diff options
Diffstat (limited to 'sound/soc')
39 files changed, 618 insertions, 151 deletions
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index dd0c2a4f83a3..e0869aaa1e93 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -111,6 +111,7 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, bf5xx_i2s->tcr2 |= 7; bf5xx_i2s->rcr2 |= 7; sport_handle->wdsize = 1; + break; case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s->tcr2 |= 15; bf5xx_i2s->rcr2 |= 15; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c07448d..6fd174be3bdf 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -351,6 +351,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err; diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index a153b168129b..bce45c197e1d 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index dafdbe87edeb..0c499c638692 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -64,7 +64,7 @@ #define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 #define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 -#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0002 #define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 #define ADAU1701_AUXNPOW_VBPD 0x40 diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d0378709702..687565d08d9c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 389f23253831..663a2a748626 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1455,6 +1455,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, try_wait_for_completion(&fll->ok); regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (fll->ref_src >= 0 && fll->sync_src >= 0 && fll->ref_src != fll->sync_src) @@ -1473,6 +1475,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 1e0fa3b5f79a..e1dfebbea650 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 ee25f325d65c..b99af6362de6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -350,7 +350,7 @@ 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); @@ -359,7 +359,7 @@ 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); @@ -368,7 +368,7 @@ 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); @@ -377,7 +377,7 @@ 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/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012c4719..a935d7381af6 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -179,7 +179,7 @@ #define CS42L52_MICB_CTL 0x11 #define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF #define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6 -#define CS42L52_MIC_CTL_TYPE_MASK 0xDF +#define CS42L52_MIC_CTL_TYPE_MASK 0x20 #define CS42L52_MIC_CTL_TYPE_SHIFT 5 diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3b20c86cdb01..eade6e2d883d 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -325,7 +325,7 @@ 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); @@ -343,7 +343,7 @@ 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); @@ -352,7 +352,7 @@ 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 dc0284dc9e6f..76fdf0a598bc 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/max98090.c b/sound/soc/codecs/max98090.c index 8d14a76c7249..be8de7ce1cda 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -255,6 +255,7 @@ static struct reg_default max98090_reg[] = { static bool max98090_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { + case M98090_REG_SOFTWARE_RESET: case M98090_REG_DEVICE_STATUS: case M98090_REG_JACK_STATUS: case M98090_REG_REVISION_ID: @@ -336,6 +337,7 @@ 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; @@ -1362,8 +1364,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"STENL Mux", "Sidetone Left", "DMICL"}, {"STENR Mux", "Sidetone Right", "ADCR"}, {"STENR Mux", "Sidetone Right", "DMICR"}, - {"DACL", "NULL", "STENL Mux"}, - {"DACR", "NULL", "STENL Mux"}, + {"DACL", NULL, "STENL Mux"}, + {"DACR", NULL, "STENL Mux"}, {"AIFINL", NULL, "SHDN"}, {"AIFINR", NULL, "SHDN"}, @@ -1755,16 +1757,6 @@ 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. @@ -1778,6 +1770,16 @@ 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, @@ -2232,7 +2234,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { @@ -2342,6 +2344,8 @@ static int max98090_runtime_resume(struct device *dev) regcache_cache_only(max98090->regmap, false); + max98090_reset(max98090); + regcache_sync(max98090->regmap); return 0; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd1642970..8dbcacd44e6a 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ea479388fb5c..23670737116e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1317,8 +1317,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - SGTL5000_SMALL_POP, - SGTL5000_SMALL_POP); + SGTL5000_SMALL_POP, 1); /* disable short cut detector */ snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index d3a68bbfea00..0bd6e1cd8200 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0x0001 +#define SGTL5000_SMALL_POP 0 /* * SGTL5000_CHIP_MIC_CTRL diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f2491232..bb3878c9625f 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -176,6 +176,13 @@ static int _process_sigma_firmware(struct device *dev, goto done; } + if (ssfw_head->version != 1) { + dev_err(dev, + "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", + ssfw_head->version); + goto done; + } + crc = crc32(0, fw->data + sizeof(*ssfw_head), fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index cfb55fe35e98..8517e70bc24b 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; @@ -432,7 +432,7 @@ 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/wm5110.c b/sound/soc/codecs/wm5110.c index 88ad7db52dde..3775394c9c8b 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -37,6 +37,95 @@ struct wm5110_priv { struct arizona_fll fll[2]; }; +static const struct reg_default wm5110_sysclk_revd_patch[] = { + { 0x3093, 0x1001 }, + { 0x30E3, 0x1301 }, + { 0x3133, 0x1201 }, + { 0x3183, 0x1501 }, + { 0x31D3, 0x1401 }, + { 0x0049, 0x01ea }, + { 0x004a, 0x01f2 }, + { 0x0057, 0x01e7 }, + { 0x0058, 0x01fb }, + { 0x33ce, 0xc4f5 }, + { 0x33cf, 0x1361 }, + { 0x33d0, 0x0402 }, + { 0x33d1, 0x4700 }, + { 0x33d2, 0x026d }, + { 0x33d3, 0xff00 }, + { 0x33d4, 0x026d }, + { 0x33d5, 0x0101 }, + { 0x33d6, 0xc4f5 }, + { 0x33d7, 0x0361 }, + { 0x33d8, 0x0402 }, + { 0x33d9, 0x6701 }, + { 0x33da, 0xc4f5 }, + { 0x33db, 0x136f }, + { 0x33dc, 0xc4f5 }, + { 0x33dd, 0x134f }, + { 0x33de, 0xc4f5 }, + { 0x33df, 0x131f }, + { 0x33e0, 0x026d }, + { 0x33e1, 0x4f01 }, + { 0x33e2, 0x026d }, + { 0x33e3, 0xf100 }, + { 0x33e4, 0x026d }, + { 0x33e5, 0x0001 }, + { 0x33e6, 0xc4f5 }, + { 0x33e7, 0x0361 }, + { 0x33e8, 0x0402 }, + { 0x33e9, 0x6601 }, + { 0x33ea, 0xc4f5 }, + { 0x33eb, 0x136f }, + { 0x33ec, 0xc4f5 }, + { 0x33ed, 0x134f }, + { 0x33ee, 0xc4f5 }, + { 0x33ef, 0x131f }, + { 0x33f0, 0x026d }, + { 0x33f1, 0x4e01 }, + { 0x33f2, 0x026d }, + { 0x33f3, 0xf000 }, + { 0x33f6, 0xc4f5 }, + { 0x33f7, 0x1361 }, + { 0x33f8, 0x0402 }, + { 0x33f9, 0x4600 }, + { 0x33fa, 0x026d }, + { 0x33fb, 0xfe00 }, +}; + +static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 3: + patch = wm5110_sysclk_revd_patch; + patch_size = ARRAY_SIZE(wm5110_sysclk_revd_patch); + break; + default: + return 0; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -386,7 +475,7 @@ static const struct snd_kcontrol_new wm5110_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5110_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, @@ -856,7 +945,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "HPOUT2R", NULL, "OUT2R" }, { "HPOUT3L", NULL, "OUT3L" }, - { "HPOUT3R", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5276062d6c79..10d492b6a5b4 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -407,10 +407,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; 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/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c541db..af62f843a691 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1449,7 +1449,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: - aif1 |= WM8904_AIF_LRCLK_INV; + aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b0710d817a65..754f88e1fdab 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/wm8962.c b/sound/soc/codecs/wm8962.c index 730dd0c0f0ab..1ae1f8bd9c36 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -153,6 +153,7 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -794,7 +795,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg) case WM8962_ALC2: case WM8962_THERMAL_SHUTDOWN_STATUS: case WM8962_ADDITIONAL_CONTROL_4: - case WM8962_CLASS_D_CONTROL_1: case WM8962_DC_SERVO_6: case WM8962_INTERRUPT_STATUS_1: case WM8962_INTERRUPT_STATUS_2: @@ -2901,13 +2901,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static int wm8962_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int val; + int val, ret; if (mute) - val = WM8962_DAC_MUTE; + val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT; else val = 0; + /** + * The DAC mute bit is mirrored in two registers, update both to keep + * the register cache consistent. + */ + ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1, + WM8962_DAC_MUTE_ALT, val); + if (ret < 0) + return ret; + return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, WM8962_DAC_MUTE, val); } @@ -3686,6 +3695,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err_enable; + regcache_cache_only(wm8962->regmap, true); + /* The drivers should power up as needed */ regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index a1a5d5294c19..910aafd09d21 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -1954,6 +1954,10 @@ #define WM8962_SPKOUTL_ENA_MASK 0x0040 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_SHIFT 6 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */ +#define WM8962_DAC_MUTE_ALT 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_MASK 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_SHIFT 4 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_WIDTH 1 /* DAC_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_MASK 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_SHIFT 1 /* SPKOUTL_PGA_MUTE */ diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 837978e16e9d..ded9ed854a1f 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1264,6 +1264,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); + + codec->cache_sync = 1; break; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b649c0b2..d785e46be47c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -964,6 +964,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) file, blocks, pos - firmware->size); out_fw: + regmap_async_complete(regmap); release_firmware(firmware); wm_adsp_buf_free(&buf_list); out: @@ -1073,13 +1074,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return ret; /* Wait for the RAM to start, should be near instantaneous */ - count = 0; - do { + for (count = 0; count < 10; ++count) { ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, &val); if (ret != 0) return ret; - } while (!(val & ADSP2_RAM_RDY) && ++count < 10); + + if (val & ADSP2_RAM_RDY) + break; + + msleep(1); + } if (!(val & ADSP2_RAM_RDY)) { adsp_err(dsp, "Failed to start DSP RAM\n"); @@ -1280,3 +1285,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f5d81b948759..7a0466eb7ede 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -530,6 +530,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, hubs->hp_startup_mode); break; } + break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 81490febac6d..ade9d6379c1b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -632,8 +632,17 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 593a3ea12d4c..489a9abf112b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -263,6 +263,19 @@ static void dw_i2s_shutdown(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, NULL); } +static int dw_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s_write_reg(dev->i2s_base, TXFFR, 1); + else + i2s_write_reg(dev->i2s_base, RXFFR, 1); + + return 0; +} + static int dw_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -294,6 +307,7 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = { .startup = dw_i2s_startup, .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, + .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, }; diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 670b96b0ce2f..dcfd0fae0b35 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -42,7 +42,8 @@ struct imx_pcm_runtime_data { struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; - atomic_t running; + atomic_t playing; + atomic_t capturing; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) @@ -54,7 +55,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct pt_regs regs; unsigned long delta; - if (!atomic_read(&iprtd->running)) + if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing)) return HRTIMER_NORESTART; get_fiq_regs(®s); @@ -122,7 +123,6 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static int fiq_enable; static int imx_pcm_fiq; static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -134,23 +134,27 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - atomic_set(&iprtd->running, 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + atomic_set(&iprtd->playing, 1); + else + atomic_set(&iprtd->capturing, 1); hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), HRTIMER_MODE_REL); - if (++fiq_enable == 1) - enable_fiq(imx_pcm_fiq); - + enable_fiq(imx_pcm_fiq); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - atomic_set(&iprtd->running, 0); - - if (--fiq_enable == 0) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + atomic_set(&iprtd->playing, 0); + else + atomic_set(&iprtd->capturing, 0); + if (!atomic_read(&iprtd->playing) && + !atomic_read(&iprtd->capturing)) disable_fiq(imx_pcm_fiq); - break; + default: return -EINVAL; } @@ -198,7 +202,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd->substream = substream; - atomic_set(&iprtd->running, 0); + atomic_set(&iprtd->playing, 0); + atomic_set(&iprtd->capturing, 0); hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); iprtd->hrt.function = snd_hrtimer_callback; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6f4dd7543e82..95a9b07bbe96 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -757,9 +757,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .startup = pxa_ssp_startup, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 82ebb1a51479..5c9b5e4f94c3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -853,11 +853,9 @@ static int i2s_suspend(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - } + i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); + i2s->suspend_i2scon = readl(i2s->addr + I2SCON); + i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); return 0; } @@ -866,11 +864,9 @@ static int i2s_resume(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); - } + writel(i2s->suspend_i2scon, i2s->addr + I2SCON); + writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); + writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 06a8000aa07b..5e9690c85d8f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -24,6 +24,7 @@ #include <sound/compress_driver.h> #include <sound/soc.h> #include <sound/initval.h> +#include <sound/soc-dpcm.h> static int soc_compr_open(struct snd_compr_stream *cstream) { @@ -75,6 +76,98 @@ out: return ret; } +static int soc_compr_open_fe(struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *fe = cstream->private_data; + struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; + struct snd_soc_platform *platform = fe->platform; + struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *codec_dai = fe->codec_dai; + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list; + int stream; + int ret = 0; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + if (platform->driver->compr_ops && platform->driver->compr_ops->open) { + ret = platform->driver->compr_ops->open(cstream); + if (ret < 0) { + pr_err("compress asoc: can't open platform %s\n", platform->name); + goto out; + } + } + + if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) { + ret = fe->dai_link->compr_ops->startup(cstream); + if (ret < 0) { + pr_err("compress asoc: %s startup failed\n", fe->dai_link->name); + goto machine_err; + } + } + + fe->dpcm[stream].runtime = fe_substream->runtime; + + if (dpcm_path_get(fe, stream, &list) <= 0) { + dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + } + + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + goto fe_err; + } + + dpcm_clear_pending_state(fe, stream); + dpcm_path_put(&list); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + + cpu_dai->active++; + codec_dai->active++; + fe->codec->active++; + + mutex_unlock(&fe->card->mutex); + + return 0; + +fe_err: + if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) + fe->dai_link->compr_ops->shutdown(cstream); +machine_err: + if (platform->driver->compr_ops && platform->driver->compr_ops->free) + platform->driver->compr_ops->free(cstream); +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + return ret; +} + /* * Power down the audio subsystem pmdown_time msecs after close is called. * This is to ensure there are no pops or clicks in between any music tracks @@ -149,8 +242,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } else { rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ @@ -163,6 +257,65 @@ static int soc_compr_free(struct snd_compr_stream *cstream) return 0; } +static int soc_compr_free_fe(struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *fe = cstream->private_data; + struct snd_soc_platform *platform = fe->platform; + struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *codec_dai = fe->codec_dai; + struct snd_soc_dpcm *dpcm; + int stream, ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + stream = SNDRV_PCM_STREAM_PLAYBACK; + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + stream = SNDRV_PCM_STREAM_CAPTURE; + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + fe->codec->active--; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_free(fe, stream); + if (ret < 0) + dev_err(fe->dev, "compressed hw_free failed %d\n", ret); + + ret = dpcm_be_dai_shutdown(fe, stream); + + /* mark FE's links ready to prune */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + else + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + dpcm_be_disconnect(fe, stream); + + fe->dpcm[stream].runtime = NULL; + + if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) + fe->dai_link->compr_ops->shutdown(cstream); + + if (platform->driver->compr_ops && platform->driver->compr_ops->free) + platform->driver->compr_ops->free(cstream); + + mutex_unlock(&fe->card->mutex); + return 0; +} + static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) { @@ -193,6 +346,59 @@ out: return ret; } +static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = cstream->private_data; + struct snd_soc_platform *platform = fe->platform; + int ret = 0, stream; + + if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN || + cmd == SND_COMPR_TRIGGER_DRAIN) { + + if (platform->driver->compr_ops && + platform->driver->compr_ops->trigger) + return platform->driver->compr_ops->trigger(cstream, cmd); + } + + if (cstream->direction == SND_COMPRESS_PLAYBACK) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; + + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + if (platform->driver->compr_ops && platform->driver->compr_ops->trigger) { + ret = platform->driver->compr_ops->trigger(cstream, cmd); + if (ret < 0) + goto out; + } + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_trigger(fe, stream, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; + } + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + return ret; +} + static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_compr_params *params) { @@ -240,6 +446,64 @@ err: return ret; } +static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct snd_soc_pcm_runtime *fe = cstream->private_data; + struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; + struct snd_soc_platform *platform = fe->platform; + int ret = 0, stream; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { + ret = platform->driver->compr_ops->set_params(cstream, params); + if (ret < 0) + goto out; + } + + if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->set_params) { + ret = fe->dai_link->compr_ops->set_params(cstream); + if (ret < 0) + goto out; + } + + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) + goto out; + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) + goto out; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + else + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + return ret; +} + static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_codec *params) { @@ -334,7 +598,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_set_metadata(struct snd_compr_stream *cstream, +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -347,7 +611,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_get_metadata(struct snd_compr_stream *cstream, +static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -359,13 +623,14 @@ static int sst_compr_get_metadata(struct snd_compr_stream *cstream, return ret; } + /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, - .set_metadata = sst_compr_set_metadata, - .get_metadata = sst_compr_get_metadata, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, @@ -374,6 +639,21 @@ static struct snd_compr_ops soc_compr_ops = { .get_codec_caps = soc_compr_get_codec_caps }; +/* ASoC Dynamic Compress operations */ +static struct snd_compr_ops soc_compr_dyn_ops = { + .open = soc_compr_open_fe, + .free = soc_compr_free_fe, + .set_params = soc_compr_set_params_fe, + .get_params = soc_compr_get_params, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, + .trigger = soc_compr_trigger_fe, + .pointer = soc_compr_pointer, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + /* create a new compress */ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { @@ -382,6 +662,7 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_compr *compr; + struct snd_pcm *be_pcm; char new_name[64]; int ret = 0, direction = 0; @@ -409,7 +690,26 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) ret = -ENOMEM; goto compr_err; } - memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + + if (rtd->dai_link->dynamic) { + snprintf(new_name, sizeof(new_name), "(%s)", + rtd->dai_link->stream_name); + + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + 1, 0, &be_pcm); + if (ret < 0) { + dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", + rtd->dai_link->name); + goto compr_err; + } + + rtd->pcm = be_pcm; + rtd->fe_compr = 1; + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); + } else + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); /* Add copy callback for not memory mapped DSPs */ if (platform->driver->compr_ops && platform->driver->compr_ops->copy) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 360638362e98..c2ecb4e01597 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1797,7 +1797,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5cccd7766..7aa26b5178aa 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ccb6be4d658d..7b4bf0cc1f19 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,7 +34,7 @@ #define DPCM_MAX_BE_USERS 8 /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -408,8 +408,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ @@ -757,7 +758,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -853,7 +854,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -875,11 +876,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -949,7 +945,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -974,7 +970,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -983,7 +979,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1021,7 +1017,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1163,7 +1159,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1224,7 +1220,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1289,7 +1285,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1419,7 +1415,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1587,7 +1583,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; @@ -1886,6 +1882,7 @@ int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget) dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); } + dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ if (!fe->cpu_dai->driver->capture.channels_min) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 52af7f6fb37f..540832e9e684 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 551b3c93ce93..2e7d4aca3d7d 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index a5432b10eaca..5c6520b8ec0e 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -117,7 +117,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -126,10 +126,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; |