July 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: 88pm860x: Don't change pm860x->dir setting if pm860x_set_dai_sysclk fails
by Axel Lin 07 Jul '15

07 Jul '15

88pm860x does not support slave mode, so it returns -EINVAL for PM860X_CLK_DIR_IN. Current code changes pm860x->dir setting before return error, so it has impact on the logic of pm860x_pcm_set_dai_fmt. This patch adds comment for the reason to return -EINVAL for PM860X_CLK_DIR_IN, and avoid changing pm860x->dir setting if pm860x_set_dai_sysclk fails. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/88pm860x-codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 38b3dad..4d91a6a 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1028,10 +1028,8 @@ static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (dir == PM860X_CLK_DIR_OUT) pm860x->dir = PM860X_CLK_DIR_OUT; - else { - pm860x->dir = PM860X_CLK_DIR_IN; + else /* Slave mode is not supported */ return -EINVAL; - } return 0; } -- 2.1.0

3 2

[alsa-devel] [PATCH v2] ASoC: ak4642: Fix up max_register setting
by Axel Lin 07 Jul '15

07 Jul '15

The max_register setting for ak4642, ak4643 and ak4648 are wrong, fix it. According to the datasheet: the maximum valid register for ak4642 is 0x1f the maximum valid register for ak4643 is 0x24 the maximum valid register for ak4648 is 0x27 The default settings for ak4642 and ak4643 are the same for 0x0 ~ 0x1f registers, so it's fine to use the same reg_default table with differnt num_reg_defaults setting. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: Improve readability by using below defines (with comment). #define ak4642_reg ak4643_reg #define NUM_AK4642_REG_DEFAULTS (FIL1_3 + 1) sound/soc/codecs/ak4642.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7c0f6552..fe963e1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -64,12 +64,15 @@ #define FIL1_0 0x1c #define FIL1_1 0x1d #define FIL1_2 0x1e -#define FIL1_3 0x1f +#define FIL1_3 0x1f /* The maximum valid register for ak4642 */ #define PW_MGMT4 0x20 #define MD_CTL5 0x21 #define LO_MS 0x22 #define HP_MS 0x23 -#define SPK_MS 0x24 +#define SPK_MS 0x24 /* The maximum valid register for ak4643 */ +#define EQ_FBEQAB 0x25 +#define EQ_FBEQCD 0x26 +#define EQ_FBEQE 0x27 /* The maximum valid register for ak4648 */ /* PW_MGMT1*/ #define PMVCM (1 << 6) /* VCOM Power Management */ @@ -241,7 +244,7 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { /* * ak4642 register cache */ -static const struct reg_default ak4642_reg[] = { +static const struct reg_default ak4643_reg[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, @@ -254,6 +257,14 @@ static const struct reg_default ak4642_reg[] = { { 36, 0x00 }, }; +/* The default settings for 0x0 ~ 0x1f registers are the same for ak4642 + and ak4643. So we reuse the ak4643 reg_default for ak4642. + The valid registers for ak4642 are 0x0 ~ 0x1f which is a subset of ak4643, + so define NUM_AK4642_REG_DEFAULTS for ak4642. +*/ +#define ak4642_reg ak4643_reg +#define NUM_AK4642_REG_DEFAULTS (FIL1_3 + 1) + static const struct reg_default ak4648_reg[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, @@ -535,15 +546,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { static const struct regmap_config ak4642_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .max_register = FIL1_3, .reg_defaults = ak4642_reg, - .num_reg_defaults = ARRAY_SIZE(ak4642_reg), + .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, +}; + +static const struct regmap_config ak4643_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = SPK_MS, + .reg_defaults = ak4643_reg, + .num_reg_defaults = ARRAY_SIZE(ak4643_reg), }; static const struct regmap_config ak4648_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; @@ -553,7 +572,7 @@ static const struct ak4642_drvdata ak4642_drvdata = { }; static const struct ak4642_drvdata ak4643_drvdata = { - .regmap_config = &ak4642_regmap, + .regmap_config = &ak4643_regmap, }; static const struct ak4642_drvdata ak4648_drvdata = { -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: lm49453: Remove fs_rate from struct lm49453_priv
by Axel Lin 07 Jul '15

07 Jul '15

fs_rate is only used in lm49453_hw_params() so don't need to store it in private data. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/lm49453.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 6600aa0..5b4086d 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -188,7 +188,6 @@ static struct reg_default lm49453_reg_defs[] = { /* codec private data */ struct lm49453_priv { struct regmap *regmap; - int fs_rate; }; /* capture path controls */ @@ -1112,13 +1111,10 @@ static int lm49453_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); u16 clk_div = 0; - lm49453->fs_rate = params_rate(params); - /* Setting DAC clock dividers based on substream sample rate. */ - switch (lm49453->fs_rate) { + switch (params_rate(params)) { case 8000: case 16000: case 32000: -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: gtm601: Constify soc_codec_dev_gtm601
by Axel Lin 07 Jul '15

07 Jul '15

Also clean up the code a bit by fixing indent. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/gtm601.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/gtm601.c b/sound/soc/codecs/gtm601.c index 12d15e5..0b80052 100644 --- a/sound/soc/codecs/gtm601.c +++ b/sound/soc/codecs/gtm601.c @@ -51,7 +51,7 @@ static struct snd_soc_dai_driver gtm601_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_gtm601 = { +static const struct snd_soc_codec_driver soc_codec_dev_gtm601 = { .dapm_widgets = gtm601_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(gtm601_dapm_widgets), .dapm_routes = gtm601_dapm_routes, @@ -80,10 +80,9 @@ MODULE_DEVICE_TABLE(of, gtm601_codec_of_match); static struct platform_driver gtm601_codec_driver = { .driver = { - .name = "gtm601", - .of_match_table = of_match_ptr(gtm601_codec_of_match), + .name = "gtm601", + .of_match_table = of_match_ptr(gtm601_codec_of_match), }, - .probe = gtm601_platform_probe, .remove = gtm601_platform_remove, }; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Constify reg_default tables
by Axel Lin 07 Jul '15

07 Jul '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/da7210.c | 6 +++--- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/rt286.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tfa9879.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- 20 files changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 36d8425..88fd37c 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -113,7 +113,7 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static struct reg_default adav80x_reg_defaults[] = { +static const struct reg_default adav80x_reg_defaults[] = { { ADAV80X_PLAYBACK_CTRL, 0x01 }, { ADAV80X_AUX_IN_CTRL, 0x01 }, { ADAV80X_REC_CTRL, 0x02 }, diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 607a63b..29a2645 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -35,7 +35,7 @@ /* * ALC5632 register cache */ -static struct reg_default alc5632_reg_defaults[] = { +static const struct reg_default alc5632_reg_defaults[] = { { 2, 0x8080 }, /* R2 - Speaker Output Volume */ { 4, 0x8080 }, /* R4 - Headphone Output Volume */ { 6, 0x8080 }, /* R6 - AUXOUT Volume */ diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 21810e5..a44ead6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -680,7 +680,7 @@ struct da7210_priv { int master; }; -static struct reg_default da7210_reg_defaults[] = { +static const struct reg_default da7210_reg_defaults[] = { { 0x00, 0x00 }, { 0x01, 0x11 }, { 0x03, 0x00 }, @@ -1182,7 +1182,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { #if IS_ENABLED(CONFIG_I2C) -static struct reg_default da7210_regmap_i2c_patch[] = { +static const struct reg_default da7210_regmap_i2c_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, @@ -1269,7 +1269,7 @@ static struct i2c_driver da7210_i2c_driver = { #if defined(CONFIG_SPI_MASTER) -static struct reg_default da7210_regmap_spi_patch[] = { +static const struct reg_default da7210_regmap_spi_patch[] = { /* Dummy read to give two pulses over nCS for SPI */ { DA7210_AUX2, 0x00 }, { DA7210_AUX2, 0x00 }, diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 238e48a..baa36f6 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -954,7 +954,7 @@ static const struct snd_soc_dapm_route da7213_audio_map[] = { {"LINE", NULL, "Lineout PGA"}, }; -static struct reg_default da7213_reg_defaults[] = { +static const struct reg_default da7213_reg_defaults[] = { { DA7213_DIG_ROUTING_DAI, 0x10 }, { DA7213_SR, 0x0A }, { DA7213_REFERENCES, 0x80 }, diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 2075236..b1cb465 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -43,7 +43,7 @@ struct da732x_priv { /* * da732x register cache - default settings */ -static struct reg_default da732x_reg_cache[] = { +static const struct reg_default da732x_reg_cache[] = { { DA732X_REG_REF1 , 0x02 }, { DA732X_REG_BIAS_EN , 0x80 }, { DA732X_REG_BIAS1 , 0x00 }, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 66bb446..925d822 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -948,7 +948,7 @@ struct da9055_priv { struct da9055_platform_data *pdata; }; -static struct reg_default da9055_reg_defaults[] = { +static const struct reg_default da9055_reg_defaults[] = { { 0x21, 0x10 }, { 0x22, 0x0A }, { 0x23, 0x00 }, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index ebd9028..ac7dfe7 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -33,7 +33,7 @@ /* Register default values for ISABELLE driver. */ -static struct reg_default isabelle_reg_defs[] = { +static const struct reg_default isabelle_reg_defs[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 6600aa0..bf77462 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -30,7 +30,7 @@ #include <asm/div64.h> #include "lm49453.h" -static struct reg_default lm49453_reg_defs[] = { +static const struct reg_default lm49453_reg_defs[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index e1c196a..e3ccea2 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -35,7 +35,7 @@ struct max9768 { u32 flags; }; -static struct reg_default max9768_default_regs[] = { +static const struct reg_default max9768_default_regs[] = { { 0, 0 }, { 3, MAX9768_CTRL_FILTERLESS}, }; diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 29549cd..8df99fb 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -22,7 +22,7 @@ static struct regmap *regmap; -static struct reg_default max9877_regs[] = { +static const struct reg_default max9877_regs[] = { { 0, 0x40 }, { 1, 0x00 }, { 2, 0x00 }, diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index b74118e..1e8ede8 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -199,7 +199,7 @@ static const struct clk_coeff coeff_div[] = { {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, }; -static struct reg_default ml26124_reg[] = { +static const struct reg_default ml26124_reg[] = { /* CLOCK control Register */ {0x00, 0x00 }, /* Sampling Rate */ {0x02, 0x00}, /* PLL NL */ diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 5c43e26..7069fe8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -38,7 +38,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - struct reg_default *index_cache; + const struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -50,7 +50,7 @@ struct rt286_priv { int clk_id; }; -static struct reg_default rt286_index_def[] = { +static const struct reg_default rt286_index_def[] = { { 0x01, 0xaaaa }, { 0x02, 0x8aaa }, { 0x03, 0x0002 }, diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index a3506e1..3ee0574 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -46,7 +46,7 @@ static const struct regmap_range_cfg rt5651_ranges[] = { .window_len = 0x1, }, }; -static struct reg_default init_list[] = { +static const struct reg_default init_list[] = { {RT5651_PR_BASE + 0x3d, 0x3e00}, }; diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 4f25a7d..71cfae0 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -38,7 +38,7 @@ #include "tas2552.h" -static struct reg_default tas2552_reg_defs[] = { +static const struct reg_default tas2552_reg_defs[] = { {TAS2552_CFG_1, 0x22}, {TAS2552_CFG_3, 0x80}, {TAS2552_DOUT, 0x00}, diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index aab0af6..bf7596b 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -160,7 +160,7 @@ static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static struct reg_default tfa9879_regs[] = { +static const struct reg_default tfa9879_regs[] = { { TFA9879_DEVICE_CONTROL, 0x0000 }, /* 0x00 */ { TFA9879_SERIAL_INTERFACE_1, 0x0a18 }, /* 0x01 */ { TFA9879_PCM_IOM2_FORMAT_1, 0x0007 }, /* 0x02 */ diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index c830832..01aeb75 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -166,7 +166,7 @@ static const struct wm_adsp_region wm2200_dsp2_regions[] = { { .type = WMFW_ADSP1_ZM, .base = WM2200_DSP2_ZM_BASE }, }; -static struct reg_default wm2200_reg_defaults[] = { +static const struct reg_default wm2200_reg_defaults[] = { { 0x000B, 0x0000 }, /* R11 - Tone Generator 1 */ { 0x0102, 0x0000 }, /* R258 - Clocking 3 */ { 0x0103, 0x0011 }, /* R259 - Clocking 4 */ diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c5748fd..8edc6d2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -113,7 +113,7 @@ WM8962_REGULATOR_EVENT(5) WM8962_REGULATOR_EVENT(6) WM8962_REGULATOR_EVENT(7) -static struct reg_default wm8962_reg[] = { +static const struct reg_default wm8962_reg[] = { { 0, 0x009F }, /* R0 - Left Input volume */ { 1, 0x049F }, /* R1 - Right Input volume */ { 2, 0x0000 }, /* R2 - HPOUTL volume */ diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 8a8db86..f2d5211 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -41,7 +41,7 @@ static const char *wm8993_supply_names[WM8993_NUM_SUPPLIES] = { "SPKVDD", }; -static struct reg_default wm8993_reg_defaults[] = { +static const struct reg_default wm8993_reg_defaults[] = { { 1, 0x0000 }, /* R1 - Power Management (1) */ { 2, 0x6000 }, /* R2 - Power Management (2) */ { 3, 0x0000 }, /* R3 - Power Management (3) */ @@ -1595,7 +1595,7 @@ static int wm8993_resume(struct snd_soc_codec *codec) #endif /* Tune DC servo configuration */ -static struct reg_default wm8993_regmap_patch[] = { +static const struct reg_default wm8993_regmap_patch[] = { { 0x44, 3 }, { 0x56, 3 }, { 0x44, 0 }, diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 3dd063f..66f32be 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -117,7 +117,7 @@ WM8996_REGULATOR_EVENT(0) WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) -static struct reg_default wm8996_reg[] = { +static const struct reg_default wm8996_reg[] = { { WM8996_POWER_MANAGEMENT_1, 0x0 }, { WM8996_POWER_MANAGEMENT_2, 0x0 }, { WM8996_POWER_MANAGEMENT_3, 0x0 }, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 8a8b1c0..ec91ce8 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -30,7 +30,7 @@ #include <sound/wm9081.h> #include "wm9081.h" -static struct reg_default wm9081_reg[] = { +static const struct reg_default wm9081_reg[] = { { 2, 0x00B9 }, /* R2 - Analogue Lineout */ { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ { 4, 0x0001 }, /* R4 - VMID Control */ -- 2.1.0

5 4

[alsa-devel] [PATCH] ASoC: fsl-ssi: Fix bitclock calculation for master mode
by Juergen Borleis 07 Jul '15

07 Jul '15

According to the datasheet 'pm', 'psr' and 'div2' should never be all 0. Since commit 541b03ad6cfe ("ASoC: fsl_ssi: Fix the incorrect limitation of the bit clock rate") this can happen, because for some bitclock rates 'pm' = 0 seems to be a valid choice but does not work due to hardware restrictions. This results into a bad hardware behaviour (slow audio for example). Feature tested on a i.MX25. Signed-off-by: Juergen Borleis <jbe(a)pengutronix.de> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..c0b940e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -633,7 +633,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, sub *= 100000; do_div(sub, freq); - if (sub < savesub) { + if (sub < savesub && !(i == 0 && psr == 0 && div2 == 0)) { baudrate = tmprate; savesub = sub; pm = i; -- 2.1.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: qcom: move board Kconfig deps to parent config
by Kenneth Westfield 07 Jul '15

07 Jul '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Rather than have each board define the same set of dependencies; move the common dependencies to the SND_SOC_QCOM parent config. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- sound/soc/qcom/Kconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 807fedf..d6ccda7 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -1,5 +1,6 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" + depends on ARCH_QCOM || COMPILE_TEST help Say Y or M if you want to add support to use audio devices in Qualcomm Technologies SOC-based platforms. @@ -26,7 +27,7 @@ config SND_SOC_LPASS_APQ8016 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM && (ARCH_QCOM || COMPILE_TEST) + depends on SND_SOC_QCOM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -35,7 +36,7 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM && (ARCH_QCOM || COMPILE_TEST) + depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help Support for Qualcomm Technologies LPASS audio block in -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 2

[alsa-devel] [PATCH] ASoC: qcom: Remove QCOM dependency from LPASS variants
by Kenneth Westfield 07 Jul '15

07 Jul '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Remove the dependency on SND_SOC_QCOM from the user- invisible LPASS variant options. This will fix randconfig build errors. Reported-by: Jim Davis <jim.epost(a)gmail.com> Fixes: dc1ebd1811e9 ("ASoC: qcom: Add apq8016 lpass driver support") Fixes: 9bae4880acee ("ASoC: qcom: move ipq806x specific bits out of lpass driver") Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- Created fix in response to Jim's bug report: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-May/092870.html Note: I did some searching but didn't see an example of a fix for the same type of bug introduced by 2 commits. If this needs to be split into two fixes, let me know. sound/soc/qcom/Kconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 29fff6d..10837bd 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -14,13 +14,11 @@ config SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_IPQ806X tristate - depends on SND_SOC_QCOM select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate - depends on SND_SOC_QCOM select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 2

[alsa-devel] [PATCH 1/6] ASoC: rt5645: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 07 Jul '15

07 Jul '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all other manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/rt5645.c | 73 +++++++++++++++++++++-------------------------- 1 file changed, 32 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a5bc96a..6335746 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2642,7 +2642,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) { snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2, @@ -2760,20 +2760,17 @@ static int rt5650_calibration(struct rt5645_priv *rt5645) static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, bool enable) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); if (enable) { - snd_soc_dapm_mutex_lock(&codec->dapm); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "ADC L power"); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "ADC R power"); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "LDO2"); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync_unlocked(&codec->dapm); - snd_soc_dapm_mutex_unlock(&codec->dapm); + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Mic Det Power"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x8); @@ -2786,23 +2783,20 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); - snd_soc_dapm_mutex_lock(&codec->dapm); - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "ADC L power"); - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "ADC R power"); + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(dapm, "ADC R power"); if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "LDO2"); - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync_unlocked(&codec->dapm); - snd_soc_dapm_mutex_unlock(&codec->dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Det Power"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); } } static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); unsigned int val; @@ -2811,10 +2805,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) if (codec->component.card->instantiated) { /* for jack type detect */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); } else { /* Power up necessary bits for JD if dapm is not ready yet */ @@ -2847,9 +2840,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } } else { if (codec->component.card->instantiated) { - snd_soc_dapm_disable_pin(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); } else regmap_update_bits(rt5645->regmap, RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); @@ -2863,11 +2855,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) else { if (codec->component.card->instantiated) { if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(&codec->dapm, - "LDO2"); - snd_soc_dapm_disable_pin(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_disable_pin(dapm, "LDO2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); } else { if (rt5645->pdata.jd_mode == 0) regmap_update_bits(rt5645->regmap, @@ -3043,21 +3033,22 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) static int rt5645_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); rt5645->codec = codec; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - snd_soc_dapm_add_routes(&codec->dapm, + snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); break; case CODEC_TYPE_RT5650: - snd_soc_dapm_new_controls(&codec->dapm, + snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - snd_soc_dapm_add_routes(&codec->dapm, + snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); break; @@ -3067,9 +3058,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) /* for JD function */ if (rt5645->pdata.jd_mode) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_sync(dapm); } return 0; -- 2.1.4

3 12

[alsa-devel] [PATCH] ASoC: dpcm: Add checks of playback/capture before dpcm_get_be
by Koro Chen 07 Jul '15

07 Jul '15

In dpcm_get_be(), it looks for a BE rtd that has the DAI widget according to current stream type. Only playback_widgets are searched in the case of playback stream and vice versa. However, the DAI widget itself can be playback or capture. If the DAI widget is capture, but current stream type is playback, dpcm_get_be() will always fail to find a rtd, print error messages, and continue to the next DAI widget in list. We can just skip this DAI widget to further suppress error messages. This happens in a special case when 2 codecs are inter-connected, and the 1st codec's "capture" widget is used to send data to the 2nd codec during "playback": mtk-rt5650-rt5676 sound: ASoC: can't get playback BE for Sub AIF2 Capture rt5650_rt5676 Playback: ASoC: no BE found for Sub AIF2 Capture Add checks to continue to next DAI widget if current DAI widget's direction does not match the stream type. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-pcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 256b9c9..a6d3313 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1306,7 +1306,12 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, switch (list->widgets[i]->id) { case snd_soc_dapm_dai_in: + if (stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; + break; case snd_soc_dapm_dai_out: + if (stream != SNDRV_PCM_STREAM_CAPTURE) + continue; break; default: continue; -- 1.8.1.1.dirty

2 1