diff options
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/codecs/arizona.c | 14 | ||||
-rw-r--r-- | sound/soc/codecs/arizona.h | 18 | ||||
-rw-r--r-- | sound/soc/codecs/cs4271.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l52.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/lm49453.c | 106 | ||||
-rw-r--r-- | sound/soc/codecs/sgtl5000.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/sta529.c | 9 | ||||
-rw-r--r-- | sound/soc/codecs/wm2000.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm2200.c | 11 | ||||
-rw-r--r-- | sound/soc/codecs/wm5100.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/wm5102.c | 51 | ||||
-rw-r--r-- | sound/soc/codecs/wm5110.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/wm_adsp.c | 25 | ||||
-rw-r--r-- | sound/soc/fsl/Kconfig | 9 | ||||
-rw-r--r-- | sound/soc/fsl/Makefile | 5 | ||||
-rw-r--r-- | sound/soc/fsl/imx-pcm.c | 3 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 35 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 12 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 1 |
19 files changed, 189 insertions, 137 deletions
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b..ef62c43 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mode = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mode = 1; - break; case SND_SOC_DAIFMT_I2S: mode = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mode = 3; - break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); @@ -691,7 +685,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, } sr_val = i; - lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + lrclk = rates[bclk] / params_rate(params); arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); @@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); @@ -1087,6 +1082,9 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, id, ret); } + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + return 0; } EXPORT_SYMBOL_GPL(arizona_init_fll); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 41dae1e..4deebeb 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 4f11279..ac8742a 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int amutec_eq_bmutec = 0; + bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); - if (!of_get_property(codec->dev->of_node, + if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) - amutec_eq_bmutec = 1; + amutec_eq_bmutec = true; } #endif diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..9811a54 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = { static int cs42l52_get_clk(int mclk, int rate) { - int i, ret = 0; + int i, ret = -EINVAL; u_int mclk1, mclk2 = 0; for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { @@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) - return -EINVAL; return ret; } diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d..e19490c 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, }; @@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv), @@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4..40c07be 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f..12bcae6 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); if (wm2000->speech_clarity) - ret &= ~WM2000_SPEECH_CLARITY; - else ret |= WM2000_SPEECH_CLARITY; + else + ret &= ~WM2000_SPEECH_CLARITY; wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31d..d8c65f5 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1019,8 +1019,6 @@ static const char *wm2200_mixer_texts[] = { "EQR", "LHPF1", "LHPF2", - "LHPF3", - "LHPF4", "DSP1.1", "DSP1.2", "DSP1.3", @@ -1053,7 +1051,6 @@ static int wm2200_mixer_values[] = { 0x25, 0x50, /* EQ */ 0x51, - 0x52, 0x60, /* LHPF1 */ 0x61, /* LHPF2 */ 0x68, /* DSP1 */ @@ -1566,15 +1563,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: fmt_val = 0; break; - case SND_SOC_DAIFMT_DSP_B: - fmt_val = 1; - break; case SND_SOC_DAIFMT_I2S: fmt_val = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - fmt_val = 3; - break; default: dev_err(codec->dev, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); @@ -1626,7 +1617,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 5a5f369..54397a5 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mask = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mask = 1; - break; case SND_SOC_DAIFMT_I2S: mask = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mask = 3; - break; default: dev_err(codec->dev, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 688ade0..1440b3f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -36,6 +36,9 @@ struct wm5102_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int spk_ena:2; + unsigned int spk_ena_pending:1; }; static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); @@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), }; +static int wm5102_spk_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 wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); + + if (arizona->rev < 1) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (!wm5102->spk_ena) { + snd_soc_write(codec, 0x4f5, 0x25a); + wm5102->spk_ena_pending = true; + } + break; + case SND_SOC_DAPM_POST_PMU: + if (wm5102->spk_ena_pending) { + msleep(75); + snd_soc_write(codec, 0x4f5, 0xda); + wm5102->spk_ena_pending = false; + wm5102->spk_ena++; + } + break; + case SND_SOC_DAPM_PRE_PMD: + wm5102->spk_ena--; + if (!wm5102->spk_ena) + snd_soc_write(codec, 0x4f5, 0x25a); + break; + case SND_SOC_DAPM_POST_PMD: + if (!wm5102->spk_ena) + snd_soc_write(codec, 0x4f5, 0x0da); + break; + } + + return 0; +} + + ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); @@ -852,8 +896,7 @@ static const unsigned int wm5102_aec_loopback_values[] = { static const struct soc_enum wm5102_aec_loopback = SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_SRC_SHIFT, - ARIZONA_AEC_LOOPBACK_SRC_MASK, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, ARRAY_SIZE(wm5102_aec_loopback_texts), wm5102_aec_loopback_texts, wm5102_aec_loopback_values); @@ -1034,10 +1077,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ae80c8c..7a09096 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -344,8 +344,7 @@ static const unsigned int wm5110_aec_loopback_values[] = { static const struct soc_enum wm5110_aec_loopback = SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_SRC_SHIFT, - ARIZONA_AEC_LOOPBACK_SRC_MASK, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, ARRAY_SIZE(wm5110_aec_loopback_texts), wm5110_aec_loopback_texts, wm5110_aec_loopback_values); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ffc89fa..b6b6548 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL | GFP_DMA); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; + void *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -384,7 +396,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) hdr = (void*)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); - return -EINVAL; + goto out_fw; } adsp_dbg(dsp, "%s: v%d.%d.%d\n", file, @@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + buf = kmemdup(blk->data, le32_to_cpu(blk->len), + GFP_KERNEL | GFP_DMA); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { @@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } + + kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3b98159..a210c8d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,18 +108,13 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM - tristate - config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ - select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_DMAENGINE_PCM - select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index afd3479..ec14579 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,7 +41,10 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index d5cd9ef..0c9f188 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,7 +31,6 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -80,7 +79,6 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } -EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -102,7 +100,6 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } -EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 91d592f..2370063 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card, INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); ret = device_add(rtd->dev); if (ret < 0) { + /* calling put_device() here to free the rtd->dev */ + put_device(rtd->dev); dev_err(card->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; @@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { device_remove_file(rtd->dev, &dev_attr_codec_reg); - device_del(rtd->dev); + device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; @@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1e36bc8..258acad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, true); + ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to bypass %s: %d\n", @@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, false); + ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to unbypass %s: %d\n", @@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } + + if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + ret = regulator_allow_bypass(w->regulator, true); + if (ret != 0) + dev_warn(w->dapm->dev, + "ASoC: Failed to unbypass %s: %d\n", + w->name, ret); + } break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d7711fc..cf191e6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; 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