diff options
Diffstat (limited to 'sound/soc')
33 files changed, 435 insertions, 286 deletions
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..4544d8eb1452 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -11,7 +11,7 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 @@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602 config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && I2C select SND_BF5XX_SOC_I2S select SND_SOC_ADAU1701 - select I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ board connected to one of the Blackfin evaluation boards like the @@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_ADAV80X help @@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X config SND_BF5XX_SOC_AD1836 tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SPI_MASTER select SND_BF5XX_SOC_I2S select SND_SOC_AD1836 help @@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836 config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_AD193X help diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 75d0ad5d2dcb..647a72cda005 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1328,6 +1328,9 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; codec->control_data = pm860x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret) + return ret; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 7257a8885f42..34d965a4a040 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = { static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; -static const struct soc_enum ad1980_cap_src = - SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); +static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src, + AC97_REC_SEL, 8, 0, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..7a272fa90b39 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -124,9 +124,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); -/* This is a lie. after -102 db, it stays at -102 */ -/* maybe a range would be better */ -static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); static const char *chan_mix[] = { @@ -141,7 +140,7 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", @@ -149,7 +148,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..1102ced9b20e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -347,7 +347,7 @@ static const char * const right_swap_text[] = { static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -356,7 +356,7 @@ static const struct snd_kcontrol_new adca_mixer = SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -365,7 +365,7 @@ static const struct snd_kcontrol_new pcma_mixer = SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -374,7 +374,7 @@ static const struct snd_kcontrol_new adcb_mixer = SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..7b95f7cbc515 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -319,7 +319,7 @@ static const char * const cs42l73_mono_mix_texts[] = { static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; static const struct soc_enum spk_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -337,7 +337,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer = SOC_DAPM_ENUM("Route", spk_xsp_enum); static const struct soc_enum esl_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -346,7 +346,7 @@ static const struct snd_kcontrol_new esl_asp_mixer = SOC_DAPM_ENUM("Route", esl_asp_enum); static const struct soc_enum esl_xsp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b6569910..f4d965ebc29e 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..422812613a28 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1523,8 +1523,15 @@ static int da9055_remove(struct i2c_client *client) return 0; } +/* + * DO NOT change the device Ids. The naming is intentionally specific as both + * the CODEC and PMIC parts of this chip are instantiated separately as I2C + * devices (both have configurable I2C addresses, and are to all intents and + * purposes separate). As a result there are specific DA9055 Ids for CODEC + * and PMIC, which must be different to operate together. + */ static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1532,7 +1539,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..cb736ddc446d 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; static const struct soc_enum isabelle_rx1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), }; static const struct soc_enum isabelle_rx2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), }; /* Headset DAC playback switches */ @@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; static const struct soc_enum isabelle_atx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), }; static const struct soc_enum isabelle_vtx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), }; static const struct snd_kcontrol_new atx_mux_controls = @@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = { /* Left analog microphone selection */ static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; -static const struct soc_enum isabelle_amic1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, - ARRAY_SIZE(isabelle_amic1_texts), - isabelle_amic1_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum, + ISABELLE_AMIC_CFG_REG, 5, + isabelle_amic1_texts); -static const struct soc_enum isabelle_amic2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, - ARRAY_SIZE(isabelle_amic2_texts), - isabelle_amic2_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum, + ISABELLE_AMIC_CFG_REG, 4, + isabelle_amic2_texts); static const struct snd_kcontrol_new amic1_control = SOC_DAPM_ENUM("Route", isabelle_amic1_enum); @@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; static const struct soc_enum isabelle_st_audio_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), }; static const struct soc_enum isabelle_st_voice_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), - SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), }; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d16b41..9f714ea86613 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg) case M98090_REG_RECORD_TDM_SLOT: case M98090_REG_SAMPLE_RATE: case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_REVISION_ID: return true; default: return false; @@ -1769,16 +1770,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regcache_sync(max98090->regmap); - - if (ret != 0) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } - } - if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { /* * Set to normal bias level. @@ -1792,6 +1783,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: /* Set internal pull-up to lowest power mode */ snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..886924934aa5 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 52e7cb08434b..fa2b8e07f420 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -210,7 +210,7 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return 0; + return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); } static struct snd_soc_dai_ops si476x_dai_ops = { diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 06edb396e733..2735361a4c3c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = { 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), }; -static const struct soc_enum sta32x_drc_ac_enum = - SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, - 2, sta32x_drc_ac); -static const struct soc_enum sta32x_auto_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, - 3, sta32x_auto_eq_mode); -static const struct soc_enum sta32x_auto_gc_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, - 4, sta32x_auto_gc_mode); -static const struct soc_enum sta32x_auto_xo_enum = - SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, - 16, sta32x_auto_xo_mode); -static const struct soc_enum sta32x_preset_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, - 32, sta32x_preset_eq_mode); -static const struct soc_enum sta32x_limiter_ch1_enum = - SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch2_enum = - SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch3_enum = - SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter1_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter2_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter1_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); -static const struct soc_enum sta32x_limiter2_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum, + STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + sta32x_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum, + STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + sta32x_auto_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum, + STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + sta32x_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum, + STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + sta32x_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum, + STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + sta32x_preset_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum, + STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum, + STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum, + STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum, + STA32X_L1AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum, + STA32X_L2AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum, + STA32X_L1AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, + STA32X_L2AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); /* byte array controls for setting biquad, mixer, scaling coefficients; * for biquads all five coefficients need to be set in one go, @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; @@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), -SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum), /* depending on mode, the attack/release thresholds have * two different enum definitions; provide both diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 48dc7d2fee36..6d684d934f4d 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -117,19 +117,23 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8400_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8400_left_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_left_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACL_SHIFT, + wm8400_digital_sidetone); -static const struct soc_enum wm8400_right_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_right_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACR_SHIFT, + wm8400_digital_sidetone); static const char *wm8400_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8400_right_adcmode_enum = -SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8400_right_adcmode_enum, + WM8400_ADC_CTRL, + WM8400_ADC_HPF_CUT_SHIFT, + wm8400_adcmode); static const struct snd_kcontrol_new wm8400_snd_controls[] = { /* INMIXL */ @@ -422,9 +426,10 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, static const char *wm8400_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8400_ainlmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT, - ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainlmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINLMODE_SHIFT, + wm8400_ainlmux); static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); @@ -435,9 +440,10 @@ SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); static const char *wm8400_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8400_ainrmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT, - ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINRMODE_SHIFT, + wm8400_ainrmux); static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 89a18d82f303..5bce21013485 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -196,8 +196,8 @@ static const char *ain_text[] = { "AIN5", "AIN6", "AIN7", "AIN8" }; -static const struct soc_enum ain_enum = - SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text); +static SOC_ENUM_DOUBLE_DECL(ain_enum, + WM8770_ADCMUX, 0, 4, ain_text); static const struct snd_kcontrol_new ain_mux = SOC_DAPM_ENUM("Capture Mux", ain_enum); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e98bc7038a08..43c2201cb901 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" }; -static const struct soc_enum mic_bias_level = -SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt); +static SOC_ENUM_SINGLE_DECL(mic_bias_level, + WM8900_REG_INCTL, 8, mic_bias_level_txt); static const char *dac_mute_rate_txt[] = { "Fast", "Slow" }; -static const struct soc_enum dac_mute_rate = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(dac_mute_rate, + WM8900_REG_DACCTRL, 7, dac_mute_rate_txt); static const char *dac_deemphasis_txt[] = { "Disabled", "32kHz", "44.1kHz", "48kHz" }; -static const struct soc_enum dac_deemphasis = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt); +static SOC_ENUM_SINGLE_DECL(dac_deemphasis, + WM8900_REG_DACCTRL, 4, dac_deemphasis_txt); static const char *adc_hpf_cut_txt[] = { "Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3" }; -static const struct soc_enum adc_hpf_cut = -SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt); +static SOC_ENUM_SINGLE_DECL(adc_hpf_cut, + WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt); static const char *lr_txt[] = { "Left", "Right" }; -static const struct soc_enum aifl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifl_src, + WM8900_REG_AUDIO1, 15, lr_txt); -static const struct soc_enum aifr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifr_src, + WM8900_REG_AUDIO1, 14, lr_txt); -static const struct soc_enum dacl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacl_src, + WM8900_REG_AUDIO2, 15, lr_txt); -static const struct soc_enum dacr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacr_src, + WM8900_REG_AUDIO2, 14, lr_txt); static const char *sidetone_txt[] = { "Disabled", "Left ADC", "Right ADC" }; -static const struct soc_enum dacl_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone, + WM8900_REG_SIDETONE, 2, sidetone_txt); -static const struct soc_enum dacr_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone, + WM8900_REG_SIDETONE, 0, sidetone_txt); static const struct snd_kcontrol_new wm8900_snd_controls[] = { SOC_ENUM("Mic Bias Level", mic_bias_level), @@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0), static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" }; -static const struct soc_enum wm8900_lineout2_lp_mux = -SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux); +static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux, + WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux); static const struct snd_kcontrol_new wm8900_lineout2_lp = SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux); diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b7488f190d2b..d4248e00160e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, data32 &= 0xffffff; - wm8994_bulk_write(codec->control_data, + wm8994_bulk_write(wm8994->wm8994, data32 & 0xffffff, block_len / 2, (void *)(data + 8)); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a0f3ef..2ee23a39622c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cbc4603..adb72063d44e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -265,21 +265,21 @@ static const char *sidetone_hpf_text[] = { "2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz" }; -static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text); +static SOC_ENUM_SINGLE_DECL(sidetone_hpf, + WM8994_SIDETONE, 7, sidetone_hpf_text); static const char *adc_hpf_text[] = { "HiFi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum aif1adc1_hpf = - SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif1adc1_hpf, + WM8994_AIF1_ADC1_FILTERS, 13, adc_hpf_text); -static const struct soc_enum aif1adc2_hpf = - SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif1adc2_hpf, + WM8994_AIF1_ADC2_FILTERS, 13, adc_hpf_text); -static const struct soc_enum aif2adc_hpf = - SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif2adc_hpf, + WM8994_AIF2_ADC_FILTERS, 13, adc_hpf_text); static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0); static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); @@ -501,39 +501,39 @@ static const char *aif_chan_src_text[] = { "Left", "Right" }; -static const struct soc_enum aif1adcl_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1adcl_src, + WM8994_AIF1_CONTROL_1, 15, aif_chan_src_text); -static const struct soc_enum aif1adcr_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1adcr_src, + WM8994_AIF1_CONTROL_1, 14, aif_chan_src_text); -static const struct soc_enum aif2adcl_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2adcl_src, + WM8994_AIF2_CONTROL_1, 15, aif_chan_src_text); -static const struct soc_enum aif2adcr_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2adcr_src, + WM8994_AIF2_CONTROL_1, 14, aif_chan_src_text); -static const struct soc_enum aif1dacl_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1dacl_src, + WM8994_AIF1_CONTROL_2, 15, aif_chan_src_text); -static const struct soc_enum aif1dacr_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1dacr_src, + WM8994_AIF1_CONTROL_2, 14, aif_chan_src_text); -static const struct soc_enum aif2dacl_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacl_src, + WM8994_AIF2_CONTROL_2, 15, aif_chan_src_text); -static const struct soc_enum aif2dacr_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacr_src, + WM8994_AIF2_CONTROL_2, 14, aif_chan_src_text); static const char *osr_text[] = { "Low Power", "High Performance", }; -static const struct soc_enum dac_osr = - SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(dac_osr, + WM8994_OVERSAMPLING, 0, osr_text); -static const struct soc_enum adc_osr = - SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(adc_osr, + WM8994_OVERSAMPLING, 1, osr_text); static const struct snd_kcontrol_new wm8994_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, @@ -690,17 +690,20 @@ static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; -static const struct soc_enum wm8958_aif1dac1_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, - WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac1_ng_hold, + WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, + wm8958_ng_text); -static const struct soc_enum wm8958_aif1dac2_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, - WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac2_ng_hold, + WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, + wm8958_ng_text); -static const struct soc_enum wm8958_aif2dac_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, - WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif2dac_ng_hold, + WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, + wm8958_ng_text); static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), @@ -1341,8 +1344,8 @@ static const char *adc_mux_text[] = { "DMIC", }; -static const struct soc_enum adc_enum = - SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); +static SOC_ENUM_SINGLE_DECL(adc_enum, + 0, 0, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); @@ -1478,14 +1481,14 @@ static const char *sidetone_text[] = { "ADC/DMIC1", "DMIC2", }; -static const struct soc_enum sidetone1_enum = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone1_enum, + WM8994_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetone1_mux = SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); -static const struct soc_enum sidetone2_enum = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone2_enum, + WM8994_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new sidetone2_mux = SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); @@ -1498,22 +1501,24 @@ static const char *loopback_text[] = { "None", "ADCDAT", }; -static const struct soc_enum aif1_loopback_enum = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, - loopback_text); +static SOC_ENUM_SINGLE_DECL(aif1_loopback_enum, + WM8994_AIF1_CONTROL_2, + WM8994_AIF1_LOOPBACK_SHIFT, + loopback_text); static const struct snd_kcontrol_new aif1_loopback = SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); -static const struct soc_enum aif2_loopback_enum = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, - loopback_text); +static SOC_ENUM_SINGLE_DECL(aif2_loopback_enum, + WM8994_AIF2_CONTROL_2, + WM8994_AIF2_LOOPBACK_SHIFT, + loopback_text); static const struct snd_kcontrol_new aif2_loopback = SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); -static const struct soc_enum aif1dac_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); +static SOC_ENUM_SINGLE_DECL(aif1dac_enum, + WM8994_POWER_MANAGEMENT_6, 0, aif1dac_text); static const struct snd_kcontrol_new aif1dac_mux = SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum); @@ -1522,8 +1527,8 @@ static const char *aif2dac_text[] = { "AIF2DACDAT", "AIF3DACDAT", }; -static const struct soc_enum aif2dac_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text); +static SOC_ENUM_SINGLE_DECL(aif2dac_enum, + WM8994_POWER_MANAGEMENT_6, 1, aif2dac_text); static const struct snd_kcontrol_new aif2dac_mux = SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum); @@ -1532,8 +1537,8 @@ static const char *aif2adc_text[] = { "AIF2ADCDAT", "AIF3DACDAT", }; -static const struct soc_enum aif2adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text); +static SOC_ENUM_SINGLE_DECL(aif2adc_enum, + WM8994_POWER_MANAGEMENT_6, 2, aif2adc_text); static const struct snd_kcontrol_new aif2adc_mux = SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum); @@ -1542,14 +1547,14 @@ static const char *aif3adc_text[] = { "AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM", }; -static const struct soc_enum wm8994_aif3adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text); +static SOC_ENUM_SINGLE_DECL(wm8994_aif3adc_enum, + WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text); static const struct snd_kcontrol_new wm8994_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum); -static const struct soc_enum wm8958_aif3adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif3adc_enum, + WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text); static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); @@ -1558,8 +1563,8 @@ static const char *mono_pcm_out_text[] = { "None", "AIF2ADCL", "AIF2ADCR", }; -static const struct soc_enum mono_pcm_out_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text); +static SOC_ENUM_SINGLE_DECL(mono_pcm_out_enum, + WM8994_POWER_MANAGEMENT_6, 9, mono_pcm_out_text); static const struct snd_kcontrol_new mono_pcm_out_mux = SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum); @@ -1569,14 +1574,14 @@ static const char *aif2dac_src_text[] = { }; /* Note that these two control shouldn't be simultaneously switched to AIF3 */ -static const struct soc_enum aif2dacl_src_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacl_src_enum, + WM8994_POWER_MANAGEMENT_6, 7, aif2dac_src_text); static const struct snd_kcontrol_new aif2dacl_src_mux = SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum); -static const struct soc_enum aif2dacr_src_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacr_src_enum, + WM8994_POWER_MANAGEMENT_6, 8, aif2dac_src_text); static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 70ff3772079f..5e3bc3c6801a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -399,6 +399,7 @@ static struct platform_driver davinci_evm_driver = { .driver = { .name = "davinci_evm", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = of_match_ptr(davinci_evm_dt_ids), }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b7858bfa0295..670afa29e30d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -263,7 +263,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: @@ -317,7 +319,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -354,10 +357,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + break; } - - return 0; +out: + pm_runtime_put_sync(mcasp->dev); + return ret; } static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) @@ -448,7 +453,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, return 0; } -static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, +static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int channels) { int i; @@ -524,12 +529,18 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, return 0; } -static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) +static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; u32 busel = 0; + if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) { + dev_err(mcasp->dev, "tdm slot %d not supported\n", + mcasp->tdm_slots); + return -EINVAL; + } + active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); @@ -539,35 +550,21 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) if (!mcasp->dat_port) busel = TXSEL; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); - else - printk(KERN_ERR "playback tdm slot %d not supported\n", - mcasp->tdm_slots); - } else { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); - else - printk(KERN_ERR "capture tdm slot %d not supported\n", - mcasp->tdm_slots); - } + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + + return 0; } /* S/PDIF */ -static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ @@ -589,6 +586,8 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + + return 0; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -605,13 +604,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, u8 slots = mcasp->tdm_slots; u8 active_serializers; int channels; + int ret; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; active_serializers = (channels + slots - 1) / slots; - if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL) + if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = mcasp->txnumevt * active_serializers; @@ -619,9 +619,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - davinci_hw_dit_param(mcasp); + ret = mcasp_dit_hw_param(mcasp); else - davinci_hw_param(mcasp, substream->stream); + ret = mcasp_i2s_hw_param(mcasp, substream->stream); + + if (ret) + return ret; switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: @@ -678,19 +681,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = pm_runtime_get_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n"); davinci_mcasp_start(mcasp, substream->stream); break; - case SNDRV_PCM_TRIGGER_SUSPEND: - davinci_mcasp_stop(mcasp, substream->stream); - ret = pm_runtime_put_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n"); - break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(mcasp, substream->stream); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed261e7..c84026c99134 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); @@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 9c9f957fcae1..75e14033e8d8 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -322,7 +322,7 @@ #define ESAI_xSMB_xS_SHIFT 0 #define ESAI_xSMB_xS_WIDTH 16 #define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) -#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK) /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ #define ESAI_PRRC_PDC_SHIFT 0 diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee782dbbf..a2fd7321b5a9 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae78969f..1cb22dd034eb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -33,8 +33,7 @@ struct imx_sgtl5000_data { static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct imx_sgtl5000_data *data = container_of(rtd->card, - struct imx_sgtl5000_data, card); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card); struct device *dev = rtd->card->dev; int ret; @@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -184,7 +185,8 @@ fail: static int imx_sgtl5000_remove(struct platform_device *pdev) { - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card); clk_put(data->codec_clk); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc391de..3a3d17ce6ba4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -289,7 +291,8 @@ fail: static int imx_wm8962_remove(struct platform_device *pdev) { - struct imx_wm8962_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e402710..d163e18d85d4 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -305,7 +305,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41cfc828..350757400391 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help - Sat Y if you want to add support for SoC audio on the Jive. + Say Y if you want to add support for SoC audio on the Jive. config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" @@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help - Sat Y if you want to add support for SoC audio on the SMDK. + Say Y if you want to add support for SoC audio on the SMDK. config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1967f44e7cd4..710a079a7377 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1711,9 +1711,9 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - fsi->clk_master = 1; break; case SND_SOC_DAIFMT_CBS_CFS: + fsi->clk_master = 1; /* codec is slave, cpu is master */ break; default: return -EINVAL; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 743de5e3b1e1..3a4fe9d0d4f2 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -486,10 +486,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - rdai->clk_master = 1; + rdai->clk_master = 0; break; case SND_SOC_DAIFMT_CBS_CFS: - rdai->clk_master = 0; + rdai->clk_master = 1; /* codec is slave, cpu is master */ break; default: return -EINVAL; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13187f7..b9dc6acbba8c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1218,7 +1218,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to bypass %s: %d\n", + "ASoC: Failed to unbypass %s: %d\n", w->name, ret); } @@ -1228,7 +1228,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3210,15 +3210,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - mutex_unlock(&card->dapm_mutex); - snd_soc_dapm_sync(&card->dapm); return 0; } @@ -3248,7 +3244,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } break; @@ -3767,23 +3763,52 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, } /** + * snd_soc_dapm_enable_pin_unlocked - enable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Enables input/output pin and its parents or children widgets iff there is + * a valid audio route and active audio stream. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin_unlocked); + +/** * snd_soc_dapm_enable_pin - enable pin. * @dapm: DAPM context * @pin: pin name * * Enables input/output pin and its parents or children widgets iff there is * a valid audio route and active audio stream. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 1); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 1); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); /** - * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * snd_soc_dapm_force_enable_pin_unlocked - force a pin to be enabled * @dapm: DAPM context * @pin: pin name * @@ -3791,11 +3816,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * intended for use with microphone bias supplies used in microphone * jack detection. * + * Requires external locking. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, - const char *pin) +int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); @@ -3811,25 +3838,103 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin_unlocked); + +/** + * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * @dapm: DAPM context + * @pin: pin name + * + * Enables input/output pin regardless of any other state. This is + * intended for use with microphone bias supplies used in microphone + * jack detection. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, pin); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin); /** + * snd_soc_dapm_disable_pin_unlocked - disable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Disables input/output pin and its parents or children widgets. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin_unlocked); + +/** * snd_soc_dapm_disable_pin - disable pin. * @dapm: DAPM context * @pin: pin name * * Disables input/output pin and its parents or children widgets. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 0); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 0); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** + * snd_soc_dapm_nc_pin_unlocked - permanently disable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin_unlocked); + +/** * snd_soc_dapm_nc_pin - permanently disable pin. * @dapm: DAPM context * @pin: pin name @@ -3845,7 +3950,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); */ int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 0); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 0); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index fe99f461aff0..19cca043e6e4 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -213,10 +213,7 @@ static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) static int spdif_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); ucontrol->value.integer.value[0] = host->saved_params.mute; @@ -226,10 +223,7 @@ static int spdif_mute_get(struct snd_kcontrol *kcontrol, static int spdif_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); if (host->saved_params.mute == ucontrol->value.integer.value[0]) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e0305a148568..9edd68db9f48 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); - drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); - if (!drvdata) - return -ENOMEM; platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; if (sizeof(drvdata->physbase) > sizeof(r->start) && |