SND_SOC_DAIFMT_NB_IF: + cgfr |= I2S_CGFR_WSINV; + break; + case SND_SOC_DAIFMT_IB_IF: + cgfr |= I2S_CGFR_CKPOL; + cgfr |= I2S_CGFR_WSINV; + break; + default: + dev_err(cpu_dai->dev, "Unsupported strobing %#x\n", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->ms_flg = I2S_MS_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->ms_flg = I2S_MS_MASTER; + break; + default: + dev_err(cpu_dai->dev, "Unsupported mode %#x\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + i2s->fmt = fmt; + return regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, + cgfr_mask, cgfr); +} + +static int stm32_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + + dev_dbg(cpu_dai->dev, "I2S MCLK frequency is %uHz\n", freq); + + if ((dir == SND_SOC_CLOCK_OUT) && STM32_I2S_IS_MASTER(i2s)) { + i2s->mclk_rate = freq; + + /* Enable master clock if master mode and mclk-fs are set */ + return regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, + I2S_CGFR_MCKOE, I2S_CGFR_MCKOE); + } + + return 0; +} + +static int stm32_i2s_configure_clock(struct snd_soc_dai *cpu_dai, + struct snd_pcm_hw_params *params) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long i2s_clock_rate; + unsigned int tmp, div, real_div, nb_bits, frame_len; + unsigned int rate = params_rate(params); + int ret; + u32 cgfr, cgfr_mask; + bool odd; + + if (!(rate % 11025)) + clk_set_parent(i2s->i2sclk, i2s->x11kclk); + else + clk_set_parent(i2s->i2sclk, i2s->x8kclk); + i2s_clock_rate = clk_get_rate(i2s->i2sclk); + + /* + * mckl = mclk_ratio x ws + * i2s mode : mclk_ratio = 256 + * dsp mode : mclk_ratio = 128 + * + * mclk on + * i2s mode : div = i2s_clk / (mclk_ratio * ws) + * dsp mode : div = i2s_clk / (mclk_ratio * ws) + * mclk off + * i2s mode : div = i2s_clk / (nb_bits x ws) + * dsp mode : div = i2s_clk / (nb_bits x ws) + */ + if (i2s->mclk_rate) { + tmp = DIV_ROUND_CLOSEST(i2s_clock_rate, i2s->mclk_rate); + } else { + frame_len = 32; + if ((i2s->fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_DSP_A) + frame_len = 16; + + /* master clock not enabled */ + ret = regmap_read(i2s->regmap, STM32_I2S_CGFR_REG, &cgfr); + if (ret < 0) + return ret; + + nb_bits = frame_len * ((cgfr & I2S_CGFR_CHLEN) + 1); + tmp = DIV_ROUND_CLOSEST(i2s_clock_rate, (nb_bits * rate)); + } + + /* Check the parity of the divider */ + odd = tmp & 0x1; + + /* Compute the div prescaler */ + div = tmp >> 1; + + cgfr = I2S_CGFR_I2SDIV_SET(div) | (odd << I2S_CGFR_ODD_SHIFT); + cgfr_mask = I2S_CGFR_I2SDIV_MASK | I2S_CGFR_ODD; + + real_div = ((2 * div) + odd); + dev_dbg(cpu_dai->dev, "I2S clk: %ld, SCLK: %d\n", + i2s_clock_rate, rate); + dev_dbg(cpu_dai->dev, "Divider: 2*%d(div)+%d(odd) = %d\n", + div, odd, real_div); + + if (((div == 1) && odd) || (div > I2S_CGFR_I2SDIV_MAX)) { + dev_err(cpu_dai->dev, "Wrong divider setting\n"); + return -EINVAL; + } + + if (!div && !odd) + dev_warn(cpu_dai->dev, "real divider forced to 1\n"); + + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, + cgfr_mask, cgfr); + if (ret < 0) + return ret; + + /* Set bitclock and frameclock to their inactive state */ + return regmap_update_bits(i2s->regmap, STM32_I2S_CFG2_REG, + I2S_CFG2_AFCNTR, I2S_CFG2_AFCNTR); +} + +static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, + struct snd_pcm_hw_params *params, + struct snd_pcm_substream *substream) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + int format = params_width(params); + u32 cfgr, cfgr_mask, cfg1, cfg1_mask; + bool playback_flg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + unsigned int fthlv; + int ret; + + if ((params_channels(params) == 1) && + ((i2s->fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_DSP_A)) { + dev_err(cpu_dai->dev, "Mono mode supported only by DSP_A\n"); + return -EINVAL; + } + + switch (format) { + case 16: + cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_16); + cfgr_mask = I2S_CGFR_DATLEN_MASK; + break; + case 32: + cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_32) | + I2S_CGFR_CHLEN; + cfgr_mask = I2S_CGFR_DATLEN_MASK | I2S_CGFR_CHLEN; + break; + default: + dev_err(cpu_dai->dev, "Unexpected format %d", format); + return -EINVAL; + } + + if (STM32_I2S_IS_SLAVE(i2s)) { + if (playback_flg) + cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_TX_SLAVE); + else + cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_RX_SLAVE); + + /* As data length is either 16 or 32 bits, fixch always set */ + cfgr |= I2S_CGFR_FIXCH; + cfgr_mask |= I2S_CGFR_FIXCH; + } else { + if (playback_flg) + cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_TX_MASTER); + else + cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_RX_MASTER); + } + cfgr_mask |= I2S_CGFR_I2SCFG_MASK; + + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, + cfgr_mask, cfgr); + if (ret < 0) + return ret; + + cfg1 = I2S_CFG1_RXDMAEN; + if (playback_flg) + cfg1 = I2S_CFG1_TXDMAEN; + cfg1_mask = cfg1; + + fthlv = STM32_I2S_FIFO_SIZE * I2S_FIFO_TH_ONE_QUARTER / 4; + cfg1 |= I2S_CFG1_FTHVL_SET(fthlv - 1); + cfg1_mask |= I2S_CFG1_FTHVL_MASK; + + return regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, + cfg1_mask, cfg1); +} + +static int stm32_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + int ret, ier; + + i2s->substream = substream; + + spin_lock(&i2s->lock_fd); + if (i2s->refcount) { + dev_err(cpu_dai->dev, "%s stream already started\n", + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "Capture" : "Playback")); + spin_unlock(&i2s->lock_fd); + return -EBUSY; + } + i2s->refcount = 1; + spin_unlock(&i2s->lock_fd); + + ret = regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); + if (ret < 0) + return ret; + + /* Enable ITs */ + ier = I2S_IER_OVRIE | I2S_IER_UDRIE; + if (STM32_I2S_IS_SLAVE(i2s)) + ier |= I2S_IER_TIFREIE; + + return regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, ier, ier); +} + +static int stm32_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + int ret; + + ret = stm32_i2s_configure(cpu_dai, params, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, "Configuration returned error %d\n", ret); + return ret; + } + + if (STM32_I2S_IS_MASTER(i2s)) + ret = stm32_i2s_configure_clock(cpu_dai, params); + + return ret; +} + +static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + bool playback_flg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 cfg1_mask; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* Enable i2s */ + dev_dbg(cpu_dai->dev, "start I2S\n"); + + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, + I2S_CR1_SPE, I2S_CR1_SPE); + if (ret < 0) { + dev_err(cpu_dai->dev, "Error %d enabling I2S\n", ret); + return ret; + } + + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, + I2S_CR1_CSTART, I2S_CR1_CSTART); + if (ret < 0) { + dev_err(cpu_dai->dev, "Error %d starting I2S\n", ret); + return ret; + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(cpu_dai->dev, "stop I2S\n"); + + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, + I2S_CR1_SPE, 0); + if (ret < 0) { + dev_err(cpu_dai->dev, "Error %d disabling I2S\n", ret); + return ret; + } + + cfg1_mask = I2S_CFG1_RXDMAEN; + if (playback_flg) + cfg1_mask = I2S_CFG1_TXDMAEN; + + regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, + cfg1_mask, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static void stm32_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); + + i2s->substream = NULL; + + spin_lock(&i2s->lock_fd); + i2s->refcount = 0; + spin_unlock(&i2s->lock_fd); + + regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, + I2S_CGFR_MCKOE, (unsigned int)~I2S_CGFR_MCKOE); +} + +static int stm32_i2s_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct stm32_i2s_data *i2s = dev_get_drvdata(cpu_dai->dev); + struct snd_dmaengine_dai_dma_data *dma_data_tx = &i2s->dma_data_tx; + struct snd_dmaengine_dai_dma_data *dma_data_rx = &i2s->dma_data_rx; + + /* Buswidth will be set by framework */ + dma_data_tx->addr_width = DMA_SLAVE_BUSWIDTH_UNDEFINED; + dma_data_tx->addr = (dma_addr_t)(i2s->phys_addr) + STM32_I2S_TXDR_REG; + dma_data_tx->maxburst = 1; + dma_data_rx->addr_width = DMA_SLAVE_BUSWIDTH_UNDEFINED; + dma_data_rx->addr = (dma_addr_t)(i2s->phys_addr) + STM32_I2S_RXDR_REG; + dma_data_rx->maxburst = 1; + + snd_soc_dai_init_dma_data(cpu_dai, dma_data_tx, dma_data_rx); + + return 0; +} + +static const struct regmap_config stm32_h7_i2s_regmap_conf = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = STM32_I2S_CGFR_REG, + .readable_reg = stm32_i2s_readable_reg, + .volatile_reg = stm32_i2s_volatile_reg, + .writeable_reg = stm32_i2s_writeable_reg, + .fast_io = true, +}; + +static const struct snd_soc_dai_ops stm32_i2s_pcm_dai_ops = { + .set_sysclk = stm32_i2s_set_sysclk, + .set_fmt = stm32_i2s_set_dai_fmt, + .startup = stm32_i2s_startup, + .hw_params = stm32_i2s_hw_params, + .trigger = stm32_i2s_trigger, + .shutdown = stm32_i2s_shutdown, +}; + +static const struct snd_pcm_hardware stm32_i2s_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP, + .buffer_bytes_max = 8 * PAGE_SIZE, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 8, +}; + +static const struct snd_dmaengine_pcm_config stm32_i2s_pcm_config = { + .pcm_hardware = &stm32_i2s_pcm_hw, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .prealloc_buffer_size = PAGE_SIZE * 8, +}; + +static const struct snd_soc_component_driver stm32_i2s_component = { + .name = "stm32-i2s", +}; + +static void stm32_i2s_dai_init(struct snd_soc_pcm_stream *stream, + char *stream_name) +{ + stream->stream_name = stream_name; + stream->channels_min = 1; + stream->channels_max = 2; + stream->rates = SNDRV_PCM_RATE_8000_192000; + stream->formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; +} + +static int stm32_i2s_dais_init(struct platform_device *pdev, + struct stm32_i2s_data *i2s) +{ + struct snd_soc_dai_driver *dai_ptr; + + dai_ptr = devm_kzalloc(&pdev->dev, sizeof(struct snd_soc_dai_driver), + GFP_KERNEL); + if (!dai_ptr) + return -ENOMEM; + + snprintf(i2s->dais_name, STM32_I2S_DAI_NAME_SIZE, + "%s", dev_name(&pdev->dev)); + + dai_ptr->probe = stm32_i2s_dai_probe; + dai_ptr->ops = &stm32_i2s_pcm_dai_ops; + dai_ptr->name = i2s->dais_name; + dai_ptr->id = 1; + stm32_i2s_dai_init(&dai_ptr->playback, "playback"); + stm32_i2s_dai_init(&dai_ptr->capture, "capture"); + i2s->dai_drv = dai_ptr; + + return 0; +} + +static const struct of_device_id stm32_i2s_ids[] = { + { + .compatible = "st,stm32h7-i2s", + .data = &stm32_h7_i2s_regmap_conf + }, + {}, +}; + +static int stm32_i2s_parse_dt(struct platform_device *pdev, + struct stm32_i2s_data *i2s) +{ + struct device_node *np = pdev->dev.of_node; + const struct of_device_id *of_id; + struct reset_control *rst; + struct resource *res; + int irq, ret; + + if (!np) + return -ENODEV; + + of_id = of_match_device(stm32_i2s_ids, &pdev->dev); + if (of_id) + i2s->regmap_conf = (const struct regmap_config *)of_id->data; + else + return -EINVAL; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); + + i2s->phys_addr = res->start; + + /* Get clocks */ + i2s->pclk = devm_clk_get(&pdev->dev, "pclk"); + if (IS_ERR(i2s->pclk)) { + dev_err(&pdev->dev, "Could not get pclk\n"); + return PTR_ERR(i2s->pclk); + } + + i2s->i2sclk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(i2s->i2sclk)) { + dev_err(&pdev->dev, "Could not get i2sclk\n"); + return PTR_ERR(i2s->i2sclk); + } + + i2s->x8kclk = devm_clk_get(&pdev->dev, "x8k"); + if (IS_ERR(i2s->x8kclk)) { + dev_err(&pdev->dev, "missing x8k parent clock\n"); + return PTR_ERR(i2s->x8kclk); + } + + i2s->x11kclk = devm_clk_get(&pdev->dev, "x11k"); + if (IS_ERR(i2s->x11kclk)) { + dev_err(&pdev->dev, "missing x11k parent clock\n"); + return PTR_ERR(i2s->x11kclk); + } + + /* Get irqs */ + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + return -ENOENT; + } + + ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT, + dev_name(&pdev->dev), i2s); + if (ret) { + dev_err(&pdev->dev, "irq request returned %d\n", ret); + return ret; + } + + /* Reset */ + rst = devm_reset_control_get(&pdev->dev, NULL); + if (!IS_ERR(rst)) { + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); + } + + return 0; +} + +static int stm32_i2s_probe(struct platform_device *pdev) +{ + struct stm32_i2s_data *i2s; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + ret = stm32_i2s_parse_dt(pdev, i2s); + if (ret) + return ret; + + i2s->pdev = pdev; + i2s->ms_flg = I2S_MS_NOT_SET; + spin_lock_init(&i2s->lock_fd); + platform_set_drvdata(pdev, i2s); + + ret = stm32_i2s_dais_init(pdev, i2s); + if (ret) + return ret; + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->base, + i2s->regmap_conf); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(i2s->regmap); + } + + ret = clk_prepare_enable(i2s->pclk); + if (ret) { + dev_err(&pdev->dev, "Enable pclk failed: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(i2s->i2sclk); + if (ret) { + dev_err(&pdev->dev, "Enable i2sclk failed: %d\n", ret); + goto err_pclk_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &stm32_i2s_component, + i2s->dai_drv, 1); + if (ret) + goto err_clocks_disable; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, + &stm32_i2s_pcm_config, 0); + if (ret) + goto err_clocks_disable; + + /* Set SPI/I2S in i2s mode */ + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, + I2S_CGFR_I2SMOD, I2S_CGFR_I2SMOD); + if (ret) + goto err_clocks_disable; + + return ret; + +err_clocks_disable: + clk_disable_unprepare(i2s->i2sclk); +err_pclk_disable: + clk_disable_unprepare(i2s->pclk); + + return ret; +} + +static int stm32_i2s_remove(struct platform_device *pdev) +{ + struct stm32_i2s_data *i2s = platform_get_drvdata(pdev); + + clk_disable_unprepare(i2s->i2sclk); + clk_disable_unprepare(i2s->pclk); + + return 0; +} + +MODULE_DEVICE_TABLE(of, stm32_i2s_ids); + +static struct platform_driver stm32_i2s_driver = { + .driver = { + .name = "st,stm32-i2s", + .of_match_table = stm32_i2s_ids, + }, + .probe = stm32_i2s_probe, + .remove = stm32_i2s_remove, +}; + +module_platform_driver(stm32_i2s_driver); + +MODULE_DESCRIPTION("STM32 Soc i2s Interface"); +MODULE_AUTHOR("Olivier Moysan, "); +MODULE_ALIAS("platform:stm32-i2s"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From e7cc49b8adf25e7bae6acaeb37036ef8726b902c Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Thu, 18 May 2017 17:19:53 +0200 Subject: ASoC: stm32: Add full duplex support to i2s This patch allows to use i2s interface either as single audio path (rx or tx), or bidirectional audio path. This patch is added separately, as the driver does not follow recommended use of the interface, to support this configuration. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 87 +++++++++++++++++++++++++---------------------- 1 file changed, 46 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 22152a1ca733..8052629a89df 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -489,7 +489,6 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); int format = params_width(params); u32 cfgr, cfgr_mask, cfg1, cfg1_mask; - bool playback_flg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); unsigned int fthlv; int ret; @@ -515,19 +514,13 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, } if (STM32_I2S_IS_SLAVE(i2s)) { - if (playback_flg) - cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_TX_SLAVE); - else - cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_RX_SLAVE); + cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_FD_SLAVE); /* As data length is either 16 or 32 bits, fixch always set */ cfgr |= I2S_CGFR_FIXCH; cfgr_mask |= I2S_CGFR_FIXCH; } else { - if (playback_flg) - cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_TX_MASTER); - else - cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_RX_MASTER); + cfgr |= I2S_CGFR_I2SCFG_SET(I2S_I2SMOD_FD_MASTER); } cfgr_mask |= I2S_CGFR_I2SCFG_MASK; @@ -536,9 +529,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, if (ret < 0) return ret; - cfg1 = I2S_CFG1_RXDMAEN; - if (playback_flg) - cfg1 = I2S_CFG1_TXDMAEN; + cfg1 = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; cfg1_mask = cfg1; fthlv = STM32_I2S_FIFO_SIZE * I2S_FIFO_TH_ONE_QUARTER / 4; @@ -553,32 +544,15 @@ static int stm32_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); - int ret, ier; i2s->substream = substream; spin_lock(&i2s->lock_fd); - if (i2s->refcount) { - dev_err(cpu_dai->dev, "%s stream already started\n", - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "Capture" : "Playback")); - spin_unlock(&i2s->lock_fd); - return -EBUSY; - } - i2s->refcount = 1; + i2s->refcount++; spin_unlock(&i2s->lock_fd); - ret = regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, I2S_IFCR_MASK); - if (ret < 0) - return ret; - - /* Enable ITs */ - ier = I2S_IER_OVRIE | I2S_IER_UDRIE; - if (STM32_I2S_IS_SLAVE(i2s)) - ier |= I2S_IER_TIFREIE; - - return regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, ier, ier); + return regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); } static int stm32_i2s_hw_params(struct snd_pcm_substream *substream, @@ -605,7 +579,7 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); bool playback_flg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - u32 cfg1_mask; + u32 cfg1_mask, ier; int ret; switch (cmd) { @@ -628,10 +602,48 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(cpu_dai->dev, "Error %d starting I2S\n", ret); return ret; } + + regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); + + if (playback_flg) { + ier = I2S_IER_UDRIE; + } else { + ier = I2S_IER_OVRIE; + + spin_lock(&i2s->lock_fd); + if (i2s->refcount == 1) + /* dummy write to trigger capture */ + regmap_write(i2s->regmap, + STM32_I2S_TXDR_REG, 0); + spin_unlock(&i2s->lock_fd); + } + + if (STM32_I2S_IS_SLAVE(i2s)) + ier |= I2S_IER_TIFREIE; + + regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, ier, ier); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (playback_flg) + regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, + I2S_IER_UDRIE, + (unsigned int)~I2S_IER_UDRIE); + else + regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, + I2S_IER_OVRIE, + (unsigned int)~I2S_IER_OVRIE); + + spin_lock(&i2s->lock_fd); + i2s->refcount--; + if (i2s->refcount) { + spin_unlock(&i2s->lock_fd); + break; + } + spin_unlock(&i2s->lock_fd); + dev_dbg(cpu_dai->dev, "stop I2S\n"); ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, @@ -641,10 +653,7 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } - cfg1_mask = I2S_CFG1_RXDMAEN; - if (playback_flg) - cfg1_mask = I2S_CFG1_TXDMAEN; - + cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, cfg1_mask, 0); break; @@ -662,10 +671,6 @@ static void stm32_i2s_shutdown(struct snd_pcm_substream *substream, i2s->substream = NULL; - spin_lock(&i2s->lock_fd); - i2s->refcount = 0; - spin_unlock(&i2s->lock_fd); - regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, I2S_CGFR_MCKOE, (unsigned int)~I2S_CGFR_MCKOE); } -- cgit v1.2.3 From 7ac45d1635a4cd2e99a4b11903d4a2815ca1b27b Mon Sep 17 00:00:00 2001 From: Julian Scheel Date: Wed, 24 May 2017 12:28:23 +0200 Subject: ASoC: simple-card: Fix misleading error message In case cpu could not be found the error message would always refer to /codec/ not being found in DT. Fix this by catching the cpu node not found case explicitly. Signed-off-by: Julian Scheel Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2c9dedab5184..565d057f0d14 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -233,13 +233,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, snprintf(prop, sizeof(prop), "%scpu", prefix); cpu = of_get_child_by_name(node, prop); + if (!cpu) { + ret = -EINVAL; + dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + goto dai_link_of_err; + } + snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); snprintf(prop, sizeof(prop), "%scodec", prefix); codec = of_get_child_by_name(node, prop); - if (!cpu || !codec) { + if (!codec) { ret = -EINVAL; dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); goto dai_link_of_err; -- cgit v1.2.3 From b08a20f58d2efcd88bf5276e34cd4020028accb7 Mon Sep 17 00:00:00 2001 From: Icenowy Zheng Date: Wed, 24 May 2017 18:05:59 +0800 Subject: ASoC: sun8i-codec-analog: split out mbias Allwinner V3s features an analog codec without MBIAS pin. Split out this part, in order to prepare for the V3s analog codec. Signed-off-by: Icenowy Zheng Reviewed-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec-analog.c | 35 ++++++++++++++++++++++++++++++----- 1 file changed, 30 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 6c17c99c2c8d..edcc3eb7cd9a 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -289,11 +289,6 @@ static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = { /* Microphone input */ SND_SOC_DAPM_INPUT("MIC1"), - /* Microphone Bias */ - SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, - SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN, - 0, NULL, 0), - /* Mic input path */ SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN, 0, NULL, 0), @@ -453,6 +448,27 @@ static int sun8i_codec_add_headphone(struct snd_soc_component *cmpnt) return 0; } +/* mbias specific widget */ +static const struct snd_soc_dapm_widget sun8i_codec_mbias_widgets[] = { + SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, + SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), +}; + +static int sun8i_codec_add_mbias(struct snd_soc_component *cmpnt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); + struct device *dev = cmpnt->dev; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_mbias_widgets, + ARRAY_SIZE(sun8i_codec_mbias_widgets)); + if (ret) + dev_err(dev, "Failed to add MBIAS DAPM widgets: %d\n", ret); + + return ret; +} + /* hmic specific widget */ static const struct snd_soc_dapm_widget sun8i_codec_hmic_widgets[] = { SND_SOC_DAPM_SUPPLY("HBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, @@ -679,6 +695,7 @@ struct sun8i_codec_analog_quirks { bool has_hmic; bool has_linein; bool has_lineout; + bool has_mbias; bool has_mic2; }; @@ -686,12 +703,14 @@ static const struct sun8i_codec_analog_quirks sun8i_a23_quirks = { .has_headphone = true, .has_hmic = true, .has_linein = true, + .has_mbias = true, .has_mic2 = true, }; static const struct sun8i_codec_analog_quirks sun8i_h3_quirks = { .has_linein = true, .has_lineout = true, + .has_mbias = true, .has_mic2 = true, }; @@ -734,6 +753,12 @@ static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) return ret; } + if (quirks->has_mbias) { + ret = sun8i_codec_add_mbias(cmpnt); + if (ret) + return ret; + } + if (quirks->has_mic2) { ret = sun8i_codec_add_mic2(cmpnt); if (ret) -- cgit v1.2.3 From 6298117a5c5c5c5217b59640d6df7fe078fa7d88 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 24 May 2017 10:59:38 +0100 Subject: ASoC: wm_adsp: Fix type warning in sprintf The shift member of struct soc_mixer_control is unsigned int. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 20695b691aff..a7dc76030ee4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2654,7 +2654,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); dsp->preloaded = ucontrol->value.integer.value[0]; -- cgit v1.2.3 From f6db09488f58372909728cea5a7c063ebf78f386 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 24 May 2017 10:59:39 +0100 Subject: ASoC: wm_adsp: Remove unused member of struct wm_coeff_ctl_ops The xinfo member of struct wm_coeff_ctl_ops is never used. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a7dc76030ee4..5aff83be375c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -482,8 +482,6 @@ struct wm_coeff_ctl_ops { struct snd_ctl_elem_value *ucontrol); int (*xput)(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - int (*xinfo)(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); }; struct wm_coeff_ctl { -- cgit v1.2.3 From 24069b589b02cc1292761b0f72623dd50ad1e19c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 18 May 2017 01:40:02 +0000 Subject: ASoC: hdmi-codec: remove multi detection support DesignWare HDMI driver (= dw-hdmi) is supporting HDMI sound, and its probe function was calling sound binding function multiple times as same HDMI device different port. Because of this behavior, commit 9731f82d601 ("ASoC: hdmi-codec: enable multi probe for ...") was added for multi detection case. But, this DesignWare HDMI detection/bind code was exchanged/adjusted by commit 69497eb9234 ("drm: bridge: dw-hdmi: Implement DRM bridge..."). Now, all DesignWare HDMI sound ports are detected as 1 bindng function. Because of this, hdmi-codec multi detection support is no longer needed. Thus, this patch removes commit 9731f82d601 ("ASoC: hdmi-codec: enable multi probe for ..."), and its related commit 340327a62c4 ("ASoC: hdmi-codec: Fix hdmi_of_xlate_dai_name...") commit 8480ac56795 ("ASoC: hdmi-codec: remove HDMI device unregister") commit 0c343a35bfe ("ASoC: hdmi-codec: fix spelling mistake: ...) Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 88 ++----------------------------------------- 1 file changed, 3 insertions(+), 85 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index a3f15149afcf..8659b76b066a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -25,17 +25,6 @@ #include /* This is only to get MAX_ELD_BYTES */ -struct hdmi_device { - struct device *dev; - struct list_head list; - int cnt; -}; -#define pos_to_hdmi_device(pos) container_of((pos), struct hdmi_device, list) -LIST_HEAD(hdmi_device_list); -static DEFINE_MUTEX(hdmi_mutex); - -#define DAI_NAME_SIZE 16 - #define HDMI_CODEC_CHMAP_IDX_UNKNOWN -1 struct hdmi_codec_channel_map_table { @@ -702,6 +691,7 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, } static struct snd_soc_dai_driver hdmi_i2s_dai = { + .name = "i2s-hifi", .id = DAI_ID_I2S, .playback = { .stream_name = "Playback", @@ -716,6 +706,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = { }; static const struct snd_soc_dai_driver hdmi_spdif_dai = { + .name = "spdif-hifi", .id = DAI_ID_SPDIF, .playback = { .stream_name = "Playback", @@ -728,32 +719,6 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .pcm_new = hdmi_codec_pcm_new, }; -static char hdmi_dai_name[][DAI_NAME_SIZE] = { - "hdmi-hifi.0", - "hdmi-hifi.1", - "hdmi-hifi.2", - "hdmi-hifi.3", -}; - -static int hdmi_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, - const char **dai_name) -{ - int id; - - if (args->args_count) - id = args->args[0]; - else - id = 0; - - if (id < ARRAY_SIZE(hdmi_dai_name)) { - *dai_name = hdmi_dai_name[id]; - return 0; - } - - return -EAGAIN; -} - static struct snd_soc_codec_driver hdmi_codec = { .component_driver = { .controls = hdmi_controls, @@ -762,7 +727,6 @@ static struct snd_soc_codec_driver hdmi_codec = { .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, .num_dapm_routes = ARRAY_SIZE(hdmi_routes), - .of_xlate_dai_name = hdmi_of_xlate_dai_name, }, }; @@ -771,8 +735,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) struct hdmi_codec_pdata *hcd = pdev->dev.platform_data; struct device *dev = &pdev->dev; struct hdmi_codec_priv *hcp; - struct hdmi_device *hd; - struct list_head *pos; int dai_count, i = 0; int ret; @@ -794,35 +756,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) if (!hcp) return -ENOMEM; - hd = NULL; - mutex_lock(&hdmi_mutex); - list_for_each(pos, &hdmi_device_list) { - struct hdmi_device *tmp = pos_to_hdmi_device(pos); - - if (tmp->dev == dev->parent) { - hd = tmp; - break; - } - } - - if (!hd) { - hd = devm_kzalloc(dev, sizeof(*hd), GFP_KERNEL); - if (!hd) { - mutex_unlock(&hdmi_mutex); - return -ENOMEM; - } - - hd->dev = dev->parent; - - list_add_tail(&hd->list, &hdmi_device_list); - } - mutex_unlock(&hdmi_mutex); - - if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) { - dev_err(dev, "too many hdmi codec are detected\n"); - return -EINVAL; - } - hcp->hcd = *hcd; mutex_init(&hcp->current_stream_lock); @@ -835,14 +768,11 @@ static int hdmi_codec_probe(struct platform_device *pdev) hcp->daidrv[i] = hdmi_i2s_dai; hcp->daidrv[i].playback.channels_max = hcd->max_i2s_channels; - hcp->daidrv[i].name = hdmi_dai_name[hd->cnt++]; i++; } - if (hcd->spdif) { + if (hcd->spdif) hcp->daidrv[i] = hdmi_spdif_dai; - hcp->daidrv[i].name = hdmi_dai_name[hd->cnt++]; - } ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, dai_count); @@ -859,20 +789,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) static int hdmi_codec_remove(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct list_head *pos; struct hdmi_codec_priv *hcp; - mutex_lock(&hdmi_mutex); - list_for_each(pos, &hdmi_device_list) { - struct hdmi_device *tmp = pos_to_hdmi_device(pos); - - if (tmp->dev == dev->parent) { - list_del(pos); - break; - } - } - mutex_unlock(&hdmi_mutex); - hcp = dev_get_drvdata(dev); kfree(hcp->chmap_info); snd_soc_unregister_codec(dev); -- cgit v1.2.3 From 96203fb4237bf70f0fd0fa307ca2975077db3ceb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 18 May 2017 01:40:20 +0000 Subject: ASoC: hdmi-codec: add .get_dai_id support ALSA SoC needs to know connected DAI ID for probing. It is not a big problem if device/driver was only for sound, but getting DAI ID will be difficult if device includes both Video/Sound, like HDMI. To solve this issue, this patch adds new .get_dai_id callback on hdmi_codec_ops Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/hdmi-codec.h | 9 +++++++++ sound/soc/codecs/hdmi-codec.c | 13 +++++++++++++ 2 files changed, 22 insertions(+) (limited to 'sound') diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 915c4357945c..9483c55f871b 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -18,9 +18,11 @@ #ifndef __HDMI_CODEC_H__ #define __HDMI_CODEC_H__ +#include #include #include #include +#include #include /* @@ -87,6 +89,13 @@ struct hdmi_codec_ops { */ int (*get_eld)(struct device *dev, void *data, uint8_t *buf, size_t len); + + /* + * Getting DAI ID + * Optional + */ + int (*get_dai_id)(struct snd_soc_component *comment, + struct device_node *endpoint); }; /* HDMI codec initalization data */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8659b76b066a..6d05161b625d 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -719,6 +719,18 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .pcm_new = hdmi_codec_pcm_new, }; +static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = -ENOTSUPP; /* see snd_soc_get_dai_id() */ + + if (hcp->hcd.ops->get_dai_id) + ret = hcp->hcd.ops->get_dai_id(component, endpoint); + + return ret; +} + static struct snd_soc_codec_driver hdmi_codec = { .component_driver = { .controls = hdmi_controls, @@ -727,6 +739,7 @@ static struct snd_soc_codec_driver hdmi_codec = { .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, .num_dapm_routes = ARRAY_SIZE(hdmi_routes), + .of_xlate_dai_id = hdmi_of_xlate_dai_id, }, }; -- cgit v1.2.3 From 503ada8a6d00c70f5b6fe37249e9a5e2f9c9e202 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 26 May 2017 10:47:07 +0100 Subject: ASoC: wm_adsp: Fix typo in algorithm list warning message The list terminator is 0xbedead but the message warning if it wasn't found was showing that 0xbeadead was expected. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5aff83be375c..65c059b5ffd7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1888,7 +1888,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } if (be32_to_cpu(val) != 0xbedead) - adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", + adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", pos + len, be32_to_cpu(val)); alg = kzalloc(len * 2, GFP_KERNEL | GFP_DMA); -- cgit v1.2.3 From 50aadc14cee74009c72e7d66954b15f27d45c02f Mon Sep 17 00:00:00 2001 From: Icenowy Zheng Date: Mon, 5 Jun 2017 21:27:20 +0800 Subject: ASoC: sun8i-codec-analog: prepare a mixer control/widget/route set for V3s Allwinner V3s has an analog codec without MIC2 and Line In, which will need a special set of mixer controls/widgets/routes, otherwise meaningless controls will be exported to userspace and confuse the user. Add the special set, and use it when the SoC has no MIC2 and Line In. Signed-off-by: Icenowy Zheng Reviewed-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec-analog.c | 101 ++++++++++++++++++++++++++++++++++- 1 file changed, 100 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index edcc3eb7cd9a..29c446068151 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -219,6 +219,22 @@ static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = { SUN8I_ADDA_LOMIXSC_MIC2, 1, 0), }; +/* mixer controls */ +static const struct snd_kcontrol_new sun8i_v3s_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("DAC Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_DACL, 1, 0), + SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_DACR, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_MIC1, 1, 0), +}; + /* ADC mixer controls */ static const struct snd_kcontrol_new sun8i_codec_adc_mixer_controls[] = { SOC_DAPM_DOUBLE_R("Mixer Capture Switch", @@ -243,6 +259,22 @@ static const struct snd_kcontrol_new sun8i_codec_adc_mixer_controls[] = { SUN8I_ADDA_LADCMIXSC_MIC2, 1, 0), }; +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun8i_v3s_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Mixer Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_OMIXRL, 1, 0), + SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_OMIXRR, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_MIC1, 1, 0), +}; + /* volume / mute controls */ static const DECLARE_TLV_DB_SCALE(sun8i_codec_out_mixer_pregain_scale, -450, 150, 0); @@ -292,8 +324,9 @@ static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = { /* Mic input path */ SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN, 0, NULL, 0), +}; - /* Mixers */ +static const struct snd_soc_dapm_widget sun8i_codec_mixer_widgets[] = { SND_SOC_DAPM_MIXER("Left Mixer", SUN8I_ADDA_DAC_PA_SRC, SUN8I_ADDA_DAC_PA_SRC_LMIXEN, 0, sun8i_codec_mixer_controls, @@ -312,10 +345,31 @@ static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = { ARRAY_SIZE(sun8i_codec_adc_mixer_controls)), }; +static const struct snd_soc_dapm_widget sun8i_v3s_codec_mixer_widgets[] = { + SND_SOC_DAPM_MIXER("Left Mixer", SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_LMIXEN, 0, + sun8i_v3s_codec_mixer_controls, + ARRAY_SIZE(sun8i_v3s_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_RMIXEN, 0, + sun8i_v3s_codec_mixer_controls, + ARRAY_SIZE(sun8i_v3s_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0, + sun8i_v3s_codec_adc_mixer_controls, + ARRAY_SIZE(sun8i_v3s_codec_adc_mixer_controls)), + SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCREN, 0, + sun8i_v3s_codec_adc_mixer_controls, + ARRAY_SIZE(sun8i_v3s_codec_adc_mixer_controls)), +}; + static const struct snd_soc_dapm_route sun8i_codec_common_routes[] = { /* Microphone Routes */ { "Mic1 Amplifier", NULL, "MIC1"}, +}; +static const struct snd_soc_dapm_route sun8i_codec_mixer_routes[] = { /* Left Mixer Routes */ { "Left Mixer", "DAC Playback Switch", "Left DAC" }, { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, @@ -714,6 +768,48 @@ static const struct sun8i_codec_analog_quirks sun8i_h3_quirks = { .has_mic2 = true, }; +static int sun8i_codec_analog_add_mixer(struct snd_soc_component *cmpnt, + const struct sun8i_codec_analog_quirks *quirks) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); + struct device *dev = cmpnt->dev; + int ret; + + if (!quirks->has_mic2 && !quirks->has_linein) { + /* + * Apply the special widget set which has uses a control + * without MIC2 and Line In, for SoCs without these. + * TODO: not all special cases are supported now, this case + * is present because it's the case of V3s. + */ + ret = snd_soc_dapm_new_controls(dapm, + sun8i_v3s_codec_mixer_widgets, + ARRAY_SIZE(sun8i_v3s_codec_mixer_widgets)); + if (ret) { + dev_err(dev, "Failed to add V3s Mixer DAPM widgets: %d\n", ret); + return ret; + } + } else { + /* Apply the generic mixer widget set. */ + ret = snd_soc_dapm_new_controls(dapm, + sun8i_codec_mixer_widgets, + ARRAY_SIZE(sun8i_codec_mixer_widgets)); + if (ret) { + dev_err(dev, "Failed to add Mixer DAPM widgets: %d\n", ret); + return ret; + } + } + + ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_mixer_routes, + ARRAY_SIZE(sun8i_codec_mixer_routes)); + if (ret) { + dev_err(dev, "Failed to add Mixer DAPM routes: %d\n", ret); + return ret; + } + + return 0; +} + static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) { struct device *dev = cmpnt->dev; @@ -728,6 +824,9 @@ static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) quirks = of_device_get_match_data(dev); /* Add controls, widgets, and routes for individual features */ + ret = sun8i_codec_analog_add_mixer(cmpnt, quirks); + if (ret) + return ret; if (quirks->has_headphone) { ret = sun8i_codec_add_headphone(cmpnt); -- cgit v1.2.3 From 2cfeaec0ec896bc0b8aad2de28a3de4572c7e4a1 Mon Sep 17 00:00:00 2001 From: Icenowy Zheng Date: Mon, 5 Jun 2017 21:27:21 +0800 Subject: ASoC: sun8i-codec-analog: add support for V3s SoC The V3s SoC features an analog codec with headphone support but without mic2 and linein. Add support for it. Signed-off-by: Icenowy Zheng Reviewed-by: Chen-Yu Tsai Acked-by: Rob Herring Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt | 1 + sound/soc/sunxi/sun8i-codec-analog.c | 9 +++++++++ 2 files changed, 10 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt index 779b735781ba..1b6e7c4e50ab 100644 --- a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt +++ b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt @@ -4,6 +4,7 @@ Required properties: - compatible: must be one of the following compatibles: - "allwinner,sun8i-a23-codec-analog" - "allwinner,sun8i-h3-codec-analog" + - "allwinner,sun8i-v3s-codec-analog" Required properties if not a sub-node of the PRCM node: - reg: must contain the registers location and length diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 29c446068151..485e79f292c4 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -810,6 +810,11 @@ static int sun8i_codec_analog_add_mixer(struct snd_soc_component *cmpnt, return 0; } +static const struct sun8i_codec_analog_quirks sun8i_v3s_quirks = { + .has_headphone = true, + .has_hmic = true, +}; + static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) { struct device *dev = cmpnt->dev; @@ -886,6 +891,10 @@ static const struct of_device_id sun8i_codec_analog_of_match[] = { .compatible = "allwinner,sun8i-h3-codec-analog", .data = &sun8i_h3_quirks, }, + { + .compatible = "allwinner,sun8i-v3s-codec-analog", + .data = &sun8i_v3s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun8i_codec_analog_of_match); -- cgit v1.2.3 From 8b2840b6daca728cecfa925b50bf638189e2fbca Mon Sep 17 00:00:00 2001 From: Icenowy Zheng Date: Mon, 5 Jun 2017 21:27:22 +0800 Subject: ASoC: sun4i-codec: Add support for V3s codec The codec in the V3s is similar to the one found on the A31. One key difference is the analog path controls are routed through the PRCM block. This is supported by the sun8i-codec-analog driver, and tied into this codec driver with the audio card's aux_dev. In addition, the V3s does not have LINEIN, LINEOUT, MBIAS and MIC2, MIC3, and the FIFO related registers are like H3. Signed-off-by: Icenowy Zheng Reviewed-by: Chen-Yu Tsai Acked-by: Rob Herring Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sun4i-codec.txt | 11 ++-- sound/soc/sunxi/sun4i-codec.c | 63 ++++++++++++++++++++++ 2 files changed, 70 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 3863531d1e6d..2d4e10deb6f4 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -7,6 +7,7 @@ Required properties: - "allwinner,sun7i-a20-codec" - "allwinner,sun8i-a23-codec" - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" - reg: must contain the registers location and length - interrupts: must contain the codec interrupt - dmas: DMA channels for tx and rx dma. See the DMA client binding, @@ -25,6 +26,7 @@ Required properties for the following compatibles: - "allwinner,sun6i-a31-codec" - "allwinner,sun8i-a23-codec" - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" - resets: phandle to the reset control for this device - allwinner,audio-routing: A list of the connections between audio components. Each entry is a pair of strings, the first being the @@ -34,15 +36,15 @@ Required properties for the following compatibles: Audio pins on the SoC: "HP" "HPCOM" - "LINEIN" - "LINEOUT" (not on sun8i-a23) + "LINEIN" (not on sun8i-v3s) + "LINEOUT" (not on sun8i-a23 or sun8i-v3s) "MIC1" - "MIC2" + "MIC2" (not on sun8i-v3s) "MIC3" (sun6i-a31 only) Microphone biases from the SoC: "HBIAS" - "MBIAS" + "MBIAS" (not on sun8i-v3s) Board connectors: "Headphone" @@ -55,6 +57,7 @@ Required properties for the following compatibles: Required properties for the following compatibles: - "allwinner,sun8i-a23-codec" - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" - allwinner,codec-analog-controls: A phandle to the codec analog controls block in the PRCM. diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index c3aab10fa085..150069987c0c 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1339,6 +1339,44 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return card; }; +static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return ERR_PTR(-ENOMEM); + + aux_dev.codec_of_node = of_parse_phandle(dev->of_node, + "allwinner,codec-analog-controls", + 0); + if (!aux_dev.codec_of_node) { + dev_err(dev, "Can't find analog controls for codec.\n"); + return ERR_PTR(-EINVAL); + }; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return ERR_PTR(-ENOMEM); + + card->dev = dev; + card->name = "V3s Audio Codec"; + card->dapm_widgets = sun6i_codec_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); + card->dapm_routes = sun8i_codec_card_routes; + card->num_dapm_routes = ARRAY_SIZE(sun8i_codec_card_routes); + card->aux_dev = &aux_dev; + card->num_aux_devs = 1; + card->fully_routed = true; + + ret = snd_soc_of_parse_audio_routing(card, "allwinner,audio-routing"); + if (ret) + dev_warn(dev, "failed to parse audio-routing: %d\n", ret); + + return card; +}; + static const struct regmap_config sun4i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, @@ -1374,6 +1412,13 @@ static const struct regmap_config sun8i_h3_codec_regmap_config = { .max_register = SUN8I_H3_CODEC_ADC_DBG, }; +static const struct regmap_config sun8i_v3s_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN8I_H3_CODEC_ADC_DBG, +}; + struct sun4i_codec_quirks { const struct regmap_config *regmap_config; const struct snd_soc_codec_driver *codec; @@ -1437,6 +1482,20 @@ static const struct sun4i_codec_quirks sun8i_h3_codec_quirks = { .has_reset = true, }; +static const struct sun4i_codec_quirks sun8i_v3s_codec_quirks = { + .regmap_config = &sun8i_v3s_codec_regmap_config, + /* + * TODO The codec structure should be split out, like + * H3, when adding digital audio processing support. + */ + .codec = &sun8i_a23_codec_codec, + .create_card = sun8i_v3s_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN6I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN8I_H3_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN6I_CODEC_ADC_RXDATA, + .has_reset = true, +}; + static const struct of_device_id sun4i_codec_of_match[] = { { .compatible = "allwinner,sun4i-a10-codec", @@ -1458,6 +1517,10 @@ static const struct of_device_id sun4i_codec_of_match[] = { .compatible = "allwinner,sun8i-h3-codec", .data = &sun8i_h3_codec_quirks, }, + { + .compatible = "allwinner,sun8i-v3s-codec", + .data = &sun8i_v3s_codec_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); -- cgit v1.2.3 From c8597af855f3e34aaebaff0e5c3dbd07611c87f1 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Jun 2017 15:45:07 +0100 Subject: ASoC: topology: Allow bespoke configuration post widget creation Current topology only allows for widget configuration before the widget is registered. This patch also allows further configuration and usage after registration is complete. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 002772e3ba2c..273a374e741c 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -344,12 +344,24 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic widget to component driver. This is mainly for + * external widgets where we can assign private data/ops */ +static int soc_tplg_widget_ready(struct soc_tplg *tplg, + struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) +{ + if (tplg->comp && tplg->ops && tplg->ops->widget_ready) + return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + + return 0; +} + /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, dai_drv, pcm, dai); return 0; } @@ -1580,8 +1592,16 @@ widget: kfree(template.sname); kfree(template.name); list_add(&widget->dobj.list, &tplg->comp->dobj_list); + + ret = soc_tplg_widget_ready(tplg, widget, w); + if (ret < 0) + goto ready_err; + return 0; +ready_err: + snd_soc_tplg_widget_remove(widget); + snd_soc_dapm_free_widget(widget); hdr_err: kfree(template.sname); err: -- cgit v1.2.3 From cc9d4714a8da98f905c63d74e9897fc6f4563fca Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Jun 2017 15:45:08 +0100 Subject: ASoC: topology: rephrase deferred binding warning. Rewrite the message to be more meaningful. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 273a374e741c..f24d1f2e82a0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1648,7 +1648,7 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) */ if (!card || !card->instantiated) { dev_warn(tplg->dev, "ASoC: Parent card not yet available," - "Do not add new widgets now\n"); + " widget card binding deferred\n"); return 0; } -- cgit v1.2.3 From c3421a6a65abc636b067eb15a4c5e9cb59e91c95 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Jun 2017 15:45:09 +0100 Subject: ASoC: topology: Dont free template strings whilst they are in use. Template name pointers are copied when creating new widgets and are freed in widget destroy. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f24d1f2e82a0..7006cf3007b5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1477,6 +1477,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, if (template.id < 0) return template.id; + /* strings are allocated here, but used and freed by the widget */ template.name = kstrdup(w->name, GFP_KERNEL); if (!template.name) return -ENOMEM; @@ -1589,8 +1590,6 @@ widget: widget->dobj.widget.kcontrol_type = kcontrol_type; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; - kfree(template.sname); - kfree(template.name); list_add(&widget->dobj.list, &tplg->comp->dobj_list); ret = soc_tplg_widget_ready(tplg, widget, w); -- cgit v1.2.3 From ca3b5ad30c0b32f8d12ddb307b698ff56d56c2aa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 6 Jun 2017 23:16:01 +0000 Subject: ASoC: hdmi-codec: remove unused ratec struct snd_pcm_hw_constraint_list ratec is not used. Let's remove it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6d05161b625d..22ed0dc88f0a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -282,7 +282,6 @@ struct hdmi_codec_priv { struct hdmi_codec_daifmt daifmt[2]; struct mutex current_stream_lock; struct snd_pcm_substream *current_stream; - struct snd_pcm_hw_constraint_list ratec; uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; -- cgit v1.2.3 From d2fdcc285f8c79ab1a6d20e5196eb173a105b365 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 7 Jun 2017 00:37:05 +0000 Subject: ASoC: simple-card: remove duplicate parameter from asoc_simple_card_parse_of() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 565d057f0d14..33ff487193f9 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -356,12 +356,12 @@ static int asoc_simple_card_parse_aux_devs(struct device_node *node, return 0; } -static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *priv) +static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_card *card = simple_priv_to_card(priv); struct device_node *dai_link; + struct device_node *node = dev->of_node; int ret; if (!node) @@ -460,7 +460,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv); + ret = asoc_simple_card_parse_of(priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); -- cgit v1.2.3 From 102ebe266c317da59471e2cde0dce603de031482 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Jun 2017 17:02:02 +0100 Subject: ASoC: Back out post commit widget creation changes Due to build errors revert commit c8597af855f3 (ASoC: topology: Allow bespoke configuration post widget creation) until they can be fixed. Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 24 ++---------------------- 1 file changed, 2 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 7006cf3007b5..f4ec236a418e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -344,24 +344,12 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, return 0; } -/* optionally pass new dynamic widget to component driver. This is mainly for - * external widgets where we can assign private data/ops */ -static int soc_tplg_widget_ready(struct soc_tplg *tplg, - struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) -{ - if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); - - return 0; -} - /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) + struct snd_soc_dai_driver *dai_drv) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv, pcm, dai); + return tplg->ops->dai_load(tplg->comp, dai_drv); return 0; } @@ -1591,16 +1579,8 @@ widget: widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; list_add(&widget->dobj.list, &tplg->comp->dobj_list); - - ret = soc_tplg_widget_ready(tplg, widget, w); - if (ret < 0) - goto ready_err; - return 0; -ready_err: - snd_soc_tplg_widget_remove(widget); - snd_soc_dapm_free_widget(widget); hdr_err: kfree(template.sname); err: -- cgit v1.2.3 From 69beca69d68b69b38c2610b8b5fd2e27a40d441b Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Fri, 9 Jun 2017 15:06:54 +0300 Subject: ASoC: omap-mcbsp: Use sysfs_match_string() helper Use sysfs_match_string() helper instead of open coded variant. Cc: Peter Ujfalusi Cc: Jarkko Nikula Cc: Mark Brown Signed-off-by: Andy Shevchenko Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..7a54e3083203 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -835,15 +835,11 @@ static ssize_t dma_op_mode_store(struct device *dev, const char *buf, size_t size) { struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - const char * const *s; - int i = 0; - - for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) - if (sysfs_streq(buf, *s)) - break; + int i; - if (i == ARRAY_SIZE(dma_op_modes)) - return -EINVAL; + i = sysfs_match_string(dma_op_modes, buf); + if (i < 0) + return i; spin_lock_irq(&mcbsp->lock); if (!mcbsp->free) { -- cgit v1.2.3 From ebd259d33a900b28ef774c4c26e8ce6e2baea7e5 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 9 Jun 2017 15:43:23 +0100 Subject: ASoC: topology: Allow bespoke configuration post widget creation Current topology only allows for widget configuration before the widget is registered. This patch also allows further configuration and usage after registration is complete. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 3 +++ sound/soc/soc-topology.c | 19 +++++++++++++++++++ 2 files changed, 22 insertions(+) (limited to 'sound') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index b8da221615e0..f552c3f56368 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -118,6 +118,9 @@ struct snd_soc_tplg_ops { int (*widget_load)(struct snd_soc_component *, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); + int (*widget_ready)(struct snd_soc_component *, + struct snd_soc_dapm_widget *, + struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f4ec236a418e..12e189701924 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -344,6 +344,17 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic widget to component driver. This is mainly for + * external widgets where we can assign private data/ops */ +static int soc_tplg_widget_ready(struct soc_tplg *tplg, + struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) +{ + if (tplg->comp && tplg->ops && tplg->ops->widget_ready) + return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + + return 0; +} + /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, struct snd_soc_dai_driver *dai_drv) @@ -1579,8 +1590,16 @@ widget: widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; list_add(&widget->dobj.list, &tplg->comp->dobj_list); + + ret = soc_tplg_widget_ready(tplg, widget, w); + if (ret < 0) + goto ready_err; + return 0; +ready_err: + snd_soc_tplg_widget_remove(widget); + snd_soc_dapm_free_widget(widget); hdr_err: kfree(template.sname); err: -- cgit v1.2.3 From fc05a5b222530617d99d0e803abb262130fdb0c4 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Tue, 13 Jun 2017 15:27:46 +0800 Subject: ASoC: rockchip: add support for pdm controller The Pulse Density Modulation Interface Controller (PDMC) is a PDM interface controller and decoder that support PDM format. It integrates a clock generator driving the PDM microphone and embeds filters which decimate the incoming bit stream to obtain most common audio rates. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip,pdm.txt | 39 ++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_pdm.c | 516 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.h | 83 ++++ 5 files changed, 649 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 sound/soc/rockchip/rockchip_pdm.c create mode 100644 sound/soc/rockchip/rockchip_pdm.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.txt b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt new file mode 100644 index 000000000000..921729de7346 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt @@ -0,0 +1,39 @@ +* Rockchip PDM controller + +Required properties: + +- compatible: "rockchip,pdm" +- reg: physical base address of the controller and length of memory mapped + region. +- dmas: DMA specifiers for rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain following: + - "pdm_hclk": clock for PDM BUS + - "pdm_clk" : clock for PDM controller +- pinctrl-names: Must contain a "default" entry. +- pinctrl-N: One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. + +Example for rk3328 PDM controller: + +pdm: pdm@ff040000 { + compatible = "rockchip,pdm"; + reg = <0x0 0xff040000 0x0 0x1000>; + clocks = <&clk_pdm>, <&clk_gates28 0>; + clock-names = "pdm_clk", "pdm_hclk"; + dmas = <&pdma 16>; + #dma-cells = <1>; + dma-names = "rx"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pdmm0_clk + &pdmm0_fsync + &pdmm0_sdi0 + &pdmm0_sdi1 + &pdmm0_sdi2 + &pdmm0_sdi3>; + pinctrl-1 = <&pdmm0_sleep>; + status = "disabled"; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index e3ca1e973de5..c84487805876 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -15,6 +15,15 @@ config SND_SOC_ROCKCHIP_I2S Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. +config SND_SOC_ROCKCHIP_PDM + tristate "Rockchip PDM Controller Driver" + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for PDM driver for + Rockchip PDM Controller. The Controller supports up to maximum of + 8 channels record. + config SND_SOC_ROCKCHIP_SPDIF tristate "Rockchip SPDIF Device Driver" depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 991f91bea9f9..105f0e14a4ab 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,8 +1,10 @@ # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c new file mode 100644 index 000000000000..c5ddeed97260 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -0,0 +1,516 @@ +/* + * Rockchip PDM ALSA SoC Digital Audio Interface(DAI) driver + * + * Copyright (C) 2017 Fuzhou Rockchip Electronics Co., Ltd + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "rockchip_pdm.h" + +#define PDM_DMA_BURST_SIZE (16) /* size * width: 16*4 = 64 bytes */ + +struct rk_pdm_dev { + struct device *dev; + struct clk *clk; + struct clk *hclk; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data capture_dma_data; +}; + +struct rk_pdm_clkref { + unsigned int sr; + unsigned int clk; +}; + +static struct rk_pdm_clkref clkref[] = { + { 8000, 40960000 }, + { 11025, 56448000 }, + { 12000, 61440000 }, +}; + +static unsigned int get_pdm_clk(unsigned int sr) +{ + unsigned int i, count, clk, div; + + clk = 0; + if (!sr) + return clk; + + count = ARRAY_SIZE(clkref); + for (i = 0; i < count; i++) { + if (sr % clkref[i].sr) + continue; + div = sr / clkref[i].sr; + if ((div & (div - 1)) == 0) { + clk = clkref[i].clk; + break; + } + } + + return clk; +} + +static inline struct rk_pdm_dev *to_info(struct snd_soc_dai *dai) +{ + return snd_soc_dai_get_drvdata(dai); +} + +static void rockchip_pdm_rxctrl(struct rk_pdm_dev *pdm, int on) +{ + if (on) { + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, + PDM_DMA_RD_MSK, PDM_DMA_RD_EN); + regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, + PDM_RX_MASK, PDM_RX_START); + } else { + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, + PDM_DMA_RD_MSK, PDM_DMA_RD_DIS); + regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, + PDM_RX_MASK | PDM_RX_CLR_MASK, + PDM_RX_STOP | PDM_RX_CLR_WR); + } +} + +static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk_pdm_dev *pdm = to_info(dai); + unsigned int val = 0; + unsigned int clk_rate, clk_div, samplerate; + int ret; + + samplerate = params_rate(params); + clk_rate = get_pdm_clk(samplerate); + if (!clk_rate) + return -EINVAL; + + ret = clk_set_rate(pdm->clk, clk_rate); + if (ret) + return -EINVAL; + + clk_div = DIV_ROUND_CLOSEST(clk_rate, samplerate); + + switch (clk_div) { + case 320: + val = PDM_CLK_320FS; + break; + case 640: + val = PDM_CLK_640FS; + break; + case 1280: + val = PDM_CLK_1280FS; + break; + case 2560: + val = PDM_CLK_2560FS; + break; + case 5120: + val = PDM_CLK_5120FS; + break; + default: + dev_err(pdm->dev, "unsupported div: %d\n", clk_div); + return -EINVAL; + } + + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_DS_RATIO_MSK, val); + regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, + PDM_HPF_CF_MSK, PDM_HPF_60HZ); + regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, + PDM_HPF_LE | PDM_HPF_RE, PDM_HPF_LE | PDM_HPF_RE); + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_CLK_EN, PDM_CLK_EN); + + val = 0; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + val |= PDM_VDW(8); + break; + case SNDRV_PCM_FORMAT_S16_LE: + val |= PDM_VDW(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= PDM_VDW(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= PDM_VDW(24); + break; + case SNDRV_PCM_FORMAT_S32_LE: + val |= PDM_VDW(32); + break; + default: + return -EINVAL; + } + + switch (params_channels(params)) { + case 8: + val |= PDM_PATH3_EN; + /* fallthrough */ + case 6: + val |= PDM_PATH2_EN; + /* fallthrough */ + case 4: + val |= PDM_PATH1_EN; + /* fallthrough */ + case 2: + val |= PDM_PATH0_EN; + break; + default: + dev_err(pdm->dev, "invalid channel: %d\n", + params_channels(params)); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + regmap_update_bits(pdm->regmap, PDM_CTRL0, + PDM_PATH_MSK | PDM_VDW_MSK, + val); + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, PDM_DMA_RDL_MSK, + PDM_DMA_RDL(16)); + regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, + PDM_RX_MASK | PDM_RX_CLR_MASK, + PDM_RX_STOP | PDM_RX_CLR_WR); + } + + return 0; +} + +static int rockchip_pdm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct rk_pdm_dev *pdm = to_info(cpu_dai); + unsigned int mask = 0, val = 0; + + mask = PDM_CKP_MSK; + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val = PDM_CKP_NORMAL; + break; + case SND_SOC_DAIFMT_IB_NF: + val = PDM_CKP_INVERTED; + break; + default: + return -EINVAL; + } + + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, mask, val); + + return 0; +} + +static int rockchip_pdm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct rk_pdm_dev *pdm = to_info(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_pdm_rxctrl(pdm, 1); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_pdm_rxctrl(pdm, 0); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rockchip_pdm_dai_probe(struct snd_soc_dai *dai) +{ + struct rk_pdm_dev *pdm = to_info(dai); + + dai->capture_dma_data = &pdm->capture_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops rockchip_pdm_dai_ops = { + .set_fmt = rockchip_pdm_set_fmt, + .trigger = rockchip_pdm_trigger, + .hw_params = rockchip_pdm_hw_params, +}; + +#define ROCKCHIP_PDM_RATES SNDRV_PCM_RATE_8000_192000 +#define ROCKCHIP_PDM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rockchip_pdm_dai = { + .probe = rockchip_pdm_dai_probe, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = ROCKCHIP_PDM_RATES, + .formats = ROCKCHIP_PDM_FORMATS, + }, + .ops = &rockchip_pdm_dai_ops, + .symmetric_rates = 1, +}; + +static const struct snd_soc_component_driver rockchip_pdm_component = { + .name = "rockchip-pdm", +}; + +static int rockchip_pdm_runtime_suspend(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + + clk_disable_unprepare(pdm->clk); + clk_disable_unprepare(pdm->hclk); + + return 0; +} + +static int rockchip_pdm_runtime_resume(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(pdm->clk); + if (ret) { + dev_err(pdm->dev, "clock enable failed %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(pdm->hclk); + if (ret) { + dev_err(pdm->dev, "hclock enable failed %d\n", ret); + return ret; + } + + return 0; +} + +static bool rockchip_pdm_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_SYSCONFIG: + case PDM_CTRL0: + case PDM_CTRL1: + case PDM_CLK_CTRL: + case PDM_HPF_CTRL: + case PDM_FIFO_CTRL: + case PDM_DMA_CTRL: + case PDM_INT_EN: + case PDM_INT_CLR: + case PDM_DATA_VALID: + return true; + default: + return false; + } +} + +static bool rockchip_pdm_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_SYSCONFIG: + case PDM_CTRL0: + case PDM_CTRL1: + case PDM_CLK_CTRL: + case PDM_HPF_CTRL: + case PDM_FIFO_CTRL: + case PDM_DMA_CTRL: + case PDM_INT_EN: + case PDM_INT_CLR: + case PDM_INT_ST: + case PDM_DATA_VALID: + case PDM_VERSION: + return true; + default: + return false; + } +} + +static bool rockchip_pdm_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_SYSCONFIG: + case PDM_INT_CLR: + case PDM_INT_ST: + return true; + default: + return false; + } +} + +static const struct regmap_config rockchip_pdm_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = PDM_VERSION, + .writeable_reg = rockchip_pdm_wr_reg, + .readable_reg = rockchip_pdm_rd_reg, + .volatile_reg = rockchip_pdm_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int rockchip_pdm_probe(struct platform_device *pdev) +{ + struct rk_pdm_dev *pdm; + struct resource *res; + void __iomem *regs; + int ret; + + pdm = devm_kzalloc(&pdev->dev, sizeof(*pdm), GFP_KERNEL); + if (!pdm) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + pdm->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &rockchip_pdm_regmap_config); + if (IS_ERR(pdm->regmap)) + return PTR_ERR(pdm->regmap); + + pdm->capture_dma_data.addr = res->start + PDM_RXFIFO_DATA; + pdm->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + pdm->capture_dma_data.maxburst = PDM_DMA_BURST_SIZE; + + pdm->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, pdm); + + pdm->clk = devm_clk_get(&pdev->dev, "pdm_clk"); + if (IS_ERR(pdm->clk)) + return PTR_ERR(pdm->clk); + + pdm->hclk = devm_clk_get(&pdev->dev, "pdm_hclk"); + if (IS_ERR(pdm->hclk)) + return PTR_ERR(pdm->hclk); + + ret = clk_prepare_enable(pdm->hclk); + if (ret) + return ret; + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = rockchip_pdm_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &rockchip_pdm_component, + &rockchip_pdm_dai, 1); + + if (ret) { + dev_err(&pdev->dev, "could not register dai: %d\n", ret); + goto err_suspend; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "could not register pcm: %d\n", ret); + goto err_suspend; + } + + return 0; + +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + rockchip_pdm_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); + + clk_disable_unprepare(pdm->hclk); + + return ret; +} + +static int rockchip_pdm_remove(struct platform_device *pdev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + rockchip_pdm_runtime_suspend(&pdev->dev); + + clk_disable_unprepare(pdm->clk); + clk_disable_unprepare(pdm->hclk); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int rockchip_pdm_suspend(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + + regcache_mark_dirty(pdm->regmap); + + return 0; +} + +static int rockchip_pdm_resume(struct device *dev) +{ + struct rk_pdm_dev *pdm = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + + ret = regcache_sync(pdm->regmap); + + pm_runtime_put(dev); + + return ret; +} +#endif + +static const struct dev_pm_ops rockchip_pdm_pm_ops = { + SET_RUNTIME_PM_OPS(rockchip_pdm_runtime_suspend, + rockchip_pdm_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(rockchip_pdm_suspend, rockchip_pdm_resume) +}; + +static const struct of_device_id rockchip_pdm_match[] = { + { .compatible = "rockchip,pdm", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rockchip_pdm_match); + +static struct platform_driver rockchip_pdm_driver = { + .probe = rockchip_pdm_probe, + .remove = rockchip_pdm_remove, + .driver = { + .name = "rockchip-pdm", + .of_match_table = of_match_ptr(rockchip_pdm_match), + .pm = &rockchip_pdm_pm_ops, + }, +}; + +module_platform_driver(rockchip_pdm_driver); + +MODULE_AUTHOR("Sugar "); +MODULE_DESCRIPTION("Rockchip PDM Controller Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pdm.h b/sound/soc/rockchip/rockchip_pdm.h new file mode 100644 index 000000000000..886b48d128fd --- /dev/null +++ b/sound/soc/rockchip/rockchip_pdm.h @@ -0,0 +1,83 @@ +/* + * Rockchip PDM ALSA SoC Digital Audio Interface(DAI) driver + * + * Copyright (C) 2017 Fuzhou Rockchip Electronics Co., Ltd + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef _ROCKCHIP_PDM_H +#define _ROCKCHIP_PDM_H + +/* PDM REGS */ +#define PDM_SYSCONFIG (0x0000) +#define PDM_CTRL0 (0x0004) +#define PDM_CTRL1 (0x0008) +#define PDM_CLK_CTRL (0x000c) +#define PDM_HPF_CTRL (0x0010) +#define PDM_FIFO_CTRL (0x0014) +#define PDM_DMA_CTRL (0x0018) +#define PDM_INT_EN (0x001c) +#define PDM_INT_CLR (0x0020) +#define PDM_INT_ST (0x0024) +#define PDM_RXFIFO_DATA (0x0030) +#define PDM_DATA_VALID (0x0054) +#define PDM_VERSION (0x0058) + +/* PDM_SYSCONFIG */ +#define PDM_RX_MASK (0x1 << 2) +#define PDM_RX_START (0x1 << 2) +#define PDM_RX_STOP (0x0 << 2) +#define PDM_RX_CLR_MASK (0x1 << 0) +#define PDM_RX_CLR_WR (0x1 << 0) +#define PDM_RX_CLR_DONE (0x0 << 0) + +/* PDM CTRL0 */ +#define PDM_PATH_MSK (0xf << 27) +#define PDM_PATH3_EN BIT(30) +#define PDM_PATH2_EN BIT(29) +#define PDM_PATH1_EN BIT(28) +#define PDM_PATH0_EN BIT(27) +#define PDM_HWT_EN BIT(26) +#define PDM_VDW_MSK (0x1f << 0) +#define PDM_VDW(X) ((X - 1) << 0) + +/* PDM CLK CTRL */ +#define PDM_CLK_MSK BIT(5) +#define PDM_CLK_EN BIT(5) +#define PDM_CLK_DIS (0x0 << 5) +#define PDM_CKP_MSK BIT(3) +#define PDM_CKP_NORMAL (0x0 << 3) +#define PDM_CKP_INVERTED BIT(3) +#define PDM_DS_RATIO_MSK (0x7 << 0) +#define PDM_CLK_320FS (0x0 << 0) +#define PDM_CLK_640FS (0x1 << 0) +#define PDM_CLK_1280FS (0x2 << 0) +#define PDM_CLK_2560FS (0x3 << 0) +#define PDM_CLK_5120FS (0x4 << 0) + +/* PDM HPF CTRL */ +#define PDM_HPF_LE BIT(3) +#define PDM_HPF_RE BIT(2) +#define PDM_HPF_CF_MSK (0x3 << 0) +#define PDM_HPF_3P79HZ (0x0 << 0) +#define PDM_HPF_60HZ (0x1 << 0) +#define PDM_HPF_243HZ (0x2 << 0) +#define PDM_HPF_493HZ (0x3 << 0) + +/* PDM DMA CTRL */ +#define PDM_DMA_RD_MSK BIT(8) +#define PDM_DMA_RD_EN BIT(8) +#define PDM_DMA_RD_DIS (0x0 << 8) +#define PDM_DMA_RDL_MSK (0x7f << 0) +#define PDM_DMA_RDL(X) ((X - 1) << 0) + +#endif /* _ROCKCHIP_PDM_H */ -- cgit v1.2.3 From b02ee56087adae4819ce4d91c08d57403f71fd34 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Jun 2017 23:37:24 +0200 Subject: ASoC: mediatek: Constify hw_constraints snd_pcm_hw_constraint_list(), *_ratnums() and *_ratdens() receive the const pointers. Constify the corresponding static objects for better hardening. Signed-off-by: Takashi Iwai Acked-By: Matthias Brugger Signed-off-by: Mark Brown --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index aa5b31b121e3..70f61d53fe05 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -107,7 +107,7 @@ static const struct snd_kcontrol_new mt2701_cs42448_controls[] = { static const unsigned int mt2701_cs42448_sampling_rates[] = {48000}; -static struct snd_pcm_hw_constraint_list mt2701_cs42448_constraints_rates = { +static const struct snd_pcm_hw_constraint_list mt2701_cs42448_constraints_rates = { .count = ARRAY_SIZE(mt2701_cs42448_sampling_rates), .list = mt2701_cs42448_sampling_rates, .mask = 0, -- cgit v1.2.3 From 0994c030443b50089b8ac74bc863d71238739f2e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Jun 2017 23:37:25 +0200 Subject: ASoC: samsung: Constify hw_constraints snd_pcm_hw_constraint_list(), *_ratnums() and *_ratdens() receive the const pointers. Constify the corresponding static objects for better hardening. Signed-off-by: Takashi Iwai Reviewed-by: Krzysztof Kozlowski Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 81a78940967c..55538e333cc8 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -44,7 +44,7 @@ struct s3c24xx_uda134x { static unsigned int rates[33 * 2]; #ifdef ENFORCE_RATES -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { +static const struct snd_pcm_hw_constraint_list hw_constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, -- cgit v1.2.3 From 55f42d2e28a42b06907c916c3c71ceb6dfb5afc4 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Fri, 9 Jun 2017 15:59:32 +0800 Subject: ASoC: rockchip: add bindings for spdif controller this patch add compatible for rk3228/rk3328 spdif, Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-spdif.txt | 2 ++ sound/soc/rockchip/rockchip_spdif.c | 4 ++++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt index 11046429a118..4706b96d450b 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -9,7 +9,9 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-spdif" - "rockchip,rk3188-spdif" + - "rockchip,rk3228-spdif" - "rockchip,rk3288-spdif" + - "rockchip,rk3328-spdif" - "rockchip,rk3366-spdif" - "rockchip,rk3368-spdif" - "rockchip,rk3399-spdif" diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index fa8101d1e16f..ee5055d47d13 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -49,8 +49,12 @@ static const struct of_device_id rk_spdif_match[] = { .data = (void *)RK_SPDIF_RK3066 }, { .compatible = "rockchip,rk3188-spdif", .data = (void *)RK_SPDIF_RK3188 }, + { .compatible = "rockchip,rk3228-spdif", + .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3288-spdif", .data = (void *)RK_SPDIF_RK3288 }, + { .compatible = "rockchip,rk3328-spdif", + .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3366-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3368-spdif", -- cgit v1.2.3 From ec2212c4af20d84841ae288a397d8ee9ecec72a0 Mon Sep 17 00:00:00 2001 From: zhangjun Date: Fri, 9 Jun 2017 16:52:48 +0800 Subject: ASoC: rockchip: i2s: add other configurable formats simple-audio-card,bitclock-inversion = <1> : bclk falling edge taken simple-audio-card,format = "dsp_a" : pcm no delay mode simple-audio-card,format = "dsp_b" : pcm late 1 mode Signed-off-by: zhangjun Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 30 ++++++++++++++++++++++++++++-- sound/soc/rockchip/rockchip_i2s.h | 3 +++ 2 files changed, 31 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 974915cb4c4f..66a26c56c658 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -204,7 +204,21 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); - mask = I2S_TXCR_IBM_MASK; + mask = I2S_CKR_CKP_MASK; + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val = I2S_CKR_CKP_NEG; + break; + case SND_SOC_DAIFMT_IB_NF: + val = I2S_CKR_CKP_POS; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); + + mask = I2S_TXCR_IBM_MASK | I2S_TXCR_TFS_MASK | I2S_TXCR_PBM_MASK; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: val = I2S_TXCR_IBM_RSJM; @@ -215,13 +229,19 @@ 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 */ + val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); + break; default: return -EINVAL; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); - mask = I2S_RXCR_IBM_MASK; + mask = I2S_RXCR_IBM_MASK | I2S_RXCR_TFS_MASK | I2S_RXCR_PBM_MASK; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: val = I2S_RXCR_IBM_RSJM; @@ -232,6 +252,12 @@ 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 */ + val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); + break; default: return -EINVAL; } diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index 31f11fd25393..a7b8527d8a73 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -41,6 +41,7 @@ #define I2S_TXCR_TFS_SHIFT 5 #define I2S_TXCR_TFS_I2S (0 << I2S_TXCR_TFS_SHIFT) #define I2S_TXCR_TFS_PCM (1 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_TFS_MASK (1 << I2S_TXCR_TFS_SHIFT) #define I2S_TXCR_VDW_SHIFT 0 #define I2S_TXCR_VDW(x) ((x - 1) << I2S_TXCR_VDW_SHIFT) #define I2S_TXCR_VDW_MASK (0x1f << I2S_TXCR_VDW_SHIFT) @@ -70,6 +71,7 @@ #define I2S_RXCR_TFS_SHIFT 5 #define I2S_RXCR_TFS_I2S (0 << I2S_RXCR_TFS_SHIFT) #define I2S_RXCR_TFS_PCM (1 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_TFS_MASK (1 << I2S_RXCR_TFS_SHIFT) #define I2S_RXCR_VDW_SHIFT 0 #define I2S_RXCR_VDW(x) ((x - 1) << I2S_RXCR_VDW_SHIFT) #define I2S_RXCR_VDW_MASK (0x1f << I2S_RXCR_VDW_SHIFT) @@ -91,6 +93,7 @@ #define I2S_CKR_CKP_SHIFT 26 #define I2S_CKR_CKP_NEG (0 << I2S_CKR_CKP_SHIFT) #define I2S_CKR_CKP_POS (1 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_CKP_MASK (1 << I2S_CKR_CKP_SHIFT) #define I2S_CKR_RLP_SHIFT 25 #define I2S_CKR_RLP_NORMAL (0 << I2S_CKR_RLP_SHIFT) #define I2S_CKR_RLP_OPPSITE (1 << I2S_CKR_RLP_SHIFT) -- cgit v1.2.3 From 891caea417469b4efdf506b6be1ef461b759c999 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 9 Jun 2017 00:43:18 +0000 Subject: ASoC: simple_card_utils: add asoc_simple_card_clk_xxx() Current simple-card-utils sets asoc_simple_dai::clk via asoc_simple_card_parse_clk(). Current simple card drivers are using it directly for clk_enable/disable. Encapsulation is one of simple card util's purpose. Let's encapsulate it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 19 ++++++++++++++++++- 2 files changed, 20 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 108cae459ed0..840d624148df 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -45,6 +45,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, const char *name); +int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai); +void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d9d8b8a58348..beb4e3817d22 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -110,6 +110,22 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); +static void asoc_simple_card_clk_register(struct asoc_simple_dai *dai, + struct clk *clk) +{ + dai->clk = clk; +} + +int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) +{ + return clk_prepare_enable(dai->clk); +} + +void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) +{ + clk_disable_unprepare(dai->clk); +} + int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, @@ -128,7 +144,8 @@ int asoc_simple_card_parse_clk(struct device *dev, clk = devm_get_clk_from_child(dev, node, NULL); if (!IS_ERR(clk)) { simple_dai->sysclk = clk_get_rate(clk); - simple_dai->clk = clk; + + asoc_simple_card_clk_register(simple_dai, clk); } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; } else { -- cgit v1.2.3 From 3ab50c4f98434080c1f73fc56d8d8b38364c6cd8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 9 Jun 2017 00:44:16 +0000 Subject: ASoC: simple-card: use asoc_simple_card_clk_xxx() Current simple-card-utils sets asoc_simple_dai::clk via asoc_simple_card_parse_clk(). Current simple card drivers are using it directly for clk_enable/disable. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_clk_enable/disable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e26bd14ba70f..8828b91867b8 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -118,13 +118,13 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) simple_priv_to_props(priv, rtd->num); int ret; - ret = clk_prepare_enable(dai_props->cpu_dai.clk); + ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); if (ret) return ret; - ret = clk_prepare_enable(dai_props->codec_dai.clk); + ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); if (ret) - clk_disable_unprepare(dai_props->cpu_dai.clk); + asoc_simple_card_clk_disable(&dai_props->cpu_dai); return ret; } @@ -136,9 +136,9 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - clk_disable_unprepare(dai_props->cpu_dai.clk); + asoc_simple_card_clk_disable(&dai_props->cpu_dai); - clk_disable_unprepare(dai_props->codec_dai.clk); + asoc_simple_card_clk_disable(&dai_props->codec_dai); } static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.3 From bb24a3ba3f52942b5f3eb6c10288da830ec9ef70 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 9 Jun 2017 00:44:40 +0000 Subject: ASoC: simple-scu-card: use asoc_simple_card_clk_xxx() Current simple-card-utils sets asoc_simple_dai::clk via asoc_simple_card_parse_clk(). Current simple card drivers are using it directly for clk_enable/disable. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_clk_enable/disable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 5faf5d6c48a2..f203783b2fad 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -47,7 +47,7 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, rtd->num); - return clk_prepare_enable(dai_props->clk); + return asoc_simple_card_clk_enable(dai_props); } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) @@ -57,7 +57,7 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, rtd->num); - clk_disable_unprepare(dai_props->clk); + asoc_simple_card_clk_disable(dai_props); } static const struct snd_soc_ops asoc_simple_card_ops = { -- cgit v1.2.3 From 6654fc77797e306a3b67b3cdf0b6121294893dba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 9 Jun 2017 00:45:01 +0000 Subject: ASoC: audio-graph-scu-card: use asoc_simple_card_clk_xxx() Current simple-card-utils sets asoc_simple_dai::clk via asoc_simple_card_parse_clk(). Current simple card drivers are using it directly for clk_enable/disable. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_clk_enable/disable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 0066102f5bc4..27a261ee7302 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -45,7 +45,7 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); - return clk_prepare_enable(dai_props->clk); + return asoc_simple_card_clk_enable(dai_props); } static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) @@ -54,7 +54,7 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); - clk_disable_unprepare(dai_props->clk); + asoc_simple_card_clk_disable(dai_props); } static struct snd_soc_ops asoc_graph_card_ops = { -- cgit v1.2.3 From d471d55934ca8b4f38535207589df4e3cc8b1484 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 9 Jun 2017 00:45:23 +0000 Subject: ASoC: audio-graph-card: use asoc_simple_card_clk_xxx() Current simple-card-utils sets asoc_simple_dai::clk via asoc_simple_card_parse_clk(). Current simple card drivers are using it directly for clk_enable/disable. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_clk_enable/disable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0180b286bee3..b5bb791a6e61 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -44,13 +44,13 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret; - ret = clk_prepare_enable(dai_props->cpu_dai.clk); + ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); if (ret) return ret; - ret = clk_prepare_enable(dai_props->codec_dai.clk); + ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); if (ret) - clk_disable_unprepare(dai_props->cpu_dai.clk); + asoc_simple_card_clk_disable(&dai_props->cpu_dai); return ret; } @@ -61,9 +61,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - clk_disable_unprepare(dai_props->cpu_dai.clk); + asoc_simple_card_clk_disable(&dai_props->cpu_dai); - clk_disable_unprepare(dai_props->codec_dai.clk); + asoc_simple_card_clk_disable(&dai_props->codec_dai); } static struct snd_soc_ops asoc_graph_card_ops = { -- cgit v1.2.3 From 63a5f59208bce7110596b09950f48bf07b8baeb9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 14 Jun 2017 01:04:11 +0000 Subject: ASoC: simple_card_utils: add EXPORT_SYMBOL_GPL() for asoc_simple_card_clk_xxx() commit 891caea41746 ("ASoC: simple_card_utils: add asoc_simple_card_clk_xxx()") added new asoc_simple_card_clk_xxx(), but, it didn't have EXPORT_SYMBOL_GPL(). This patch adds it. Otherwise, we will get below error ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-audio-graph-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-audio-graph-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-audio-graph-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-audio-graph-card.ko] undefined! Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index beb4e3817d22..2ad7633292bf 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -120,11 +120,13 @@ int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) { return clk_prepare_enable(dai->clk); } +EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { clk_disable_unprepare(dai->clk); } +EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable); int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, -- cgit v1.2.3 From a729526720059ae019803acc953f07d9c17ae234 Mon Sep 17 00:00:00 2001 From: Richard Leitner Date: Wed, 14 Jun 2017 10:36:12 +0200 Subject: ASoC: sgtl5000: add avc support The sgtl5000 features an automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening. This patch adds support for this AVC block to the driver. Apart from the "AVC Switch" control which enables the block following controls for the configuration of AVC are added: + AVC Threshold Volume: threshold where audio is compressed when the measured level is above or expanded when below + AVC Max Gain Volume: maximum gain which can be applied when the measured audio level is below threshold + AVC Hard Limiter Switch: when enabled the signal is limited to the programmed threshold. + AVC Integrator Response: response time of the integrator The AVC block is enabled and configured using the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers. Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:147: CHECK: Alignment should match open parenthesis patch:150: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 89 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 89 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702edeb77..8f6814c1eb6b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,65 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = (int)ucontrol->value.integer.value[0]; + if (db < 0 || db > 96) + return -EINVAL; + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +469,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -434,6 +513,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + /* Automatic Volume Control (DAP AVC) */ + SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ -- cgit v1.2.3 From b93d2cf8c0facb593d6f008af30ae0fcd1d49ede Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 14 Jun 2017 00:34:53 +0000 Subject: ASoC: simple-card: use asoc_simple_card_of_parse_tdm() Current simple card drivers are using asoc_simple_dai's tx_slot_mask, rx_slot_mask, slots, slot_width directly to parse TDM. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_tdm for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8828b91867b8..8b414af966ee 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -271,17 +271,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, if (ret < 0) goto dai_link_of_err; - ret = snd_soc_of_parse_tdm_slot(cpu, &cpu_dai->tx_slot_mask, - &cpu_dai->rx_slot_mask, - &cpu_dai->slots, - &cpu_dai->slot_width); + ret = asoc_simple_card_of_parse_tdm(cpu, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = snd_soc_of_parse_tdm_slot(codec, &codec_dai->tx_slot_mask, - &codec_dai->rx_slot_mask, - &codec_dai->slots, - &codec_dai->slot_width); + ret = asoc_simple_card_of_parse_tdm(codec, codec_dai); if (ret < 0) goto dai_link_of_err; -- cgit v1.2.3 From 77b713b52878fbe21d9d5339cc42fbec3202392e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 14 Jun 2017 00:35:13 +0000 Subject: ASoC: simple-scu-card: use asoc_simple_card_of_parse_tdm() Current simple card drivers are using asoc_simple_dai's tx_slot_mask, rx_slot_mask, slots, slot_width directly to parse TDM. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_tdm for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index f203783b2fad..938f3f30eef1 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -171,11 +171,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, PREFIX "prefix"); } - ret = snd_soc_of_parse_tdm_slot(np, - &dai_props->tx_slot_mask, - &dai_props->rx_slot_mask, - &dai_props->slots, - &dai_props->slot_width); + ret = asoc_simple_card_of_parse_tdm(np, dai_props); if (ret) return ret; -- cgit v1.2.3 From c98907d59594827535b492309a145ac9c758fb4c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 14 Jun 2017 00:35:30 +0000 Subject: ASoC: audio-graph-card: use asoc_simple_card_of_parse_tdm() Current simple card drivers are using asoc_simple_dai's tx_slot_mask, rx_slot_mask, slots, slot_width directly to parse TDM. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_tdm for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b5bb791a6e61..885b405d7844 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -131,19 +131,11 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, if (ret < 0) goto dai_link_of_err; - ret = snd_soc_of_parse_tdm_slot(cpu_ep, - &cpu_dai->tx_slot_mask, - &cpu_dai->rx_slot_mask, - &cpu_dai->slots, - &cpu_dai->slot_width); + ret = asoc_simple_card_of_parse_tdm(cpu_ep, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = snd_soc_of_parse_tdm_slot(codec_ep, - &codec_dai->tx_slot_mask, - &codec_dai->rx_slot_mask, - &codec_dai->slots, - &codec_dai->slot_width); + ret = asoc_simple_card_of_parse_tdm(codec_ep, codec_dai); if (ret < 0) goto dai_link_of_err; -- cgit v1.2.3 From 616c3b15f596e1f1e6c2537a1ad3492052eecba6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 14 Jun 2017 00:35:47 +0000 Subject: ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_tdm() Current simple card drivers are using asoc_simple_dai's tx_slot_mask, rx_slot_mask, slots, slot_width directly to parse TDM. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_tdm for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 27a261ee7302..4d295d07858a 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -167,11 +167,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, "prefix"); } - ret = snd_soc_of_parse_tdm_slot(ep, - &dai_props->tx_slot_mask, - &dai_props->rx_slot_mask, - &dai_props->slots, - &dai_props->slot_width); + ret = asoc_simple_card_of_parse_tdm(ep, dai_props); if (ret) return ret; -- cgit v1.2.3 From c3a3d3c41b74b05267bab6173f2a8224a1443ba6 Mon Sep 17 00:00:00 2001 From: Christophe Jaillet Date: Thu, 15 Jun 2017 07:53:11 +0200 Subject: ASoC: rockchip: Fix an error handling in 'rockchip_i2s_probe' If this memory allocation fail, we must disable what has been enabled. Do not return immediately but go thrue the error handling path instead. Also use 'devm_kmemdup' instead of 'devm_kzalloc+memcpy' to simplify code. Signed-off-by: Christophe JAILLET Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 66a26c56c658..ce09dee2202e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -641,12 +641,13 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } - soc_dai = devm_kzalloc(&pdev->dev, + soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai sizeof(*soc_dai), GFP_KERNEL); - if (!soc_dai) - return -ENOMEM; + if (!soc_dai) { + err = -ENOMEM; + goto err_pm_disable; + } - memcpy(soc_dai, &rockchip_i2s_dai, sizeof(*soc_dai)); if (!of_property_read_u32(node, "rockchip,playback-channels", &val)) { if (val >= 2 && val <= 8) soc_dai->playback.channels_max = val; -- cgit v1.2.3 From 13bb1cc0ad205b2aeeb8d2ea5c790a396135283d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:24:09 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_convert_fixup() Current simple/audio scu card drivers are supporting same convert-rate/convert-channels on DT, but doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 10 +++++++++ sound/soc/generic/simple-card-utils.c | 40 +++++++++++++++++++++++++++++++++++ 2 files changed, 50 insertions(+) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 2679312228b3..cc318ccd6a2d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -22,6 +22,11 @@ struct asoc_simple_dai { struct clk *clk; }; +struct asoc_simple_card_data { + u32 convert_rate; + u32 convert_channels; +}; + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, @@ -90,4 +95,9 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int asoc_simple_card_clean_reference(struct snd_soc_card *card); +void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, + struct snd_pcm_hw_params *params); +void asoc_simple_card_parse_convert(struct device *dev, char *prefix, + struct asoc_simple_card_data *data); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 2ad7633292bf..948a18842e64 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -13,6 +13,46 @@ #include #include +void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + if (data->convert_rate) + rate->min = + rate->max = data->convert_rate; + + if (data->convert_channels) + channels->min = + channels->max = data->convert_channels; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_convert_fixup); + +void asoc_simple_card_parse_convert(struct device *dev, char *prefix, + struct asoc_simple_card_data *data) +{ + struct device_node *np = dev->of_node; + char prop[128]; + + if (!prefix) + prefix = ""; + + /* sampling rate convert */ + snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-rate"); + of_property_read_u32(np, prop, &data->convert_rate); + + /* channels transfer */ + snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); + of_property_read_u32(np, prop, &data->convert_channels); + + dev_dbg(dev, "convert_rate %d\n", data->convert_rate); + dev_dbg(dev, "convert_channels %d\n", data->convert_channels); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert); + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, -- cgit v1.2.3 From cd8957f588397498c12b258da9044b52598c9527 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:24:28 +0000 Subject: ASoC: simple-scu-card: use asoc_simple_card_convert_fixup() Current simple/audio scu card drivers are supporting same convert-rate/convert-channels on DT, but, doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 24 +++--------------------- 1 file changed, 3 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 938f3f30eef1..44da69101097 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -27,8 +27,7 @@ struct simple_card_data { struct snd_soc_codec_conf codec_conf; struct asoc_simple_dai *dai_props; struct snd_soc_dai_link *dai_link; - u32 convert_rate; - u32 convert_channels; + struct asoc_simple_card_data adata; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) @@ -86,18 +85,8 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - if (priv->convert_rate) - rate->min = - rate->max = priv->convert_rate; - - if (priv->convert_channels) - channels->min = - channels->max = priv->convert_channels; + asoc_simple_card_convert_fixup(&priv->adata, params); return 0; } @@ -206,11 +195,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) if (ret < 0) return ret; - /* sampling rate convert */ - of_property_read_u32(node, PREFIX "convert-rate", &priv->convert_rate); - - /* channels transfer */ - of_property_read_u32(node, PREFIX "convert-channels", &priv->convert_channels); + asoc_simple_card_parse_convert(dev, PREFIX, &priv->adata); /* find 1st codec */ np = of_get_child_by_name(node, PREFIX "codec"); @@ -237,9 +222,6 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) if (ret < 0) return ret; - dev_dbg(dev, "convert_rate %d\n", priv->convert_rate); - dev_dbg(dev, "convert_channels %d\n", priv->convert_channels); - return 0; } -- cgit v1.2.3 From c564a5b18710f76da222ad9f14a4c0ebc2c4c74c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:24:43 +0000 Subject: ASoC: audio-graph-scu-card: use asoc_simple_card_convert_fixup() Current simple/audio scu card drivers are supporting same convert-rate/convert-channels on DT, but, doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 24 +++--------------------- 1 file changed, 3 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 4d295d07858a..9502f6ed14b8 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -30,8 +30,7 @@ struct graph_card_data { struct snd_soc_codec_conf codec_conf; struct asoc_simple_dai *dai_props; struct snd_soc_dai_link *dai_link; - u32 convert_rate; - u32 convert_channels; + struct asoc_simple_card_data adata; }; #define graph_priv_to_card(priv) (&(priv)->snd_card) @@ -83,18 +82,8 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - if (priv->convert_rate) - rate->min = - rate->max = priv->convert_rate; - - if (priv->convert_channels) - channels->min = - channels->max = priv->convert_channels; + asoc_simple_card_convert_fixup(&priv->adata, params); return 0; } @@ -210,11 +199,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret) return ret; - /* sampling rate convert */ - of_property_read_u32(node, "convert-rate", &priv->convert_rate); - - /* channels transfer */ - of_property_read_u32(node, "convert-channels", &priv->convert_channels); + asoc_simple_card_parse_convert(dev, NULL, &priv->adata); /* * it supports multi CPU, single CODEC only here @@ -286,9 +271,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret) goto parse_of_err; - dev_dbg(dev, "convert_rate %d\n", priv->convert_rate); - dev_dbg(dev, "convert_channels %d\n", priv->convert_channels); - ret = 0; parse_of_err: -- cgit v1.2.3 From 3296d07826ebc698113832acb426f037e9b3b253 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:25:02 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_of_parse_routing() Current simple card drivers are parsing routing on each own driver. Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_of_parse_routing for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 4 ++++ sound/soc/generic/simple-card-utils.c | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index cc318ccd6a2d..889c8ff86369 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -100,4 +100,8 @@ void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, void asoc_simple_card_parse_convert(struct device *dev, char *prefix, struct asoc_simple_card_data *data); +int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, + char *prefix, + int optional); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 948a18842e64..a2b6d95bc2f9 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -375,6 +375,28 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) } EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference); +int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, + char *prefix, + int optional) +{ + struct device_node *node = card->dev->of_node; + char prop[128]; + + if (!prefix) + prefix = ""; + + snprintf(prop, sizeof(prop), "%s%s", prefix, "routing"); + + if (!of_property_read_bool(node, prop)) { + if (optional) + return 0; + return -EINVAL; + } + + return snd_soc_of_parse_audio_routing(card, prop); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From 1fdb5d258e28de85263a34aab57f0a70b1715342 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:25:17 +0000 Subject: ASoC: simple-card: use asoc_simple_card_of_parse_routing() Current simple/audio scu card drivers are supporting same routing on DT, but, doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_routing Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8b414af966ee..7b2533c7f82e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -362,13 +362,9 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) goto card_parse_end; } - /* DAPM routes */ - if (of_property_read_bool(node, PREFIX "routing")) { - ret = snd_soc_of_parse_audio_routing(card, - PREFIX "routing"); - if (ret) - goto card_parse_end; - } + ret = asoc_simple_card_of_parse_routing(card, PREFIX, 1); + if (ret < 0) + goto card_parse_end; /* Factor to mclk, used in hw_params() */ of_property_read_u32(node, PREFIX "mclk-fs", &priv->mclk_fs); -- cgit v1.2.3 From bfe6b5898269b92571e502b4d706c815dd6bf53b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:25:33 +0000 Subject: ASoC: simple-scu-card: use asoc_simple_card_of_parse_routing() Current simple/audio scu card drivers are supporting same routing on DT, but, doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_routing Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 44da69101097..a75b385455c4 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -191,7 +191,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) if (!node) return -EINVAL; - ret = snd_soc_of_parse_audio_routing(card, PREFIX "routing"); + ret = asoc_simple_card_of_parse_routing(card, PREFIX, 0); if (ret < 0) return ret; -- cgit v1.2.3 From 9fb9b2f236f05168a6138e62c82124a2f5eaf320 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:25:51 +0000 Subject: ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_routing() Current simple/audio scu card drivers are supporting same routing on DT, but, doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_routing Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 9502f6ed14b8..05934b24627b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -195,8 +195,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * see simple-card */ - ret = snd_soc_of_parse_audio_routing(card, "routing"); - if (ret) + ret = asoc_simple_card_of_parse_routing(card, NULL, 0); + if (ret < 0) return ret; asoc_simple_card_parse_convert(dev, NULL, &priv->adata); -- cgit v1.2.3 From 33c0f552c9f3721b1e9452b1c82a37992fa90bfd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Jun 2017 20:13:33 +0100 Subject: ASoC: rockchip: Fix build Reported-by: Christophe Jaillet Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index ce09dee2202e..b4a8aff69570 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -641,10 +641,10 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } - soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai + soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai, sizeof(*soc_dai), GFP_KERNEL); if (!soc_dai) { - err = -ENOMEM; + ret = -ENOMEM; goto err_pm_disable; } -- cgit v1.2.3 From b31f11d036e689ba9e60d581ffe8e032a6305da9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 16 Jun 2017 01:38:50 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_of_parse_widgets() Current simple card drivers are parsing widgets on each own driver (only simple-card at this point, but will be supported on all drivers) Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_of_parse_widgets for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 889c8ff86369..42c6a6ac3ce6 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -103,5 +103,7 @@ void asoc_simple_card_parse_convert(struct device *dev, char *prefix, int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, char *prefix, int optional); +int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, + char *prefix); #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a2b6d95bc2f9..26d64fa40c9c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -397,6 +397,25 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing); +int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, + char *prefix) +{ + struct device_node *node = card->dev->of_node; + char prop[128]; + + if (!prefix) + prefix = ""; + + snprintf(prop, sizeof(prop), "%s%s", prefix, "widgets"); + + if (of_property_read_bool(node, prop)) + return snd_soc_of_parse_audio_simple_widgets(card, prop); + + /* no widgets is not error */ + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From fa2760dd366c735637504d1d7efab7688391c6b4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 16 Jun 2017 01:39:11 +0000 Subject: ASoC: simple-card: use asoc_simple_card_of_parse_widgets() Current simple card driver is supporting widgets on DT, other simple/audio card drivers will support it. Encapsulation is one of simple card util's purpose. Let's use asoc_simple_card_of_parse_widgets Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7b2533c7f82e..8b7b47251fe1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -354,13 +354,9 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) dai_link = of_get_child_by_name(node, PREFIX "dai-link"); - /* The off-codec widgets */ - if (of_property_read_bool(node, PREFIX "widgets")) { - ret = snd_soc_of_parse_audio_simple_widgets(card, - PREFIX "widgets"); - if (ret) - goto card_parse_end; - } + ret = asoc_simple_card_of_parse_widgets(card, PREFIX); + if (ret < 0) + goto card_parse_end; ret = asoc_simple_card_of_parse_routing(card, PREFIX, 1); if (ret < 0) -- cgit v1.2.3 From 602fdadc547f3e623db32409eeea8a59a1baaf24 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:15:30 +0200 Subject: ASoC: stm32: sai: typo fixes Fix typos in sai driver. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai.h | 1 - sound/soc/stm/stm32_sai_sub.c | 28 ++++++++++++++-------------- 3 files changed, 15 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index 2a27a26bf7a1..6159d66c2c54 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -110,6 +110,6 @@ static struct platform_driver stm32_sai_driver = { module_platform_driver(stm32_sai_driver); MODULE_DESCRIPTION("STM32 Soc SAI Interface"); -MODULE_AUTHOR("Olivier Moysan, "); +MODULE_AUTHOR("Olivier Moysan "); MODULE_ALIAS("platform:st,stm32-sai"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index a801fda5066f..270be93b845e 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -125,7 +125,6 @@ #define SAI_XFRCR_FSOFF BIT(SAI_XFRCR_FSOFF_SHIFT) /****************** Bit definition for SAI_XSLOTR register ******************/ - #define SAI_XSLOTR_FBOFF_SHIFT 0 #define SAI_XSLOTR_FBOFF_MASK GENMASK(4, SAI_XSLOTR_FBOFF_SHIFT) #define SAI_XSLOTR_FBOFF_SET(x) ((x) << SAI_XSLOTR_FBOFF_SHIFT) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ae4706ca265b..d7aeed3ec3c8 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -181,29 +181,29 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) SAI_XCLRFR_MASK); if (flags & SAI_XIMR_OVRUDRIE) { - dev_err(&pdev->dev, "IT %s\n", + dev_err(&pdev->dev, "IRQ %s\n", STM_SAI_IS_PLAYBACK(sai) ? "underrun" : "overrun"); status = SNDRV_PCM_STATE_XRUN; } if (flags & SAI_XIMR_MUTEDETIE) - dev_dbg(&pdev->dev, "IT mute detected\n"); + dev_dbg(&pdev->dev, "IRQ mute detected\n"); if (flags & SAI_XIMR_WCKCFGIE) { - dev_err(&pdev->dev, "IT wrong clock configuration\n"); + dev_err(&pdev->dev, "IRQ wrong clock configuration\n"); status = SNDRV_PCM_STATE_DISCONNECTED; } if (flags & SAI_XIMR_CNRDYIE) - dev_warn(&pdev->dev, "IT Codec not ready\n"); + dev_err(&pdev->dev, "IRQ Codec not ready\n"); if (flags & SAI_XIMR_AFSDETIE) { - dev_warn(&pdev->dev, "IT Anticipated frame synchro\n"); + dev_err(&pdev->dev, "IRQ Anticipated frame synchro\n"); status = SNDRV_PCM_STATE_XRUN; } if (flags & SAI_XIMR_LFSDETIE) { - dev_warn(&pdev->dev, "IT Late frame synchro\n"); + dev_err(&pdev->dev, "IRQ Late frame synchro\n"); status = SNDRV_PCM_STATE_XRUN; } @@ -235,7 +235,7 @@ static int stm32_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int slotr, slotr_mask, slot_size; - dev_dbg(cpu_dai->dev, "masks tx/rx:%#x/%#x, slots:%d, width:%d\n", + dev_dbg(cpu_dai->dev, "Masks tx/rx:%#x/%#x, slots:%d, width:%d\n", tx_mask, rx_mask, slots, slot_width); switch (slot_width) { @@ -377,7 +377,7 @@ static int stm32_sai_startup(struct snd_pcm_substream *substream, ret = clk_prepare_enable(sai->sai_ck); if (ret < 0) { - dev_err(cpu_dai->dev, "failed to enable clock: %d\n", ret); + dev_err(cpu_dai->dev, "Failed to enable clock: %d\n", ret); return ret; } @@ -497,7 +497,7 @@ static int stm32_sai_set_slots(struct snd_soc_dai *cpu_dai) SAI_XSLOTR_SLOTEN_SET(sai->slot_mask)); } - dev_dbg(cpu_dai->dev, "slots %d, slot width %d\n", + dev_dbg(cpu_dai->dev, "Slots %d, slot width %d\n", sai->slots, sai->slot_width); return 0; @@ -521,7 +521,7 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai) frcr |= SAI_XFRCR_FSALL_SET((fs_active - 1)); frcr_mask = SAI_XFRCR_FRL_MASK | SAI_XFRCR_FSALL_MASK; - dev_dbg(cpu_dai->dev, "frame length %d, frame active %d\n", + dev_dbg(cpu_dai->dev, "Frame length %d, frame active %d\n", sai->fs_length, fs_active); regmap_update_bits(sai->regmap, STM_SAI_FRCR_REGX, frcr_mask, frcr); @@ -784,7 +784,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sai_ck = devm_clk_get(&pdev->dev, "sai_ck"); if (IS_ERR(sai->sai_ck)) { - dev_err(&pdev->dev, "missing kernel clock sai_ck\n"); + dev_err(&pdev->dev, "Missing kernel clock sai_ck\n"); return PTR_ERR(sai->sai_ck); } @@ -849,7 +849,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, sai->pdata->irq, stm32_sai_isr, IRQF_SHARED, dev_name(&pdev->dev), sai); if (ret) { - dev_err(&pdev->dev, "irq request returned %d\n", ret); + dev_err(&pdev->dev, "IRQ request returned %d\n", ret); return ret; } @@ -861,7 +861,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, &stm32_sai_pcm_config, 0); if (ret) { - dev_err(&pdev->dev, "could not register pcm dma\n"); + dev_err(&pdev->dev, "Could not register pcm dma\n"); return ret; } @@ -879,6 +879,6 @@ static struct platform_driver stm32_sai_sub_driver = { module_platform_driver(stm32_sai_sub_driver); MODULE_DESCRIPTION("STM32 Soc SAI sub-block Interface"); -MODULE_AUTHOR("Olivier Moysan, "); +MODULE_AUTHOR("Olivier Moysan "); MODULE_ALIAS("platform:st,stm32-sai-sub"); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 607c61d40b9c29ab0902541d0d372b18793d6831 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:15:31 +0200 Subject: ASoC: stm32: sai: remove spurious trace Remove spurious trace in sai driver. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d7aeed3ec3c8..24b8874cfe5f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -761,9 +761,6 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return -ENODEV; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - - dev_err(&pdev->dev, "res %pr\n", res); - base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(base)) return PTR_ERR(base); -- cgit v1.2.3 From 4fa17938ea400b6478b24565483f2ac54122316f Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:15:32 +0200 Subject: ASoC: stm32: sai: change stop sequence Disable SAI before stopping DMA data transfers. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 24b8874cfe5f..97b69a3ab46e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -629,12 +629,12 @@ static int stm32_sai_trigger(struct snd_pcm_substream *substream, int cmd, dev_dbg(cpu_dai->dev, "Disable DMA and SAI\n"); regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_DMAEN, - (unsigned int)~SAI_XCR1_DMAEN); + SAI_XCR1_SAIEN, + (unsigned int)~SAI_XCR1_SAIEN); ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_SAIEN, - (unsigned int)~SAI_XCR1_SAIEN); + SAI_XCR1_DMAEN, + (unsigned int)~SAI_XCR1_DMAEN); if (ret < 0) dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); break; -- cgit v1.2.3 From 1c77603136d00368b8cd7c10d1ca4bad5952a136 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:15:33 +0200 Subject: ASoC: stm32: sai: fix clock management Allow peripheral clock enable/disable on regmap accesses. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 97b69a3ab46e..2466af0343db 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -766,8 +766,8 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return PTR_ERR(base); sai->phys_addr = res->start; - sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, - &stm32_sai_sub_regmap_config); + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai_ck", base, + &stm32_sai_sub_regmap_config); /* Get direction property */ if (of_property_match_string(np, "dma-names", "tx") >= 0) { -- cgit v1.2.3 From 701a6ec3a3f8d30bdb45ee025fb61f7a934b6cad Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:15:34 +0200 Subject: ASoC: stm32: sai: manage master clock Disable master clock by default, and activate it only when requested. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 2466af0343db..ce48c02db051 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -220,8 +220,15 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); + int ret; if ((dir == SND_SOC_CLOCK_OUT) && sai->master) { + ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, + SAI_XCR1_NODIV, + (unsigned int)~SAI_XCR1_NODIV); + if (ret < 0) + return ret; + sai->mclk_rate = freq; dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); } @@ -356,6 +363,10 @@ static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } cr1_mask |= SAI_XCR1_SLAVE; + /* do not generate master by default */ + cr1 |= SAI_XCR1_NODIV; + cr1_mask |= SAI_XCR1_NODIV; + ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); @@ -652,6 +663,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); + regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, + SAI_XCR1_NODIV); + clk_disable_unprepare(sai->sai_ck); sai->substream = NULL; } -- cgit v1.2.3 From 03e78a242a15eca68e5c7cb606c94959382e2b18 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Fri, 16 Jun 2017 14:16:24 +0200 Subject: ASoC: stm32: sai: add h7 support Add support of SAI on STM32H7 family. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.c | 13 +++++- sound/soc/stm/stm32_sai.h | 72 ++++++++++++++++++++++++++++++--- sound/soc/stm/stm32_sai_sub.c | 92 ++++++++++++++++++++++++++++++++++++------- 3 files changed, 155 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index 6159d66c2c54..f7713314913b 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -27,8 +27,17 @@ #include "stm32_sai.h" +static const struct stm32_sai_conf stm32_sai_conf_f4 = { + .version = SAI_STM32F4, +}; + +static const struct stm32_sai_conf stm32_sai_conf_h7 = { + .version = SAI_STM32H7, +}; + static const struct of_device_id stm32_sai_ids[] = { - { .compatible = "st,stm32f4-sai", .data = (void *)SAI_STM32F4 }, + { .compatible = "st,stm32f4-sai", .data = (void *)&stm32_sai_conf_f4 }, + { .compatible = "st,stm32h7-sai", .data = (void *)&stm32_sai_conf_h7 }, {} }; @@ -52,7 +61,7 @@ static int stm32_sai_probe(struct platform_device *pdev) of_id = of_match_device(stm32_sai_ids, &pdev->dev); if (of_id) - sai->version = (enum stm32_sai_version)of_id->data; + sai->conf = (struct stm32_sai_conf *)of_id->data; else return -EINVAL; diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index 270be93b845e..889974dc62d9 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -31,6 +31,10 @@ #define STM_SAI_CLRFR_REGX 0x18 #define STM_SAI_DR_REGX 0x1C +/* Sub-block A registers, relative to sub-block A address */ +#define STM_SAI_PDMCR_REGX 0x40 +#define STM_SAI_PDMLY_REGX 0x44 + /******************** Bit definition for SAI_GCR register *******************/ #define SAI_GCR_SYNCIN_SHIFT 0 #define SAI_GCR_SYNCIN_MASK GENMASK(1, SAI_GCR_SYNCIN_SHIFT) @@ -75,10 +79,11 @@ #define SAI_XCR1_NODIV BIT(SAI_XCR1_NODIV_SHIFT) #define SAI_XCR1_MCKDIV_SHIFT 20 -#define SAI_XCR1_MCKDIV_WIDTH 4 -#define SAI_XCR1_MCKDIV_MASK GENMASK(24, SAI_XCR1_MCKDIV_SHIFT) +#define SAI_XCR1_MCKDIV_WIDTH(x) (((x) == SAI_STM32F4) ? 4 : 6) +#define SAI_XCR1_MCKDIV_MASK(x) GENMASK((SAI_XCR1_MCKDIV_SHIFT + (x) - 1),\ + SAI_XCR1_MCKDIV_SHIFT) #define SAI_XCR1_MCKDIV_SET(x) ((x) << SAI_XCR1_MCKDIV_SHIFT) -#define SAI_XCR1_MCKDIV_MAX ((1 << SAI_XCR1_MCKDIV_WIDTH) - 1) +#define SAI_XCR1_MCKDIV_MAX(x) ((1 << SAI_XCR1_MCKDIV_WIDTH(x)) - 1) #define SAI_XCR1_OSR_SHIFT 26 #define SAI_XCR1_OSR BIT(SAI_XCR1_OSR_SHIFT) @@ -178,8 +183,65 @@ #define SAI_XCLRFR_SHIFT 0 #define SAI_XCLRFR_MASK GENMASK(6, SAI_XCLRFR_SHIFT) +/****************** Bit definition for SAI_PDMCR register ******************/ +#define SAI_PDMCR_PDMEN BIT(0) + +#define SAI_PDMCR_MICNBR_SHIFT 4 +#define SAI_PDMCR_MICNBR_MASK GENMASK(5, SAI_PDMCR_MICNBR_SHIFT) +#define SAI_PDMCR_MICNBR_SET(x) ((x) << SAI_PDMCR_MICNBR_SHIFT) + +#define SAI_PDMCR_CKEN1 BIT(8) +#define SAI_PDMCR_CKEN2 BIT(9) +#define SAI_PDMCR_CKEN3 BIT(10) +#define SAI_PDMCR_CKEN4 BIT(11) + +/****************** Bit definition for (SAI_PDMDLY register ****************/ +#define SAI_PDMDLY_1L_SHIFT 0 +#define SAI_PDMDLY_1L_MASK GENMASK(2, SAI_PDMDLY_1L_SHIFT) +#define SAI_PDMDLY_1L_WIDTH 3 + +#define SAI_PDMDLY_1R_SHIFT 4 +#define SAI_PDMDLY_1R_MASK GENMASK(6, SAI_PDMDLY_1R_SHIFT) +#define SAI_PDMDLY_1R_WIDTH 3 + +#define SAI_PDMDLY_2L_SHIFT 8 +#define SAI_PDMDLY_2L_MASK GENMASK(10, SAI_PDMDLY_2L_SHIFT) +#define SAI_PDMDLY_2L_WIDTH 3 + +#define SAI_PDMDLY_2R_SHIFT 12 +#define SAI_PDMDLY_2R_MASK GENMASK(14, SAI_PDMDLY_2R_SHIFT) +#define SAI_PDMDLY_2R_WIDTH 3 + +#define SAI_PDMDLY_3L_SHIFT 16 +#define SAI_PDMDLY_3L_MASK GENMASK(18, SAI_PDMDLY_3L_SHIFT) +#define SAI_PDMDLY_3L_WIDTH 3 + +#define SAI_PDMDLY_3R_SHIFT 20 +#define SAI_PDMDLY_3R_MASK GENMASK(22, SAI_PDMDLY_3R_SHIFT) +#define SAI_PDMDLY_3R_WIDTH 3 + +#define SAI_PDMDLY_4L_SHIFT 24 +#define SAI_PDMDLY_4L_MASK GENMASK(26, SAI_PDMDLY_4L_SHIFT) +#define SAI_PDMDLY_4L_WIDTH 3 + +#define SAI_PDMDLY_4R_SHIFT 28 +#define SAI_PDMDLY_4R_MASK GENMASK(30, SAI_PDMDLY_4R_SHIFT) +#define SAI_PDMDLY_4R_WIDTH 3 + +#define STM_SAI_IS_F4(ip) ((ip)->conf->version == SAI_STM32F4) +#define STM_SAI_IS_H7(ip) ((ip)->conf->version == SAI_STM32H7) + enum stm32_sai_version { - SAI_STM32F4 + SAI_STM32F4, + SAI_STM32H7 +}; + +/** + * struct stm32_sai_conf - SAI configuration + * @version: SAI version + */ +struct stm32_sai_conf { + int version; }; /** @@ -194,6 +256,6 @@ struct stm32_sai_data { struct platform_device *pdev; struct clk *clk_x8k; struct clk *clk_x11k; - int version; + struct stm32_sai_conf *conf; int irq; }; diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ce48c02db051..ba3fdc777ed8 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -51,12 +51,15 @@ #define STM_SAI_A_ID 0x0 #define STM_SAI_B_ID 0x1 +#define STM_SAI_IS_SUB_A(x) ((x)->id == STM_SAI_A_ID) +#define STM_SAI_IS_SUB_B(x) ((x)->id == STM_SAI_B_ID) #define STM_SAI_BLOCK_NAME(x) (((x)->id == STM_SAI_A_ID) ? "A" : "B") /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) * @pdev: device data pointer * @regmap: SAI register map pointer + * @regmap_config: SAI sub block register map configuration pointer * @dma_params: dma configuration data for rx or tx channel * @cpu_dai_drv: DAI driver data pointer * @cpu_dai: DAI runtime data pointer @@ -79,6 +82,7 @@ struct stm32_sai_sub_data { struct platform_device *pdev; struct regmap *regmap; + const struct regmap_config *regmap_config; struct snd_dmaengine_dai_dma_data dma_params; struct snd_soc_dai_driver *cpu_dai_drv; struct snd_soc_dai *cpu_dai; @@ -118,6 +122,8 @@ static bool stm32_sai_sub_readable_reg(struct device *dev, unsigned int reg) case STM_SAI_SR_REGX: case STM_SAI_CLRFR_REGX: case STM_SAI_DR_REGX: + case STM_SAI_PDMCR_REGX: + case STM_SAI_PDMLY_REGX: return true; default: return false; @@ -145,13 +151,15 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) case STM_SAI_SR_REGX: case STM_SAI_CLRFR_REGX: case STM_SAI_DR_REGX: + case STM_SAI_PDMCR_REGX: + case STM_SAI_PDMLY_REGX: return true; default: return false; } } -static const struct regmap_config stm32_sai_sub_regmap_config = { +static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -162,6 +170,17 @@ static const struct regmap_config stm32_sai_sub_regmap_config = { .fast_io = true, }; +static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = STM_SAI_PDMLY_REGX, + .readable_reg = stm32_sai_sub_readable_reg, + .volatile_reg = stm32_sai_sub_volatile_reg, + .writeable_reg = stm32_sai_sub_writeable_reg, + .fast_io = true, +}; + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -551,7 +570,8 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int cr1, mask, div = 0; - int sai_clk_rate, ret; + int sai_clk_rate, mclk_ratio, den, ret; + int version = sai->pdata->conf->version; if (!sai->mclk_rate) { dev_err(cpu_dai->dev, "Mclk rate is null\n"); @@ -564,22 +584,54 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, clk_set_parent(sai->sai_ck, sai->pdata->clk_x8k); sai_clk_rate = clk_get_rate(sai->sai_ck); - /* - * mclk_rate = 256 * fs - * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate - * MCKDIV = sai_ck / (2 * mclk_rate) otherwise - */ - if (2 * sai_clk_rate >= 3 * sai->mclk_rate) - div = DIV_ROUND_CLOSEST(sai_clk_rate, 2 * sai->mclk_rate); - - if (div > SAI_XCR1_MCKDIV_MAX) { + if (STM_SAI_IS_F4(sai->pdata)) { + /* + * mclk_rate = 256 * fs + * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate + * MCKDIV = sai_ck / (2 * mclk_rate) otherwise + */ + if (2 * sai_clk_rate >= 3 * sai->mclk_rate) + div = DIV_ROUND_CLOSEST(sai_clk_rate, + 2 * sai->mclk_rate); + } else { + /* + * TDM mode : + * mclk on + * MCKDIV = sai_ck / (ws x 256) (NOMCK=0. OSR=0) + * MCKDIV = sai_ck / (ws x 512) (NOMCK=0. OSR=1) + * mclk off + * MCKDIV = sai_ck / (frl x ws) (NOMCK=1) + * Note: NOMCK/NODIV correspond to same bit. + */ + if (sai->mclk_rate) { + mclk_ratio = sai->mclk_rate / params_rate(params); + if (mclk_ratio != 256) { + if (mclk_ratio == 512) { + mask = SAI_XCR1_OSR; + cr1 = SAI_XCR1_OSR; + } else { + dev_err(cpu_dai->dev, + "Wrong mclk ratio %d\n", + mclk_ratio); + return -EINVAL; + } + } + div = DIV_ROUND_CLOSEST(sai_clk_rate, sai->mclk_rate); + } else { + /* mclk-fs not set, master clock not active. NOMCK=1 */ + den = sai->fs_length * params_rate(params); + div = DIV_ROUND_CLOSEST(sai_clk_rate, den); + } + } + + if (div > SAI_XCR1_MCKDIV_MAX(version)) { dev_err(cpu_dai->dev, "Divider %d out of range\n", div); return -EINVAL; } dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div); - mask = SAI_XCR1_MCKDIV_MASK; - cr1 = SAI_XCR1_MCKDIV_SET(div); + mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); + cr1 = SAI_XCR1_MCKDIV_SET(div); ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); @@ -780,8 +832,18 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return PTR_ERR(base); sai->phys_addr = res->start; - sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai_ck", base, - &stm32_sai_sub_regmap_config); + + sai->regmap_config = &stm32_sai_sub_regmap_config_f4; + /* Note: PDM registers not available for H7 sub-block B */ + if (STM_SAI_IS_H7(sai->pdata) && STM_SAI_IS_SUB_A(sai)) + sai->regmap_config = &stm32_sai_sub_regmap_config_h7; + + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai_ck", + base, sai->regmap_config); + if (IS_ERR(sai->regmap)) { + dev_err(&pdev->dev, "Failed to initialize MMIO\n"); + return PTR_ERR(sai->regmap); + } /* Get direction property */ if (of_property_match_string(np, "dma-names", "tx") >= 0) { -- cgit v1.2.3 From 5561b66bd0297b029d2aba40b044ac191fcca98c Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Mon, 19 Jun 2017 11:09:55 +0200 Subject: ASoC: stm32: change configuration flag Use a specific flag for SAI and I2S interfaces, instead of common flag. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/Kconfig | 19 ++++++++++++++++--- sound/soc/stm/Makefile | 6 +++--- 2 files changed, 19 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index a6372de54042..23600a5dd46f 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -1,8 +1,21 @@ -menuconfig SND_SOC_STM32 - tristate "STMicroelectronics STM32 SOC audio support" +menu "STMicroelectronics STM32 SOC audio support" + +config SND_SOC_STM32_SAI + tristate "STM32 SAI interface (Serial Audio Interface) support" depends on ARCH_STM32 || COMPILE_TEST depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help - Say Y if you want to enable ASoC support for STM32 + Say Y if you want to enable SAI for STM32 + +config SND_SOC_STM32_I2S + tristate "STM32 I2S interface (SPI/I2S block) support" + depends on ARCH_STM32 || COMPILE_TEST + depends on SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y if you want to enable I2S for STM32 + +endmenu diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile index 82519313c0b4..4140c67fa47b 100644 --- a/sound/soc/stm/Makefile +++ b/sound/soc/stm/Makefile @@ -1,10 +1,10 @@ # SAI snd-soc-stm32-sai-sub-objs := stm32_sai_sub.o -obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai-sub.o +obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai-sub.o snd-soc-stm32-sai-objs := stm32_sai.o -obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai.o +obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai.o # I2S snd-soc-stm32-i2s-objs := stm32_i2s.o -obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-i2s.o +obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o -- cgit v1.2.3 From 372f69a01be178b896ebb8ef7021e0b165084b25 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 21 Jun 2017 04:37:18 +0000 Subject: ASoC: fsl: mpc5200_dma: remove unused psc_dma MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit linux/sound/soc/fsl/mpc5200_dma.c:305:18: warning: unused variable \ psc_dma’ [-Wunused-variable] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 0b82e209b6e3..1f7e70bfbd55 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -302,7 +302,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); size_t size = psc_dma_hardware.buffer_bytes_max; int rc; -- cgit v1.2.3 From 73d7ee2e831f106ca5c745b2cf4fdbac5a4e9e4e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 21 Jun 2017 04:38:13 +0000 Subject: ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC It doesn't use asm header. We can add COMPILE_TEST Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 823b5a236d8d..960744e46edc 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA + depends on ARCH_PXA || COMPILE_TEST select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- cgit v1.2.3 From 03e4d5d56fa5cbd47d0a8964db3722e7977723a3 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Tue, 20 Jun 2017 11:58:47 +0200 Subject: ASoC: stm32: Add SPDIFRX support Add SPDIFRX support to STM32. Signed-off-by: olivier moysan Signed-off-by: Mark Brown --- sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_spdifrx.c | 998 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1012 insertions(+) create mode 100644 sound/soc/stm/stm32_spdifrx.c (limited to 'sound') diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 23600a5dd46f..3398e6c57f37 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -18,4 +18,14 @@ config SND_SOC_STM32_I2S help Say Y if you want to enable I2S for STM32 +config SND_SOC_STM32_SPDIFRX + tristate "STM32 S/PDIF receiver (SPDIFRX) support" + depends on ARCH_STM32 || COMPILE_TEST + depends on SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + select SND_SOC_SPDIF + help + Say Y if you want to enable S/PDIF capture for STM32 + endmenu diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile index 4140c67fa47b..4ed22e648a9a 100644 --- a/sound/soc/stm/Makefile +++ b/sound/soc/stm/Makefile @@ -8,3 +8,7 @@ obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai.o # I2S snd-soc-stm32-i2s-objs := stm32_i2s.o obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o + +# SPDIFRX +snd-soc-stm32-spdifrx-objs := stm32_spdifrx.o +obj-$(CONFIG_SND_SOC_STM32_SPDIFRX) += snd-soc-stm32-spdifrx.o diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c new file mode 100644 index 000000000000..4e4250bdb75a --- /dev/null +++ b/sound/soc/stm/stm32_spdifrx.c @@ -0,0 +1,998 @@ +/* + * STM32 ALSA SoC Digital Audio Interface (SPDIF-rx) driver. + * + * Copyright (C) 2017, STMicroelectronics - All Rights Reserved + * Author(s): Olivier Moysan for STMicroelectronics. + * + * License terms: GPL V2.0. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more + * details. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +/* SPDIF-rx Register Map */ +#define STM32_SPDIFRX_CR 0x00 +#define STM32_SPDIFRX_IMR 0x04 +#define STM32_SPDIFRX_SR 0x08 +#define STM32_SPDIFRX_IFCR 0x0C +#define STM32_SPDIFRX_DR 0x10 +#define STM32_SPDIFRX_CSR 0x14 +#define STM32_SPDIFRX_DIR 0x18 + +/* Bit definition for SPDIF_CR register */ +#define SPDIFRX_CR_SPDIFEN_SHIFT 0 +#define SPDIFRX_CR_SPDIFEN_MASK GENMASK(1, SPDIFRX_CR_SPDIFEN_SHIFT) +#define SPDIFRX_CR_SPDIFENSET(x) ((x) << SPDIFRX_CR_SPDIFEN_SHIFT) + +#define SPDIFRX_CR_RXDMAEN BIT(2) +#define SPDIFRX_CR_RXSTEO BIT(3) + +#define SPDIFRX_CR_DRFMT_SHIFT 4 +#define SPDIFRX_CR_DRFMT_MASK GENMASK(5, SPDIFRX_CR_DRFMT_SHIFT) +#define SPDIFRX_CR_DRFMTSET(x) ((x) << SPDIFRX_CR_DRFMT_SHIFT) + +#define SPDIFRX_CR_PMSK BIT(6) +#define SPDIFRX_CR_VMSK BIT(7) +#define SPDIFRX_CR_CUMSK BIT(8) +#define SPDIFRX_CR_PTMSK BIT(9) +#define SPDIFRX_CR_CBDMAEN BIT(10) +#define SPDIFRX_CR_CHSEL_SHIFT 11 +#define SPDIFRX_CR_CHSEL BIT(SPDIFRX_CR_CHSEL_SHIFT) + +#define SPDIFRX_CR_NBTR_SHIFT 12 +#define SPDIFRX_CR_NBTR_MASK GENMASK(13, SPDIFRX_CR_NBTR_SHIFT) +#define SPDIFRX_CR_NBTRSET(x) ((x) << SPDIFRX_CR_NBTR_SHIFT) + +#define SPDIFRX_CR_WFA BIT(14) + +#define SPDIFRX_CR_INSEL_SHIFT 16 +#define SPDIFRX_CR_INSEL_MASK GENMASK(18, PDIFRX_CR_INSEL_SHIFT) +#define SPDIFRX_CR_INSELSET(x) ((x) << SPDIFRX_CR_INSEL_SHIFT) + +#define SPDIFRX_CR_CKSEN_SHIFT 20 +#define SPDIFRX_CR_CKSEN BIT(20) +#define SPDIFRX_CR_CKSBKPEN BIT(21) + +/* Bit definition for SPDIFRX_IMR register */ +#define SPDIFRX_IMR_RXNEI BIT(0) +#define SPDIFRX_IMR_CSRNEIE BIT(1) +#define SPDIFRX_IMR_PERRIE BIT(2) +#define SPDIFRX_IMR_OVRIE BIT(3) +#define SPDIFRX_IMR_SBLKIE BIT(4) +#define SPDIFRX_IMR_SYNCDIE BIT(5) +#define SPDIFRX_IMR_IFEIE BIT(6) + +#define SPDIFRX_XIMR_MASK GENMASK(6, 0) + +/* Bit definition for SPDIFRX_SR register */ +#define SPDIFRX_SR_RXNE BIT(0) +#define SPDIFRX_SR_CSRNE BIT(1) +#define SPDIFRX_SR_PERR BIT(2) +#define SPDIFRX_SR_OVR BIT(3) +#define SPDIFRX_SR_SBD BIT(4) +#define SPDIFRX_SR_SYNCD BIT(5) +#define SPDIFRX_SR_FERR BIT(6) +#define SPDIFRX_SR_SERR BIT(7) +#define SPDIFRX_SR_TERR BIT(8) + +#define SPDIFRX_SR_WIDTH5_SHIFT 16 +#define SPDIFRX_SR_WIDTH5_MASK GENMASK(30, PDIFRX_SR_WIDTH5_SHIFT) +#define SPDIFRX_SR_WIDTH5SET(x) ((x) << SPDIFRX_SR_WIDTH5_SHIFT) + +/* Bit definition for SPDIFRX_IFCR register */ +#define SPDIFRX_IFCR_PERRCF BIT(2) +#define SPDIFRX_IFCR_OVRCF BIT(3) +#define SPDIFRX_IFCR_SBDCF BIT(4) +#define SPDIFRX_IFCR_SYNCDCF BIT(5) + +#define SPDIFRX_XIFCR_MASK GENMASK(5, 2) + +/* Bit definition for SPDIFRX_DR register (DRFMT = 0b00) */ +#define SPDIFRX_DR0_DR_SHIFT 0 +#define SPDIFRX_DR0_DR_MASK GENMASK(23, SPDIFRX_DR0_DR_SHIFT) +#define SPDIFRX_DR0_DRSET(x) ((x) << SPDIFRX_DR0_DR_SHIFT) + +#define SPDIFRX_DR0_PE BIT(24) + +#define SPDIFRX_DR0_V BIT(25) +#define SPDIFRX_DR0_U BIT(26) +#define SPDIFRX_DR0_C BIT(27) + +#define SPDIFRX_DR0_PT_SHIFT 28 +#define SPDIFRX_DR0_PT_MASK GENMASK(29, SPDIFRX_DR0_PT_SHIFT) +#define SPDIFRX_DR0_PTSET(x) ((x) << SPDIFRX_DR0_PT_SHIFT) + +/* Bit definition for SPDIFRX_DR register (DRFMT = 0b01) */ +#define SPDIFRX_DR1_PE BIT(0) +#define SPDIFRX_DR1_V BIT(1) +#define SPDIFRX_DR1_U BIT(2) +#define SPDIFRX_DR1_C BIT(3) + +#define SPDIFRX_DR1_PT_SHIFT 4 +#define SPDIFRX_DR1_PT_MASK GENMASK(5, SPDIFRX_DR1_PT_SHIFT) +#define SPDIFRX_DR1_PTSET(x) ((x) << SPDIFRX_DR1_PT_SHIFT) + +#define SPDIFRX_DR1_DR_SHIFT 8 +#define SPDIFRX_DR1_DR_MASK GENMASK(31, SPDIFRX_DR1_DR_SHIFT) +#define SPDIFRX_DR1_DRSET(x) ((x) << SPDIFRX_DR1_DR_SHIFT) + +/* Bit definition for SPDIFRX_DR register (DRFMT = 0b10) */ +#define SPDIFRX_DR1_DRNL1_SHIFT 0 +#define SPDIFRX_DR1_DRNL1_MASK GENMASK(15, SPDIFRX_DR1_DRNL1_SHIFT) +#define SPDIFRX_DR1_DRNL1SET(x) ((x) << SPDIFRX_DR1_DRNL1_SHIFT) + +#define SPDIFRX_DR1_DRNL2_SHIFT 16 +#define SPDIFRX_DR1_DRNL2_MASK GENMASK(31, SPDIFRX_DR1_DRNL2_SHIFT) +#define SPDIFRX_DR1_DRNL2SET(x) ((x) << SPDIFRX_DR1_DRNL2_SHIFT) + +/* Bit definition for SPDIFRX_CSR register */ +#define SPDIFRX_CSR_USR_SHIFT 0 +#define SPDIFRX_CSR_USR_MASK GENMASK(15, SPDIFRX_CSR_USR_SHIFT) +#define SPDIFRX_CSR_USRGET(x) (((x) & SPDIFRX_CSR_USR_MASK)\ + >> SPDIFRX_CSR_USR_SHIFT) + +#define SPDIFRX_CSR_CS_SHIFT 16 +#define SPDIFRX_CSR_CS_MASK GENMASK(23, SPDIFRX_CSR_CS_SHIFT) +#define SPDIFRX_CSR_CSGET(x) (((x) & SPDIFRX_CSR_CS_MASK)\ + >> SPDIFRX_CSR_CS_SHIFT) + +#define SPDIFRX_CSR_SOB BIT(24) + +/* Bit definition for SPDIFRX_DIR register */ +#define SPDIFRX_DIR_THI_SHIFT 0 +#define SPDIFRX_DIR_THI_MASK GENMASK(12, SPDIFRX_DIR_THI_SHIFT) +#define SPDIFRX_DIR_THI_SET(x) ((x) << SPDIFRX_DIR_THI_SHIFT) + +#define SPDIFRX_DIR_TLO_SHIFT 16 +#define SPDIFRX_DIR_TLO_MASK GENMASK(28, SPDIFRX_DIR_TLO_SHIFT) +#define SPDIFRX_DIR_TLO_SET(x) ((x) << SPDIFRX_DIR_TLO_SHIFT) + +#define SPDIFRX_SPDIFEN_DISABLE 0x0 +#define SPDIFRX_SPDIFEN_SYNC 0x1 +#define SPDIFRX_SPDIFEN_ENABLE 0x3 + +#define SPDIFRX_IN1 0x1 +#define SPDIFRX_IN2 0x2 +#define SPDIFRX_IN3 0x3 +#define SPDIFRX_IN4 0x4 +#define SPDIFRX_IN5 0x5 +#define SPDIFRX_IN6 0x6 +#define SPDIFRX_IN7 0x7 +#define SPDIFRX_IN8 0x8 + +#define SPDIFRX_NBTR_NONE 0x0 +#define SPDIFRX_NBTR_3 0x1 +#define SPDIFRX_NBTR_15 0x2 +#define SPDIFRX_NBTR_63 0x3 + +#define SPDIFRX_DRFMT_RIGHT 0x0 +#define SPDIFRX_DRFMT_LEFT 0x1 +#define SPDIFRX_DRFMT_PACKED 0x2 + +/* 192 CS bits in S/PDIF frame. i.e 24 CS bytes */ +#define SPDIFRX_CS_BYTES_NB 24 +#define SPDIFRX_UB_BYTES_NB 48 + +/* + * CSR register is retrieved as a 32 bits word + * It contains 1 channel status byte and 2 user data bytes + * 2 S/PDIF frames are acquired to get all CS/UB bits + */ +#define SPDIFRX_CSR_BUF_LENGTH (SPDIFRX_CS_BYTES_NB * 4 * 2) + +/** + * struct stm32_spdifrx_data - private data of SPDIFRX + * @pdev: device data pointer + * @base: mmio register base virtual address + * @regmap: SPDIFRX register map pointer + * @regmap_conf: SPDIFRX register map configuration pointer + * @cs_completion: channel status retrieving completion + * @kclk: kernel clock feeding the SPDIFRX clock generator + * @dma_params: dma configuration data for rx channel + * @substream: PCM substream data pointer + * @dmab: dma buffer info pointer + * @ctrl_chan: dma channel for S/PDIF control bits + * @desc:dma async transaction descriptor + * @slave_config: dma slave channel runtime config pointer + * @phys_addr: SPDIFRX registers physical base address + * @lock: synchronization enabling lock + * @cs: channel status buffer + * @ub: user data buffer + * @irq: SPDIFRX interrupt line + * @refcount: keep count of opened DMA channels + */ +struct stm32_spdifrx_data { + struct platform_device *pdev; + void __iomem *base; + struct regmap *regmap; + const struct regmap_config *regmap_conf; + struct completion cs_completion; + struct clk *kclk; + struct snd_dmaengine_dai_dma_data dma_params; + struct snd_pcm_substream *substream; + struct snd_dma_buffer *dmab; + struct dma_chan *ctrl_chan; + struct dma_async_tx_descriptor *desc; + struct dma_slave_config slave_config; + dma_addr_t phys_addr; + spinlock_t lock; /* Sync enabling lock */ + unsigned char cs[SPDIFRX_CS_BYTES_NB]; + unsigned char ub[SPDIFRX_UB_BYTES_NB]; + int irq; + int refcount; +}; + +static void stm32_spdifrx_dma_complete(void *data) +{ + struct stm32_spdifrx_data *spdifrx = (struct stm32_spdifrx_data *)data; + struct platform_device *pdev = spdifrx->pdev; + u32 *p_start = (u32 *)spdifrx->dmab->area; + u32 *p_end = p_start + (2 * SPDIFRX_CS_BYTES_NB) - 1; + u32 *ptr = p_start; + u16 *ub_ptr = (short *)spdifrx->ub; + int i = 0; + + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_CBDMAEN, + (unsigned int)~SPDIFRX_CR_CBDMAEN); + + if (!spdifrx->dmab->area) + return; + + while (ptr <= p_end) { + if (*ptr & SPDIFRX_CSR_SOB) + break; + ptr++; + } + + if (ptr > p_end) { + dev_err(&pdev->dev, "Start of S/PDIF block not found\n"); + return; + } + + while (i < SPDIFRX_CS_BYTES_NB) { + spdifrx->cs[i] = (unsigned char)SPDIFRX_CSR_CSGET(*ptr); + *ub_ptr++ = SPDIFRX_CSR_USRGET(*ptr++); + if (ptr > p_end) { + dev_err(&pdev->dev, "Failed to get channel status\n"); + return; + } + i++; + } + + complete(&spdifrx->cs_completion); +} + +static int stm32_spdifrx_dma_ctrl_start(struct stm32_spdifrx_data *spdifrx) +{ + dma_cookie_t cookie; + int err; + + spdifrx->desc = dmaengine_prep_slave_single(spdifrx->ctrl_chan, + spdifrx->dmab->addr, + SPDIFRX_CSR_BUF_LENGTH, + DMA_DEV_TO_MEM, + DMA_CTRL_ACK); + if (!spdifrx->desc) + return -EINVAL; + + spdifrx->desc->callback = stm32_spdifrx_dma_complete; + spdifrx->desc->callback_param = spdifrx; + cookie = dmaengine_submit(spdifrx->desc); + err = dma_submit_error(cookie); + if (err) + return -EINVAL; + + dma_async_issue_pending(spdifrx->ctrl_chan); + + return 0; +} + +static void stm32_spdifrx_dma_ctrl_stop(struct stm32_spdifrx_data *spdifrx) +{ + dmaengine_terminate_async(spdifrx->ctrl_chan); +} + +static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) +{ + int cr, cr_mask, imr, ret; + + /* Enable IRQs */ + imr = SPDIFRX_IMR_IFEIE | SPDIFRX_IMR_SYNCDIE | SPDIFRX_IMR_PERRIE; + ret = regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_IMR, imr, imr); + if (ret) + return ret; + + spin_lock(&spdifrx->lock); + + spdifrx->refcount++; + + regmap_read(spdifrx->regmap, STM32_SPDIFRX_CR, &cr); + + if (!(cr & SPDIFRX_CR_SPDIFEN_MASK)) { + /* + * Start sync if SPDIFRX is still in idle state. + * SPDIFRX reception enabled when sync done + */ + dev_dbg(&spdifrx->pdev->dev, "start synchronization\n"); + + /* + * SPDIFRX configuration: + * Wait for activity before starting sync process. This avoid + * to issue sync errors when spdif signal is missing on input. + * Preamble, CS, user, validity and parity error bits not copied + * to DR register. + */ + cr = SPDIFRX_CR_WFA | SPDIFRX_CR_PMSK | SPDIFRX_CR_VMSK | + SPDIFRX_CR_CUMSK | SPDIFRX_CR_PTMSK | SPDIFRX_CR_RXSTEO; + cr_mask = cr; + + cr |= SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); + cr_mask |= SPDIFRX_CR_SPDIFEN_MASK; + ret = regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + cr_mask, cr); + if (ret < 0) + dev_err(&spdifrx->pdev->dev, + "Failed to start synchronization\n"); + } + + spin_unlock(&spdifrx->lock); + + return ret; +} + +static void stm32_spdifrx_stop(struct stm32_spdifrx_data *spdifrx) +{ + int cr, cr_mask, reg; + + spin_lock(&spdifrx->lock); + + if (--spdifrx->refcount) { + spin_unlock(&spdifrx->lock); + return; + } + + cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_DISABLE); + cr_mask = SPDIFRX_CR_SPDIFEN_MASK | SPDIFRX_CR_RXDMAEN; + + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, cr_mask, cr); + + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_IMR, + SPDIFRX_XIMR_MASK, 0); + + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_IFCR, + SPDIFRX_XIFCR_MASK, SPDIFRX_XIFCR_MASK); + + /* dummy read to clear CSRNE and RXNE in status register */ + regmap_read(spdifrx->regmap, STM32_SPDIFRX_DR, ®); + regmap_read(spdifrx->regmap, STM32_SPDIFRX_CSR, ®); + + spin_unlock(&spdifrx->lock); +} + +static int stm32_spdifrx_dma_ctrl_register(struct device *dev, + struct stm32_spdifrx_data *spdifrx) +{ + int ret; + + spdifrx->dmab = devm_kzalloc(dev, sizeof(struct snd_dma_buffer), + GFP_KERNEL); + if (!spdifrx->dmab) + return -ENOMEM; + + spdifrx->dmab->dev.type = SNDRV_DMA_TYPE_DEV_IRAM; + spdifrx->dmab->dev.dev = dev; + ret = snd_dma_alloc_pages(spdifrx->dmab->dev.type, dev, + SPDIFRX_CSR_BUF_LENGTH, spdifrx->dmab); + if (ret < 0) { + dev_err(dev, "snd_dma_alloc_pages returned error %d\n", ret); + return ret; + } + + spdifrx->ctrl_chan = dma_request_chan(dev, "rx-ctrl"); + if (!spdifrx->ctrl_chan) { + dev_err(dev, "dma_request_slave_channel failed\n"); + return -EINVAL; + } + + spdifrx->slave_config.direction = DMA_DEV_TO_MEM; + spdifrx->slave_config.src_addr = (dma_addr_t)(spdifrx->phys_addr + + STM32_SPDIFRX_CSR); + spdifrx->slave_config.dst_addr = spdifrx->dmab->addr; + spdifrx->slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdifrx->slave_config.src_maxburst = 1; + + ret = dmaengine_slave_config(spdifrx->ctrl_chan, + &spdifrx->slave_config); + if (ret < 0) { + dev_err(dev, "dmaengine_slave_config returned error %d\n", ret); + dma_release_channel(spdifrx->ctrl_chan); + spdifrx->ctrl_chan = NULL; + } + + return ret; +}; + +static const char * const spdifrx_enum_input[] = { + "in0", "in1", "in2", "in3" +}; + +/* By default CS bits are retrieved from channel A */ +static const char * const spdifrx_enum_cs_channel[] = { + "A", "B" +}; + +static SOC_ENUM_SINGLE_DECL(ctrl_enum_input, + STM32_SPDIFRX_CR, SPDIFRX_CR_INSEL_SHIFT, + spdifrx_enum_input); + +static SOC_ENUM_SINGLE_DECL(ctrl_enum_cs_channel, + STM32_SPDIFRX_CR, SPDIFRX_CR_CHSEL_SHIFT, + spdifrx_enum_cs_channel); + +static int stm32_spdifrx_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 stm32_spdifrx_ub_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 stm32_spdifrx_get_ctrl_data(struct stm32_spdifrx_data *spdifrx) +{ + int ret = 0; + + memset(spdifrx->cs, 0, SPDIFRX_CS_BYTES_NB); + memset(spdifrx->ub, 0, SPDIFRX_UB_BYTES_NB); + + ret = stm32_spdifrx_dma_ctrl_start(spdifrx); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(spdifrx->kclk); + if (ret) { + dev_err(&spdifrx->pdev->dev, "Enable kclk failed: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_CBDMAEN, SPDIFRX_CR_CBDMAEN); + if (ret < 0) + goto end; + + ret = stm32_spdifrx_start_sync(spdifrx); + if (ret < 0) + goto end; + + if (wait_for_completion_interruptible_timeout(&spdifrx->cs_completion, + msecs_to_jiffies(100)) + <= 0) { + dev_err(&spdifrx->pdev->dev, "Failed to get control data\n"); + ret = -EAGAIN; + } + + stm32_spdifrx_stop(spdifrx); + stm32_spdifrx_dma_ctrl_stop(spdifrx); + +end: + clk_disable_unprepare(spdifrx->kclk); + + return ret; +} + +static int stm32_spdifrx_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + + stm32_spdifrx_get_ctrl_data(spdifrx); + + ucontrol->value.iec958.status[0] = spdifrx->cs[0]; + ucontrol->value.iec958.status[1] = spdifrx->cs[1]; + ucontrol->value.iec958.status[2] = spdifrx->cs[2]; + ucontrol->value.iec958.status[3] = spdifrx->cs[3]; + ucontrol->value.iec958.status[4] = spdifrx->cs[4]; + + return 0; +} + +static int stm32_spdif_user_bits_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + + stm32_spdifrx_get_ctrl_data(spdifrx); + + ucontrol->value.iec958.status[0] = spdifrx->ub[0]; + ucontrol->value.iec958.status[1] = spdifrx->ub[1]; + ucontrol->value.iec958.status[2] = spdifrx->ub[2]; + ucontrol->value.iec958.status[3] = spdifrx->ub[3]; + ucontrol->value.iec958.status[4] = spdifrx->ub[4]; + + return 0; +} + +static struct snd_kcontrol_new stm32_spdifrx_iec_ctrls[] = { + /* Channel status control */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = stm32_spdifrx_info, + .get = stm32_spdifrx_capture_get, + }, + /* User bits control */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 User Bit Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = stm32_spdifrx_ub_info, + .get = stm32_spdif_user_bits_get, + }, +}; + +static struct snd_kcontrol_new stm32_spdifrx_ctrls[] = { + SOC_ENUM("SPDIFRX input", ctrl_enum_input), + SOC_ENUM("SPDIFRX CS channel", ctrl_enum_cs_channel), +}; + +static int stm32_spdifrx_dai_register_ctrls(struct snd_soc_dai *cpu_dai) +{ + int ret; + + ret = snd_soc_add_dai_controls(cpu_dai, stm32_spdifrx_iec_ctrls, + ARRAY_SIZE(stm32_spdifrx_iec_ctrls)); + if (ret < 0) + return ret; + + return snd_soc_add_component_controls(cpu_dai->component, + stm32_spdifrx_ctrls, + ARRAY_SIZE(stm32_spdifrx_ctrls)); +} + +static int stm32_spdifrx_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct stm32_spdifrx_data *spdifrx = dev_get_drvdata(cpu_dai->dev); + + spdifrx->dma_params.addr = (dma_addr_t)(spdifrx->phys_addr + + STM32_SPDIFRX_DR); + spdifrx->dma_params.maxburst = 1; + + snd_soc_dai_init_dma_data(cpu_dai, NULL, &spdifrx->dma_params); + + return stm32_spdifrx_dai_register_ctrls(cpu_dai); +} + +static bool stm32_spdifrx_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case STM32_SPDIFRX_CR: + case STM32_SPDIFRX_IMR: + case STM32_SPDIFRX_SR: + case STM32_SPDIFRX_IFCR: + case STM32_SPDIFRX_DR: + case STM32_SPDIFRX_CSR: + case STM32_SPDIFRX_DIR: + return true; + default: + return false; + } +} + +static bool stm32_spdifrx_volatile_reg(struct device *dev, unsigned int reg) +{ + if (reg == STM32_SPDIFRX_DR) + return true; + + return false; +} + +static bool stm32_spdifrx_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case STM32_SPDIFRX_CR: + case STM32_SPDIFRX_IMR: + case STM32_SPDIFRX_IFCR: + return true; + default: + return false; + } +} + +static const struct regmap_config stm32_h7_spdifrx_regmap_conf = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = STM32_SPDIFRX_DIR, + .readable_reg = stm32_spdifrx_readable_reg, + .volatile_reg = stm32_spdifrx_volatile_reg, + .writeable_reg = stm32_spdifrx_writeable_reg, + .fast_io = true, +}; + +static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) +{ + struct stm32_spdifrx_data *spdifrx = (struct stm32_spdifrx_data *)devid; + struct snd_pcm_substream *substream = spdifrx->substream; + struct platform_device *pdev = spdifrx->pdev; + unsigned int cr, mask, sr, imr; + unsigned int flags; + int err = 0, err_xrun = 0; + + regmap_read(spdifrx->regmap, STM32_SPDIFRX_SR, &sr); + regmap_read(spdifrx->regmap, STM32_SPDIFRX_IMR, &imr); + + mask = imr & SPDIFRX_XIMR_MASK; + /* SERR, TERR, FERR IRQs are generated if IFEIE is set */ + if (mask & SPDIFRX_IMR_IFEIE) + mask |= (SPDIFRX_IMR_IFEIE << 1) | (SPDIFRX_IMR_IFEIE << 2); + + flags = sr & mask; + if (!flags) { + dev_err(&pdev->dev, "Unexpected IRQ. rflags=%#x, imr=%#x\n", + sr, imr); + return IRQ_NONE; + } + + /* Clear IRQs */ + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_IFCR, + SPDIFRX_XIFCR_MASK, flags); + + if (flags & SPDIFRX_SR_PERR) { + dev_dbg(&pdev->dev, "Parity error\n"); + err_xrun = 1; + } + + if (flags & SPDIFRX_SR_OVR) { + dev_dbg(&pdev->dev, "Overrun error\n"); + err_xrun = 1; + } + + if (flags & SPDIFRX_SR_SBD) + dev_dbg(&pdev->dev, "Synchronization block detected\n"); + + if (flags & SPDIFRX_SR_SYNCD) { + dev_dbg(&pdev->dev, "Synchronization done\n"); + + /* Enable spdifrx */ + cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_ENABLE); + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_SPDIFEN_MASK, cr); + } + + if (flags & SPDIFRX_SR_FERR) { + dev_dbg(&pdev->dev, "Frame error\n"); + err = 1; + } + + if (flags & SPDIFRX_SR_SERR) { + dev_dbg(&pdev->dev, "Synchronization error\n"); + err = 1; + } + + if (flags & SPDIFRX_SR_TERR) { + dev_dbg(&pdev->dev, "Timeout error\n"); + err = 1; + } + + if (err) { + /* SPDIFRX in STATE_STOP. Disable SPDIFRX to clear errors */ + cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_DISABLE); + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_SPDIFEN_MASK, cr); + + if (substream) + snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); + + return IRQ_HANDLED; + } + + if (err_xrun && substream) + snd_pcm_stop_xrun(substream); + + return IRQ_HANDLED; +} + +static int stm32_spdifrx_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + int ret; + + spdifrx->substream = substream; + + ret = clk_prepare_enable(spdifrx->kclk); + if (ret) + dev_err(&spdifrx->pdev->dev, "Enable kclk failed: %d\n", ret); + + return ret; +} + +static int stm32_spdifrx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + int data_size = params_width(params); + int fmt; + + switch (data_size) { + case 16: + fmt = SPDIFRX_DRFMT_PACKED; + spdifrx->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + fmt = SPDIFRX_DRFMT_LEFT; + spdifrx->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(&spdifrx->pdev->dev, "Unexpected data format\n"); + return -EINVAL; + } + + snd_soc_dai_init_dma_data(cpu_dai, NULL, &spdifrx->dma_params); + + return regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_DRFMT_MASK, + SPDIFRX_CR_DRFMTSET(fmt)); +} + +static int stm32_spdifrx_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_IMR, + SPDIFRX_IMR_OVRIE, SPDIFRX_IMR_OVRIE); + + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_RXDMAEN, SPDIFRX_CR_RXDMAEN); + + ret = stm32_spdifrx_start_sync(spdifrx); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + stm32_spdifrx_stop(spdifrx); + break; + default: + return -EINVAL; + } + + return ret; +} + +static void stm32_spdifrx_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + + spdifrx->substream = NULL; + clk_disable_unprepare(spdifrx->kclk); +} + +static const struct snd_soc_dai_ops stm32_spdifrx_pcm_dai_ops = { + .startup = stm32_spdifrx_startup, + .hw_params = stm32_spdifrx_hw_params, + .trigger = stm32_spdifrx_trigger, + .shutdown = stm32_spdifrx_shutdown, +}; + +static struct snd_soc_dai_driver stm32_spdifrx_dai[] = { + { + .name = "spdifrx-capture-cpu-dai", + .probe = stm32_spdifrx_dai_probe, + .capture = { + .stream_name = "CPU-Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &stm32_spdifrx_pcm_dai_ops, + } +}; + +static const struct snd_pcm_hardware stm32_spdifrx_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP, + .buffer_bytes_max = 8 * PAGE_SIZE, + .period_bytes_max = 2048, /* MDMA constraint */ + .periods_min = 2, + .periods_max = 8, +}; + +static const struct snd_soc_component_driver stm32_spdifrx_component = { + .name = "stm32-spdifrx", +}; + +static const struct snd_dmaengine_pcm_config stm32_spdifrx_pcm_config = { + .pcm_hardware = &stm32_spdifrx_pcm_hw, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + +static const struct of_device_id stm32_spdifrx_ids[] = { + { + .compatible = "st,stm32h7-spdifrx", + .data = &stm32_h7_spdifrx_regmap_conf + }, + {} +}; + +static int stm_spdifrx_parse_of(struct platform_device *pdev, + struct stm32_spdifrx_data *spdifrx) +{ + struct device_node *np = pdev->dev.of_node; + const struct of_device_id *of_id; + struct resource *res; + + if (!np) + return -ENODEV; + + of_id = of_match_device(stm32_spdifrx_ids, &pdev->dev); + if (of_id) + spdifrx->regmap_conf = + (const struct regmap_config *)of_id->data; + else + return -EINVAL; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + spdifrx->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(spdifrx->base)) + return PTR_ERR(spdifrx->base); + + spdifrx->phys_addr = res->start; + + spdifrx->kclk = devm_clk_get(&pdev->dev, "kclk"); + if (IS_ERR(spdifrx->kclk)) { + dev_err(&pdev->dev, "Could not get kclk\n"); + return PTR_ERR(spdifrx->kclk); + } + + spdifrx->irq = platform_get_irq(pdev, 0); + if (spdifrx->irq < 0) { + dev_err(&pdev->dev, "No irq for node %s\n", pdev->name); + return spdifrx->irq; + } + + return 0; +} + +static int stm32_spdifrx_probe(struct platform_device *pdev) +{ + struct stm32_spdifrx_data *spdifrx; + struct reset_control *rst; + const struct snd_dmaengine_pcm_config *pcm_config = NULL; + int ret; + + spdifrx = devm_kzalloc(&pdev->dev, sizeof(*spdifrx), GFP_KERNEL); + if (!spdifrx) + return -ENOMEM; + + spdifrx->pdev = pdev; + init_completion(&spdifrx->cs_completion); + spin_lock_init(&spdifrx->lock); + + platform_set_drvdata(pdev, spdifrx); + + ret = stm_spdifrx_parse_of(pdev, spdifrx); + if (ret) + return ret; + + spdifrx->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "kclk", + spdifrx->base, + spdifrx->regmap_conf); + if (IS_ERR(spdifrx->regmap)) { + dev_err(&pdev->dev, "Regmap init failed\n"); + return PTR_ERR(spdifrx->regmap); + } + + ret = devm_request_irq(&pdev->dev, spdifrx->irq, stm32_spdifrx_isr, 0, + dev_name(&pdev->dev), spdifrx); + if (ret) { + dev_err(&pdev->dev, "IRQ request returned %d\n", ret); + return ret; + } + + rst = devm_reset_control_get(&pdev->dev, NULL); + if (!IS_ERR(rst)) { + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &stm32_spdifrx_component, + stm32_spdifrx_dai, + ARRAY_SIZE(stm32_spdifrx_dai)); + if (ret) + return ret; + + ret = stm32_spdifrx_dma_ctrl_register(&pdev->dev, spdifrx); + if (ret) + goto error; + + pcm_config = &stm32_spdifrx_pcm_config; + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0); + if (ret) { + dev_err(&pdev->dev, "PCM DMA register returned %d\n", ret); + goto error; + } + + return 0; + +error: + if (spdifrx->ctrl_chan) + dma_release_channel(spdifrx->ctrl_chan); + if (spdifrx->dmab) + snd_dma_free_pages(spdifrx->dmab); + + return ret; +} + +static int stm32_spdifrx_remove(struct platform_device *pdev) +{ + struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); + + if (spdifrx->ctrl_chan) + dma_release_channel(spdifrx->ctrl_chan); + + if (spdifrx->dmab) + snd_dma_free_pages(spdifrx->dmab); + + return 0; +} + +MODULE_DEVICE_TABLE(of, stm32_spdifrx_ids); + +static struct platform_driver stm32_spdifrx_driver = { + .driver = { + .name = "st,stm32-spdifrx", + .of_match_table = stm32_spdifrx_ids, + }, + .probe = stm32_spdifrx_probe, + .remove = stm32_spdifrx_remove, +}; + +module_platform_driver(stm32_spdifrx_driver); + +MODULE_DESCRIPTION("STM32 Soc spdifrx Interface"); +MODULE_AUTHOR("Olivier Moysan, "); +MODULE_ALIAS("platform:stm32-spdifrx"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 1943b0661184a5d17f31624dc8ac2c02a086c998 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 22 Jun 2017 10:32:31 +0100 Subject: ASoC: max9867: make array ni_div static const The array ni_div does not need to be in global scope and is not modified, so make it static const. Cleans up sparse warning: "symbol 'ni_div' was not declared. Should it be static?" Signed-off-by: Colin Ian King Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/max9867.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 0247edc9c84e..2a40a69a7513 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -132,7 +132,7 @@ enum rates { pcm_rate_48, max_pcm_rate, }; -struct ni_div_rates { +static const struct ni_div_rates { u32 mclk; u16 ni[max_pcm_rate]; } ni_div[] = { -- cgit v1.2.3 From 0e15bdfd8b1e3a94862522580161a2d1bb3882a7 Mon Sep 17 00:00:00 2001 From: Baoyou Xie Date: Thu, 22 Jun 2017 14:51:58 +0800 Subject: ASoC: zx_aud96p22: add ZTE ZX AUD96P22 codec driver It adds ASoC driver for AUD96P22 stereo audio codec integrated on ZTE ZX family SoCs. The driver includes the support for a number of volume and mute controls, and power bits for various playback and recording components. Due to that the board for testing only supports playback, recording support is untested. Signed-off-by: Baoyou Xie Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/zx_aud96p22.c | 403 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 410 insertions(+) create mode 100644 sound/soc/codecs/zx_aud96p22.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..3425bbcea2d1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1114,6 +1114,11 @@ config SND_SOC_WM9713 tristate select REGMAP_AC97 +config SND_SOC_ZX_AUD96P22 + tristate "ZTE ZX AUD96P22 CODEC" + depends on I2C + select REGMAP_I2C + # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..d9858be7796a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -224,6 +224,7 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o +snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o @@ -455,6 +456,7 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c new file mode 100644 index 000000000000..032fb7cf6cbd --- /dev/null +++ b/sound/soc/codecs/zx_aud96p22.c @@ -0,0 +1,403 @@ +/* + * Copyright (C) 2017 Sanechips Technology Co., Ltd. + * Copyright 2017 Linaro Ltd. + * + * Author: Baoyou Xie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define AUD96P22_RESET 0x00 +#define RST_DAC_DPZ BIT(0) +#define RST_ADC_DPZ BIT(1) +#define AUD96P22_I2S1_CONFIG_0 0x03 +#define I2S1_MS_MODE BIT(3) +#define I2S1_MODE_MASK 0x7 +#define I2S1_MODE_RIGHT_J 0x0 +#define I2S1_MODE_I2S 0x1 +#define I2S1_MODE_LEFT_J 0x2 +#define AUD96P22_PD_0 0x15 +#define AUD96P22_PD_1 0x16 +#define AUD96P22_PD_3 0x18 +#define AUD96P22_PD_4 0x19 +#define AUD96P22_MUTE_0 0x1d +#define AUD96P22_MUTE_2 0x1f +#define AUD96P22_MUTE_4 0x21 +#define AUD96P22_RECVOL_0 0x24 +#define AUD96P22_RECVOL_1 0x25 +#define AUD96P22_PGA1VOL_0 0x26 +#define AUD96P22_PGA1VOL_1 0x27 +#define AUD96P22_LMVOL_0 0x34 +#define AUD96P22_LMVOL_1 0x35 +#define AUD96P22_HS1VOL_0 0x38 +#define AUD96P22_HS1VOL_1 0x39 +#define AUD96P22_PGA1SEL_0 0x47 +#define AUD96P22_PGA1SEL_1 0x48 +#define AUD96P22_LDR1SEL_0 0x59 +#define AUD96P22_LDR1SEL_1 0x60 +#define AUD96P22_LDR2SEL_0 0x5d +#define AUD96P22_REG_MAX 0xfb + +struct aud96p22_priv { + struct regmap *regmap; +}; + +static int aud96p22_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct aud96p22_priv *priv = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = priv->regmap; + + if (event != SND_SOC_DAPM_POST_PMU) + return -EINVAL; + + /* Assert/de-assert the bit to reset ADC data path */ + regmap_update_bits(regmap, AUD96P22_RESET, RST_ADC_DPZ, 0); + regmap_update_bits(regmap, AUD96P22_RESET, RST_ADC_DPZ, RST_ADC_DPZ); + + return 0; +} + +static int aud96p22_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct aud96p22_priv *priv = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = priv->regmap; + + if (event != SND_SOC_DAPM_POST_PMU) + return -EINVAL; + + /* Assert/de-assert the bit to reset DAC data path */ + regmap_update_bits(regmap, AUD96P22_RESET, RST_DAC_DPZ, 0); + regmap_update_bits(regmap, AUD96P22_RESET, RST_DAC_DPZ, RST_DAC_DPZ); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(lm_tlv, -11550, 50, 0); +static const DECLARE_TLV_DB_SCALE(hs_tlv, -3900, 300, 0); +static const DECLARE_TLV_DB_SCALE(rec_tlv, -9550, 50, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1800, 100, 0); + +static const struct snd_kcontrol_new aud96p22_snd_controls[] = { + /* Volume control */ + SOC_DOUBLE_R_TLV("Master Playback Volume", AUD96P22_LMVOL_0, + AUD96P22_LMVOL_1, 0, 0xff, 0, lm_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", AUD96P22_HS1VOL_0, + AUD96P22_HS1VOL_1, 0, 0xf, 0, hs_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", AUD96P22_RECVOL_0, + AUD96P22_RECVOL_1, 0, 0xff, 0, rec_tlv), + SOC_DOUBLE_R_TLV("Analogue Capture Volume", AUD96P22_PGA1VOL_0, + AUD96P22_PGA1VOL_1, 0, 0x37, 0, pga_tlv), + + /* Mute control */ + SOC_DOUBLE("Master Playback Switch", AUD96P22_MUTE_2, 0, 1, 1, 1), + SOC_DOUBLE("Headphone Switch", AUD96P22_MUTE_2, 4, 5, 1, 1), + SOC_DOUBLE("Line Out Switch", AUD96P22_MUTE_4, 0, 1, 1, 1), + SOC_DOUBLE("Speaker Switch", AUD96P22_MUTE_4, 2, 3, 1, 1), + SOC_DOUBLE("Master Capture Switch", AUD96P22_MUTE_0, 0, 1, 1, 1), + SOC_DOUBLE("Analogue Capture Switch", AUD96P22_MUTE_0, 2, 3, 1, 1), +}; + +/* Input mux kcontrols */ +static const unsigned int ain_mux_values[] = { + 0, 1, 3, 4, 5, +}; + +static const char * const ainl_mux_texts[] = { + "AINL1 differential", + "AINL1 single-ended", + "AINL3 single-ended", + "AINL2 differential", + "AINL2 single-ended", +}; + +static const char * const ainr_mux_texts[] = { + "AINR1 differential", + "AINR1 single-ended", + "AINR3 single-ended", + "AINR2 differential", + "AINR2 single-ended", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(ainl_mux_enum, AUD96P22_PGA1SEL_0, + 0, 0x7, ainl_mux_texts, ain_mux_values); +static SOC_VALUE_ENUM_SINGLE_DECL(ainr_mux_enum, AUD96P22_PGA1SEL_1, + 0, 0x7, ainr_mux_texts, ain_mux_values); + +static const struct snd_kcontrol_new ainl_mux_kcontrol = + SOC_DAPM_ENUM("AINL Mux", ainl_mux_enum); +static const struct snd_kcontrol_new ainr_mux_kcontrol = + SOC_DAPM_ENUM("AINR Mux", ainr_mux_enum); + +/* Output mixer kcontrols */ +static const struct snd_kcontrol_new ld1_left_kcontrols[] = { + SOC_DAPM_SINGLE("DACL LD1L Switch", AUD96P22_LDR1SEL_0, 0, 1, 0), + SOC_DAPM_SINGLE("AINL LD1L Switch", AUD96P22_LDR1SEL_0, 1, 1, 0), + SOC_DAPM_SINGLE("AINR LD1L Switch", AUD96P22_LDR1SEL_0, 2, 1, 0), +}; + +static const struct snd_kcontrol_new ld1_right_kcontrols[] = { + SOC_DAPM_SINGLE("DACR LD1R Switch", AUD96P22_LDR1SEL_1, 8, 1, 0), + SOC_DAPM_SINGLE("AINR LD1R Switch", AUD96P22_LDR1SEL_1, 9, 1, 0), + SOC_DAPM_SINGLE("AINL LD1R Switch", AUD96P22_LDR1SEL_1, 10, 1, 0), +}; + +static const struct snd_kcontrol_new ld2_kcontrols[] = { + SOC_DAPM_SINGLE("DACL LD2 Switch", AUD96P22_LDR2SEL_0, 0, 1, 0), + SOC_DAPM_SINGLE("AINL LD2 Switch", AUD96P22_LDR2SEL_0, 1, 1, 0), + SOC_DAPM_SINGLE("DACR LD2 Switch", AUD96P22_LDR2SEL_0, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aud96p22_dapm_widgets[] = { + /* Overall power bit */ + SND_SOC_DAPM_SUPPLY("POWER", AUD96P22_PD_0, 0, 0, NULL, 0), + + /* Input pins */ + SND_SOC_DAPM_INPUT("AINL1P"), + SND_SOC_DAPM_INPUT("AINL2P"), + SND_SOC_DAPM_INPUT("AINL3"), + SND_SOC_DAPM_INPUT("AINL1N"), + SND_SOC_DAPM_INPUT("AINL2N"), + SND_SOC_DAPM_INPUT("AINR2N"), + SND_SOC_DAPM_INPUT("AINR1N"), + SND_SOC_DAPM_INPUT("AINR3"), + SND_SOC_DAPM_INPUT("AINR2P"), + SND_SOC_DAPM_INPUT("AINR1P"), + + /* Input muxes */ + SND_SOC_DAPM_MUX("AINLMUX", AUD96P22_PD_1, 2, 0, &ainl_mux_kcontrol), + SND_SOC_DAPM_MUX("AINRMUX", AUD96P22_PD_1, 3, 0, &ainr_mux_kcontrol), + + /* ADCs */ + SND_SOC_DAPM_ADC_E("ADCL", "Capture Left", AUD96P22_PD_1, 0, 0, + aud96p22_adc_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCR", "Capture Right", AUD96P22_PD_1, 1, 0, + aud96p22_adc_event, SND_SOC_DAPM_POST_PMU), + + /* DACs */ + SND_SOC_DAPM_DAC_E("DACL", "Playback Left", AUD96P22_PD_3, 0, 0, + aud96p22_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_DAC_E("DACR", "Playback Right", AUD96P22_PD_3, 1, 0, + aud96p22_dac_event, SND_SOC_DAPM_POST_PMU), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("LD1L", AUD96P22_PD_3, 6, 0, ld1_left_kcontrols, + ARRAY_SIZE(ld1_left_kcontrols)), + SND_SOC_DAPM_MIXER("LD1R", AUD96P22_PD_3, 7, 0, ld1_right_kcontrols, + ARRAY_SIZE(ld1_right_kcontrols)), + SND_SOC_DAPM_MIXER("LD2", AUD96P22_PD_4, 2, 0, ld2_kcontrols, + ARRAY_SIZE(ld2_kcontrols)), + + /* Headset power switch */ + SND_SOC_DAPM_SUPPLY("HS1L", AUD96P22_PD_3, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HS1R", AUD96P22_PD_3, 5, 0, NULL, 0), + + /* Output pins */ + SND_SOC_DAPM_OUTPUT("HSOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTMP"), + SND_SOC_DAPM_OUTPUT("LINEOUTMN"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + SND_SOC_DAPM_OUTPUT("HSOUTR"), +}; + +static const struct snd_soc_dapm_route aud96p22_dapm_routes[] = { + { "AINLMUX", "AINL1 differential", "AINL1N" }, + { "AINLMUX", "AINL1 single-ended", "AINL1P" }, + { "AINLMUX", "AINL3 single-ended", "AINL3" }, + { "AINLMUX", "AINL2 differential", "AINL2N" }, + { "AINLMUX", "AINL2 single-ended", "AINL2P" }, + + { "AINRMUX", "AINR1 differential", "AINR1N" }, + { "AINRMUX", "AINR1 single-ended", "AINR1P" }, + { "AINRMUX", "AINR3 single-ended", "AINR3" }, + { "AINRMUX", "AINR2 differential", "AINR2N" }, + { "AINRMUX", "AINR2 single-ended", "AINR2P" }, + + { "ADCL", NULL, "AINLMUX" }, + { "ADCR", NULL, "AINRMUX" }, + + { "ADCL", NULL, "POWER" }, + { "ADCR", NULL, "POWER" }, + { "DACL", NULL, "POWER" }, + { "DACR", NULL, "POWER" }, + + { "LD1L", "DACL LD1L Switch", "DACL" }, + { "LD1L", "AINL LD1L Switch", "AINLMUX" }, + { "LD1L", "AINR LD1L Switch", "AINRMUX" }, + + { "LD1R", "DACR LD1R Switch", "DACR" }, + { "LD1R", "AINR LD1R Switch", "AINRMUX" }, + { "LD1R", "AINL LD1R Switch", "AINLMUX" }, + + { "LD2", "DACL LD2 Switch", "DACL" }, + { "LD2", "AINL LD2 Switch", "AINLMUX" }, + { "LD2", "DACR LD2 Switch", "DACR" }, + + { "HSOUTL", NULL, "LD1L" }, + { "HSOUTR", NULL, "LD1R" }, + { "HSOUTL", NULL, "HS1L" }, + { "HSOUTR", NULL, "HS1R" }, + + { "LINEOUTL", NULL, "LD1L" }, + { "LINEOUTR", NULL, "LD1R" }, + + { "LINEOUTMP", NULL, "LD2" }, + { "LINEOUTMN", NULL, "LD2" }, +}; + +static struct snd_soc_codec_driver aud96p22_driver = { + .component_driver = { + .controls = aud96p22_snd_controls, + .num_controls = ARRAY_SIZE(aud96p22_snd_controls), + .dapm_widgets = aud96p22_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aud96p22_dapm_widgets), + .dapm_routes = aud96p22_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aud96p22_dapm_routes), + }, +}; + +static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct aud96p22_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + struct regmap *regmap = priv->regmap; + unsigned int val; + + /* Master/slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = I2S1_MS_MODE; + break; + default: + return -EINVAL; + } + + regmap_update_bits(regmap, AUD96P22_I2S1_CONFIG_0, I2S1_MS_MODE, val); + + /* Audio format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = I2S1_MODE_RIGHT_J; + break; + case SND_SOC_DAIFMT_I2S: + val = I2S1_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = I2S1_MODE_LEFT_J; + break; + default: + return -EINVAL; + } + + regmap_update_bits(regmap, AUD96P22_I2S1_CONFIG_0, I2S1_MODE_MASK, val); + + return 0; +} + +static struct snd_soc_dai_ops aud96p22_dai_ops = { + .set_fmt = aud96p22_set_fmt, +}; + +#define AUD96P22_RATES SNDRV_PCM_RATE_8000_192000 +#define AUD96P22_FORMATS (\ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver aud96p22_dai = { + .name = "aud96p22-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AUD96P22_RATES, + .formats = AUD96P22_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AUD96P22_RATES, + .formats = AUD96P22_FORMATS, + }, + .ops = &aud96p22_dai_ops, +}; + +static const struct regmap_config aud96p22_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = AUD96P22_REG_MAX, + .cache_type = REGCACHE_RBTREE, +}; + +static int aud96p22_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct aud96p22_priv *priv; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &aud96p22_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(dev, "failed to init i2c regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + ret = snd_soc_register_codec(dev, &aud96p22_driver, &aud96p22_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec: %d\n", ret); + return ret; + } + + return 0; +} + +static int aud96p22_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +const struct of_device_id aud96p22_dt_ids[] = { + { .compatible = "zte,zx-aud96p22", }, + { } +}; +MODULE_DEVICE_TABLE(of, aud96p22_dt_ids); + +static struct i2c_driver aud96p22_i2c_driver = { + .driver = { + .name = "zx_aud96p22", + .of_match_table = aud96p22_dt_ids, + }, + .probe = aud96p22_i2c_probe, + .remove = aud96p22_i2c_remove, +}; +module_i2c_driver(aud96p22_i2c_driver); + +MODULE_DESCRIPTION("ZTE ASoC AUD96P22 CODEC driver"); +MODULE_AUTHOR("Baoyou Xie "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 403d2fef06104275dd4909d2684c86aabe25c917 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 22 Jun 2017 10:41:51 +0800 Subject: ASoC: nau8825: default value for property Assign default value for codec private data when property not given. If without those default value and property, the codec will work abnormally. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 53 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 41 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 97fbeba9498f..c00b86dd80dc 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2429,6 +2429,7 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) static int nau8825_read_device_properties(struct device *dev, struct nau8825 *nau8825) { + int ret; nau8825->jkdet_enable = device_property_read_bool(dev, "nuvoton,jkdet-enable"); @@ -2436,30 +2437,58 @@ static int nau8825_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8825->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); - device_property_read_u32(dev, "nuvoton,jkdet-polarity", + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8825->jkdet_polarity); - device_property_read_u32(dev, "nuvoton,micbias-voltage", + if (ret) + nau8825->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", &nau8825->micbias_voltage); - device_property_read_u32(dev, "nuvoton,vref-impedance", + if (ret) + nau8825->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", &nau8825->vref_impedance); - device_property_read_u32(dev, "nuvoton,sar-threshold-num", + if (ret) + nau8825->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num", &nau8825->sar_threshold_num); - device_property_read_u32_array(dev, "nuvoton,sar-threshold", + if (ret) + nau8825->sar_threshold_num = 4; + ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold", nau8825->sar_threshold, nau8825->sar_threshold_num); - device_property_read_u32(dev, "nuvoton,sar-hysteresis", + if (ret) { + nau8825->sar_threshold[0] = 0x08; + nau8825->sar_threshold[1] = 0x12; + nau8825->sar_threshold[2] = 0x26; + nau8825->sar_threshold[3] = 0x73; + } + ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis", &nau8825->sar_hysteresis); - device_property_read_u32(dev, "nuvoton,sar-voltage", + if (ret) + nau8825->sar_hysteresis = 0; + ret = device_property_read_u32(dev, "nuvoton,sar-voltage", &nau8825->sar_voltage); - device_property_read_u32(dev, "nuvoton,sar-compare-time", + if (ret) + nau8825->sar_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,sar-compare-time", &nau8825->sar_compare_time); - device_property_read_u32(dev, "nuvoton,sar-sampling-time", + if (ret) + nau8825->sar_compare_time = 1; + ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time", &nau8825->sar_sampling_time); - device_property_read_u32(dev, "nuvoton,short-key-debounce", + if (ret) + nau8825->sar_sampling_time = 1; + ret = device_property_read_u32(dev, "nuvoton,short-key-debounce", &nau8825->key_debounce); - device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + if (ret) + nau8825->key_debounce = 3; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", &nau8825->jack_insert_debounce); - device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + if (ret) + nau8825->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", &nau8825->jack_eject_debounce); + if (ret) + nau8825->jack_eject_debounce = 0; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- cgit v1.2.3 From 8fe19795da1b9dea2353f016622842a2f163039e Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 22 Jun 2017 11:21:01 +0800 Subject: ASoC: nau8825: fix jack type detection issue after resume Fix the issue that mic type detection error after resume. The microphone type detection procedure will recognize testing signal on JKSLV pin, but before the procedure, JKSLV already had supply voltage, that results in the failure. Therefore, the patch turns off the power and reset the jack type configuration before suspend. Then redo the jack detection procedure after resume. The patch help to fix the issue as follows: Google issue 37973093: CTIA/OMTP jack type detection failure after resume Reported Issue Chrome OS Version : ChromeOS R59-9460.13.0 Type of hardware : DVT sample What steps will reproduce the problem? (1 Play a music (2 Insert a headphones (3 Close laptop lid 3 sec then open it What is the expected output? The music is normal in the headphones. What do you see instead? Singer voice in the music is not clear. How frequently does this problem reproduce? Always What is the impact to the user, and is there a workaround? If so, what is it? Re-insert the headset or close the laptop lid and then open it again can be repaired. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c00b86dd80dc..503a6d8130b7 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1612,7 +1612,6 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_sync(dapm); break; case 2: - case 3: dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); type = SND_JACK_HEADSET; @@ -1632,6 +1631,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); break; + case 3: + /* detect error case */ + dev_err(nau8825->dev, "detection error; disable mic function\n"); + type = SND_JACK_HEADPHONE; + break; } /* Leaving HPOL/R grounded after jack insert by default. They will be @@ -2328,6 +2332,13 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + /* Reset the configuration of jack type for detection */ + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_HSD_CTRL, 0xf, 0xf); /* Cancel and reset cross talk detection funciton */ nau8825_xtalk_cancel(nau8825); /* Turn off all interruptions before system shutdown. Keep the @@ -2351,6 +2362,10 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec) disable_irq(nau8825->irq); snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8825->dapm, "SAR"); + snd_soc_dapm_disable_pin(nau8825->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8825->dapm); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); -- cgit v1.2.3 From 2bda4288e771e51946e70329c9b79605e4612f10 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 22 Jun 2017 11:57:55 +0800 Subject: ASoC: nau8825: make crosstalk function optional Make crosstalk functoin optional. The jack detection can speed up without crosstalk detection. Let the decision of function usage to platform design. The patch helps the issue concern as follows: Google issue 35574278: Chell_headphone pop back from S3 There is a concern as follows: cras getting blocked for 2 seconds (worst-case 3 seconds) As I understand, ChromeOS expects resume finishes in 1 seconds. Video/Audio playing after 3 seconds of resume seems against the spec. If we really have to make the choice I would choose pop noise instead of waiting for 3 seconds. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 6 +++++- sound/soc/codecs/nau8825.h | 1 + 2 files changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 503a6d8130b7..a8c7a556a6a8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1686,7 +1686,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->high_imped) { + if (!nau8825->xtalk_bypass && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -2504,6 +2504,10 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", + &nau8825->xtalk_bypass); + if (ret) + nau8825->xtalk_bypass = 1; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 514fd13c2f46..8aee5c8647ae 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,6 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int xtalk_bypass; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- cgit v1.2.3 From 47ca9593decee772a48d630af815aabedf99e694 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 22 Jun 2017 06:21:49 +0000 Subject: ASoC: audio-graph-card: tidyup asoc_simple_card_canonicalize_cpu() parameter asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's DAI links, not Card links. This patch fixup it. Otherwise, audio-graph-card can't handle CPU component correctly if CPU has mult-DAIs and Card uses only one of them Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 885b405d7844..ee752f62d89d 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -100,7 +100,6 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; - struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); struct device_node *rcpu_ep = of_graph_get_remote_endpoint(codec_ep); @@ -162,7 +161,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, dai_link->init = asoc_graph_card_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, - card->num_links == 1); + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); dai_link_of_err: of_node_put(cpu_ep); -- cgit v1.2.3 From 32f2bcce3ed10b93236d747701a9c04d51626cc2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 22 Jun 2017 06:22:14 +0000 Subject: ASoC: audio-graph-scu-card: tidyup asoc_simple_card_canonicalize_cpu() parameter asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's DAI links, not Card links. This patch fixup it. Otherwise, audio-graph-card can't handle CPU component correctly if CPU has mult-DAIs and Card uses only one of them Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 05934b24627b..061c7a60d6b4 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -125,7 +125,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, /* card->num_links includes Codec */ asoc_simple_card_canonicalize_cpu(dai_link, - (card->num_links - 1) == 1); + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); } else { /* FE is dummy */ dai_link->cpu_of_node = NULL; -- cgit v1.2.3 From f1f940490d3ccff96da9cc81d57c2c083c398a18 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 22 Jun 2017 06:22:49 +0000 Subject: ASoC: audio-graph-scu-card: support 2nd codec endpoint on DT audio-graph-scu-card can handle below connection which is mainly for sound mixing purpose. +----------+ +-------+ | CPU0--+--|-->| Codec | | | | +-------+ | CPU1--+ | +----------+ >From OF-graph point of view, it should have CPU0 <-> Codec, and CPU1 <-> Codec on DT. But current driver doesn't care about 2nd connection of Codec, because it is dummy from DPCM point of view. This patch can care 2nd Codec connection, and it should be supported from OF-graph point of view. It still have backward compatibility. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/audio-graph-scu-card.txt | 9 +++++-- sound/soc/generic/audio-graph-scu-card.c | 28 +++++++++++++++++++--- 2 files changed, 32 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b63c5594bbb3..8b8afe9fcb31 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -90,9 +90,12 @@ Example 2. 2 CPU 1 Codec (Mixing) ... port { - codec_endpoint: endpoint { + codec_endpoint0: endpoint { remote-endpoint = <&cpu_endpoint0>; }; + codec_endpoint1: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; }; }; @@ -101,7 +104,7 @@ Example 2. 2 CPU 1 Codec (Mixing) ports { cpu_port0: port { cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint>; + remote-endpoint = <&codec_endpoint0>; dai-format = "left_j"; ... @@ -109,6 +112,8 @@ Example 2. 2 CPU 1 Codec (Mixing) }; cpu_port1: port { cpu_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint1>; + dai-format = "left_j"; ... }; diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 061c7a60d6b4..dcd2df37bc3b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -183,6 +183,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *cpu_ep; struct device_node *codec_ep; struct device_node *rcpu_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; unsigned int daifmt = 0; int dai_idx, ret; int rc, codec; @@ -235,6 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } dai_idx = 0; + codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* * To listup valid sounds continuously, @@ -245,15 +248,22 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (codec) { - if (!codec_ep) + if (!codec_port) continue; + if (codec_port_old == codec_port) + continue; + + codec_port_old = codec_port; + /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); if (ret < 0) @@ -284,22 +294,34 @@ static int asoc_graph_get_dais_count(struct device *dev) struct device_node *cpu_port; struct device_node *cpu_ep; struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; int count = 0; int rc; + codec_port_old = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (cpu_ep) count++; - if (codec_ep) - count++; + + if (!codec_port) + continue; + + if (codec_port_old == codec_port) + continue; + + count++; + codec_port_old = codec_port; } return count; -- cgit v1.2.3 From 6b5da66322c50b4fa22f9343dcb968496f831361 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 14:49:36 +0200 Subject: ASoC: rt5645: read jd1_1 status for jd detection Read the jd status after invert control. The benefit is we don't need to invert the reading jd status when jd invert is needed. Signed-off-by: Bard Liao Tested-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 31 ++++++++----------------------- 1 file changed, 8 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 87844a45886a..8e419ea418e9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3238,24 +3238,16 @@ static void rt5645_jack_detect_work(struct work_struct *work) snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); return; - case 1: /* 2 port */ - val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; - break; - default: /* 1 port */ - val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020; + default: /* read rt5645 jd1_1 status */ + val = snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x1000; break; } - switch (val) { - /* jack in */ - case 0x30: /* 2 port */ - case 0x0: /* 1 port or 2 port */ - if (rt5645->jack_type == 0) { - report = rt5645_jack_detect(rt5645->codec, 1); - /* for push button and jack out */ - break; - } + if (!val && (rt5645->jack_type == 0)) { /* jack in */ + report = rt5645_jack_detect(rt5645->codec, 1); + } else if (!val && rt5645->jack_type != 0) { + /* for push button and jack out */ btn_type = 0; if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { /* button pressed */ @@ -3302,19 +3294,12 @@ static void rt5645_jack_detect_work(struct work_struct *work) mod_timer(&rt5645->btn_check_timer, msecs_to_jiffies(100)); } - - break; - /* jack out */ - case 0x70: /* 2 port */ - case 0x10: /* 2 port */ - case 0x20: /* 1 port */ + } else { + /* jack out */ report = 0; snd_soc_update_bits(rt5645->codec, RT5645_INT_IRQ_ST, 0x1, 0x0); rt5645_jack_detect(rt5645->codec, 0); - break; - default: - break; } snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); -- cgit v1.2.3 From 895750228c9d3361ed82e9786322604de3232466 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 14:49:37 +0200 Subject: ASoC: rt5645: rename jd_invert flag in platform data The jd_invert flag is actually used for level triggered IRQ. Rename it to let code more readable. Signed-off-by: Bard Liao Tested-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/rt5645.h | 4 ++-- sound/soc/codecs/rt5645.c | 8 ++++---- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf6152e778..c427f10a39ae 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -21,8 +21,8 @@ struct rt5645_platform_data { /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ unsigned int jd_mode; - /* Invert JD when jack insert */ - bool jd_invert; + /* Use level triggered irq */ + bool level_trigger_irq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 8e419ea418e9..e0c09bbd3f12 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3151,7 +3151,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } - if (rt5645->pdata.jd_invert) + if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); } else { /* jack out */ @@ -3172,7 +3172,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_disable_pin(dapm, "LDO2"); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); - if (rt5645->pdata.jd_invert) + if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } @@ -3586,7 +3586,7 @@ static struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, - .jd_invert = true, + .level_trigger_irq = true, }; static struct dmi_system_id dmi_platform_intel_broadwell[] = { @@ -3838,7 +3838,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); - if (rt5645->pdata.jd_invert) { + if (rt5645->pdata.level_trigger_irq) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } -- cgit v1.2.3 From aea086dda2d5df659a7c5d9efe85721e9442a133 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 14:49:38 +0200 Subject: ASoC: rt5645: add inv_jd1_1 flag The flag will invert jd1_1 status. Which will be used if the jack connector is normal closed. Signed-off-by: Bard Liao Tested-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/rt5645.h | 2 ++ sound/soc/codecs/rt5645.c | 4 ++++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index c427f10a39ae..d0c33a9972b9 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,8 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Use level triggered irq */ bool level_trigger_irq; + /* Invert JD1_1 status polarity */ + bool inv_jd1_1; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e0c09bbd3f12..162044d82632 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3833,6 +3833,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, default: break; } + if (rt5645->pdata.inv_jd1_1) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + } } regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, -- cgit v1.2.3 From ea2b5a6e3a386b89d7f9148ff8be6c78d13542a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 28 Jun 2017 14:49:39 +0200 Subject: ASoC: rt5645: Add jack detection workaround for GPD Win GPD Win requires jd_mode=3 and the inverted flag for making the jack detection working. Unfortunately, the BIOS doesn't give a nice way to match with DMI strings, and the only working way so far is to match with the board vendor/name/version/date to some known patterns. Hopefully other vendors won't do such a stupid setup, too... Thanks to Hans de Goede for the DMI matching suggestion. Suggested-by: Hans de Goede Tested-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 162044d82632..308c22f5909a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3599,6 +3599,33 @@ static struct dmi_system_id dmi_platform_intel_broadwell[] = { { } }; +static struct rt5645_platform_data gpd_win_platform_data = { + .jd_mode = 3, + .inv_jd1_1 = true, +}; + +static const struct dmi_system_id dmi_platform_gpd_win[] = { + { + /* + * Match for the GPDwin which unfortunately uses somewhat + * generic dmi strings, which is why we test for 4 strings. + * Comparing against 23 other byt/cht boards, board_vendor + * and board_name are unique to the GPDwin, where as only one + * other board has the same board_serial and 3 others have + * the same default product_name. Also the GPDwin is the + * only device to have both board_ and product_name not set. + */ + .ident = "GPD Win", + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Default string"), + DMI_MATCH(DMI_BOARD_SERIAL, "Default string"), + DMI_MATCH(DMI_PRODUCT_NAME, "Default string"), + }, + }, + {} +}; + static bool rt5645_check_dp(struct device *dev) { if (device_property_present(dev, "realtek,in2-differential") || @@ -3649,6 +3676,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, rt5645_parse_dt(rt5645, &i2c->dev); else if (dmi_check_system(dmi_platform_intel_braswell)) rt5645->pdata = general_platform_data; + else if (dmi_check_system(dmi_platform_gpd_win)) + rt5645->pdata = gpd_win_platform_data; rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect", GPIOD_IN); -- cgit v1.2.3 From 81321fe9fb69004e71353a602f9d51f656469cdd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 28 Jun 2017 15:20:06 +0300 Subject: ASoC: stm32: sai: remove some stray tabs This line was accidentally indented too far. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ba3fdc777ed8..90d439613899 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -631,7 +631,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div); mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); - cr1 = SAI_XCR1_MCKDIV_SET(div); + cr1 = SAI_XCR1_MCKDIV_SET(div); ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); -- cgit v1.2.3 From 812a532655f56bcf70b8cc7345748534b56278c3 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Mon, 26 Jun 2017 15:35:16 +0800 Subject: ASoC: nau8825: debug message of crosstalk bypass Add debug message for crosstalk function bypass. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index a8c7a556a6a8..80bae481e75d 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2440,6 +2440,8 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); + dev_dbg(dev, "crosstalk-bypass: %d\n", + nau8825->xtalk_bypass); } static int nau8825_read_device_properties(struct device *dev, -- cgit v1.2.3 From 90384fcc054f701e17e9cbbff5c14db5f877c614 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 15:38:01 +0800 Subject: ASoC: rt5670: remove duplicate route. { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, and { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, are defined twice in the driver. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index d95d2e693dc6..1146a968cb4d 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2022,7 +2022,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, { "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" }, - { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, { "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" }, @@ -2061,7 +2060,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, { "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" }, - { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, { "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" }, -- cgit v1.2.3 From 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 15:38:02 +0800 Subject: ASoC: rt5670: move set_sysclk to codec level Move set_sysclk to codec level and people can use it at both codec and dai level. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 1146a968cb4d..7fa63ad366dd 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2442,10 +2442,9 @@ static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) +static int rt5670_set_codec_sysclk(struct snd_soc_dai *dai, int clk_id, + int source, unsigned int freq, int dir) { - struct snd_soc_codec *codec = dai->codec; struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; @@ -2469,7 +2468,7 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (clk_id != RT5670_SCLK_S_RCCLK) rt5670->sysclk_src = clk_id; - dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + dev_dbg(codec->dev, "Sysclk : %dHz clock id : %d\n", freq, clk_id); return 0; } @@ -2721,7 +2720,6 @@ static int rt5670_resume(struct snd_soc_codec *codec) static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, - .set_sysclk = rt5670_set_dai_sysclk, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, }; @@ -2774,6 +2772,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { .resume = rt5670_resume, .set_bias_level = rt5670_set_bias_level, .idle_bias_off = true, + .set_sysclk = rt5670_set_codec_sysclk, .component_driver = { .controls = rt5670_snd_controls, .num_controls = ARRAY_SIZE(rt5670_snd_controls), -- cgit v1.2.3 From 5800b6970c6408d77c0286cba715d506313a2043 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Jun 2017 10:28:44 +0800 Subject: ASoC: rt5651: remove unexisting Muxes These MUXes are unexisting. So, remove them. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 44 +------------------------------------------- 1 file changed, 1 insertion(+), 43 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f5d34153e21f..db05b60d5002 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, RT5651_R_MUTE_SFT, 1, 1); -/* INL/R source */ -static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl_enum, RT5651_INL1_INR1_VOL, - RT5651_INL_SEL_SFT, rt5651_inl_src); - -static const struct snd_kcontrol_new rt5651_inl1_mux = - SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); - -static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr1_enum, RT5651_INL1_INR1_VOL, - RT5651_INR_SEL_SFT, rt5651_inr1_src); - -static const struct snd_kcontrol_new rt5651_inr1_mux = - SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); - -static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl2_enum, RT5651_INL2_INR2_VOL, - RT5651_INL_SEL_SFT, rt5651_inl2_src); - -static const struct snd_kcontrol_new rt5651_inl2_mux = - SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); - -static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr2_enum, RT5651_INL2_INR2_VOL, - RT5651_INR_SEL_SFT, rt5651_inr2_src); - -static const struct snd_kcontrol_new rt5651_inr2_mux = - SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); - - /* Stereo ADC source */ static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; @@ -955,11 +917,7 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), + /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- cgit v1.2.3 From 105e56f1ec335ab62b920882e755da49e81e5b60 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Jun 2017 10:05:29 +0800 Subject: ASoC: rt5645: enable speaker protection features This patch is uploaded for enabling the speaker protection features of the audio codec. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 308c22f5909a..630374ee692a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -59,7 +59,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x1c, 0xfd70}, {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, @@ -3759,6 +3759,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); } + regmap_update_bits(rt5645->regmap, RT5645_CLSD_OUT_CTRL, 0xc0, 0xc0); + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); -- cgit v1.2.3 From bb97142bcf8c042103e87d035a120f522d12e788 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 29 Jun 2017 14:22:25 +0100 Subject: ASoC: topology: Fix usage of SND_SOC_TPLG_INDEX_ALL during load SND_SOC_TPLG_INDEX_ALL is used by drivers to tell the core to load all topology component indexes, not just the index in the header. Fix this so that SND_SOC_TPLG_INDEX_ALL will load all components no matter their index. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 12e189701924..6070e35455aa 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2381,7 +2381,7 @@ static int soc_tplg_load_header(struct soc_tplg *tplg, /* check for matching ID */ if (hdr->index != tplg->req_index && - hdr->index != SND_SOC_TPLG_INDEX_ALL) + tplg->req_index != SND_SOC_TPLG_INDEX_ALL) return 0; tplg->index = hdr->index; -- cgit v1.2.3 From b75a65118d287aadeade8b106ed0da7b5e42c167 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 29 Jun 2017 14:22:26 +0100 Subject: ASoC: topology: show index in debug when adding DAPM routes Makes the debug output much more useful. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 6070e35455aa..73308e6d3729 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1163,7 +1163,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, return -EINVAL; } - dev_dbg(tplg->dev, "ASoC: adding %d DAPM routes\n", count); + dev_dbg(tplg->dev, "ASoC: adding %d DAPM routes for index %d\n", count, + hdr->index); for (i = 0; i < count; i++) { elem = (struct snd_soc_tplg_dapm_graph_elem *)tplg->pos; -- cgit v1.2.3 From c243d96378bd0dc1249a335c282133f05e93c253 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 29 Jun 2017 09:47:55 +0800 Subject: ASoC: rt5670: fix incompatible pointer type of set_sysclk The first parameter is codec not dai. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 7fa63ad366dd..64756dc95261 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2442,7 +2442,7 @@ static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int rt5670_set_codec_sysclk(struct snd_soc_dai *dai, int clk_id, +static int rt5670_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); -- cgit v1.2.3 From fc3ba81a5adac413312019413c91b1e6a5d8d1fa Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 29 Jun 2017 11:41:30 +0800 Subject: ASoC: nau8825: change crosstalk-bypass property to bool type The property type of "nuvoton,crosstalk-bypass" changes to boolean. The document is updated as well. Signed-off-by: John Hsu Signed-off-by: John Hsu Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/nau8825.txt | 3 +++ sound/soc/codecs/nau8825.c | 6 ++---- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index d3374231c871..2f5e973285a6 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,6 +69,8 @@ Optional properties: - 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,crosstalk-bypass: make crosstalk function bypass if set. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock @@ -96,6 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-bypass; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 80bae481e75d..46a30eaa7ace 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2506,10 +2506,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", - &nau8825->xtalk_bypass); - if (ret) - nau8825->xtalk_bypass = 1; + nau8825->xtalk_bypass = device_property_read_bool(dev, + "nuvoton,crosstalk-bypass"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- cgit v1.2.3 From b059ca720e2ac04380240500eb8d8ba931898570 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 29 Jun 2017 20:07:50 +0800 Subject: ASoC: rt5665: calibration should be done before jack detection We will set some volatile registers in jack detection function. But those volatile registers will be clear in rt5665_calibrate function because we set cache bypass and reset codec in rt5665_calibrate function. This patch add a flag to make sure that rt5665_calibrate is done before starting jack detection. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 7420010fd8e9..370ed54d1e15 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -70,6 +70,7 @@ struct rt5665_priv { int jack_type; int irq_work_delay_time; unsigned int sar_adc_value; + bool calibration_done; }; static const struct reg_default rt5665_reg[] = { @@ -1305,6 +1306,11 @@ static void rt5665_jack_detect_handler(struct work_struct *work) usleep_range(10000, 15000); } + while (!rt5665->calibration_done) { + pr_debug("%s calibration not ready\n", __func__); + usleep_range(10000, 15000); + } + mutex_lock(&rt5665->calibrate_mutex); val = snd_soc_read(rt5665->codec, RT5665_AJD1_CTRL) & 0x0010; @@ -4695,6 +4701,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) regmap_write(rt5665->regmap, RT5665_ASRC_8, 0x0120); out_unlock: + rt5665->calibration_done = true; mutex_unlock(&rt5665->calibrate_mutex); } -- cgit v1.2.3 From f986907c9225cf48e9a55233b086039152bb5b99 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 29 Jun 2017 21:26:38 +0800 Subject: ASoC: audio-graph-card: add widgets and routing for external amplifier support It's very common that audio card has a machine level amplifier which is controlled by GPIO. The patch adds DAPM widgets and routing support into audio-graph-card driver, and creates an output driver widget with event to control the amplifier via GPIO. Signed-off-by: Shawn Guo Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 48 +++++++++++++++++++++++++++++++++++- 1 file changed, 47 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index ee752f62d89d..105ec3a6e30d 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -30,6 +31,34 @@ struct graph_card_data { struct asoc_simple_dai codec_dai; } *dai_props; struct snd_soc_dai_link *dai_link; + struct gpio_desc *pa_gpio; +}; + +static int asoc_graph_card_outdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct graph_card_data *priv = snd_soc_card_get_drvdata(dapm->card); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + gpiod_set_value_cansleep(priv->pa_gpio, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + gpiod_set_value_cansleep(priv->pa_gpio, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget asoc_graph_card_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("Amplifier", SND_SOC_NOPM, + 0, 0, NULL, 0, asoc_graph_card_outdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; #define graph_priv_to_card(priv) (&(priv)->snd_card) @@ -180,8 +209,16 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) int rc, idx = 0; int ret; + ret = asoc_simple_card_of_parse_widgets(card, NULL); + if (ret < 0) + return ret; + + ret = asoc_simple_card_of_parse_routing(card, NULL, 1); + if (ret < 0) + return ret; + /* - * we need to consider "widgets", "routing", "mclk-fs" around here + * we need to consider "mclk-fs" around here * see simple-card */ @@ -233,6 +270,13 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (!dai_props || !dai_link) return -ENOMEM; + priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); + if (IS_ERR(priv->pa_gpio)) { + ret = PTR_ERR(priv->pa_gpio); + dev_err(dev, "failed to get amplifier gpio: %d\n", ret); + return ret; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; @@ -242,6 +286,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = dai_link; card->num_links = num; + card->dapm_widgets = asoc_graph_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); ret = asoc_graph_card_parse_of(priv); if (ret < 0) { -- cgit v1.2.3 From 4999b0214b05a08b42bbafcb29a0b9c413002d3f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 Jun 2017 18:08:33 +0200 Subject: ASoC: rt5645: Add quirk override by module option For making the development easier, add quirk module option to override the platform data setup. For example, a platform with inverted jack detection with jd_mode=2, pass the value 0x21 (0x1 = inv_jd1_1, 0x20 = jd_mode=2). It overrides the whole pdata fields, so pass it carefully. Signed-off-by: Takashi Iwai Tested-by: James Cameron Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 630374ee692a..909f4a6aaef1 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -34,6 +34,17 @@ #include "rl6231.h" #include "rt5645.h" +#define QUIRK_INV_JD1_1(q) ((q) & 1) +#define QUIRK_LEVEL_IRQ(q) (((q) >> 1) & 1) +#define QUIRK_IN2_DIFF(q) (((q) >> 2) & 1) +#define QUIRK_JD_MODE(q) (((q) >> 4) & 7) +#define QUIRK_DMIC1_DATA_PIN(q) (((q) >> 8) & 3) +#define QUIRK_DMIC2_DATA_PIN(q) (((q) >> 12) & 3) + +static unsigned int quirk = -1; +module_param(quirk, uint, 0444); +MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override"); + #define RT5645_DEVICE_ID 0x6308 #define RT5650_DEVICE_ID 0x6419 @@ -3679,6 +3690,15 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, else if (dmi_check_system(dmi_platform_gpd_win)) rt5645->pdata = gpd_win_platform_data; + if (quirk != -1) { + rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); + rt5645->pdata.level_trigger_irq = QUIRK_LEVEL_IRQ(quirk); + rt5645->pdata.inv_jd1_1 = QUIRK_INV_JD1_1(quirk); + rt5645->pdata.jd_mode = QUIRK_JD_MODE(quirk); + rt5645->pdata.dmic1_data_pin = QUIRK_DMIC1_DATA_PIN(quirk); + rt5645->pdata.dmic2_data_pin = QUIRK_DMIC2_DATA_PIN(quirk); + } + rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect", GPIOD_IN); -- cgit v1.2.3