diff options
| -rw-r--r-- | sound/soc/codecs/ak4458.c | 6 | ||||
| -rw-r--r-- | sound/soc/codecs/rt5631.c | 8 | ||||
| -rw-r--r-- | sound/soc/codecs/rt5659.c | 37 | ||||
| -rw-r--r-- | sound/soc/codecs/tlv320aic26.c | 2 | ||||
| -rw-r--r-- | sound/soc/fsl/fsl_spdif.c | 51 | ||||
| -rw-r--r-- | sound/soc/intel/boards/cht_bsw_rt5672.c | 18 | ||||
| -rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 2 | ||||
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6asm.c | 2 | ||||
| -rw-r--r-- | sound/soc/sti/uniperif.h | 2 | ||||
| -rw-r--r-- | sound/soc/sunxi/sun4i-i2s.c | 10 | ||||
| -rw-r--r-- | sound/soc/sunxi/sun4i-spdif.c | 2 | ||||
| -rw-r--r-- | sound/soc/ux500/ux500_msp_i2s.c | 8 |
12 files changed, 85 insertions, 63 deletions
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index f180cb5dfe4f..39ae089dcd1d 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -405,7 +405,7 @@ static int ak4458_set_dai_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - int nfs, ndt, ret, reg; + int nfs, ndt, reg; int ats; nfs = ak4458->fs; @@ -416,14 +416,14 @@ static int ak4458_set_dai_mute(struct snd_soc_dai *dai, int mute) ndt = att_speed[ats] / (nfs / 1000); if (mute) { - ret = snd_soc_component_update_bits(component, AK4458_01_CONTROL2, 0x01, 1); + snd_soc_component_update_bits(component, AK4458_01_CONTROL2, 0x01, 1); mdelay(ndt); if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); } else { if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 0); - ret = snd_soc_component_update_bits(component, AK4458_01_CONTROL2, 0x01, 0); + snd_soc_component_update_bits(component, AK4458_01_CONTROL2, 0x01, 0); mdelay(ndt); } diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index b5184f0e10e3..653da3eaf355 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -64,7 +64,7 @@ static const struct reg_default rt5631_reg[] = { { RT5631_PSEUDO_SPATL_CTRL, 0x0553 }, }; -/** +/* * rt5631_write_index - write index register of 2nd layer */ static void rt5631_write_index(struct snd_soc_component *component, @@ -74,7 +74,7 @@ static void rt5631_write_index(struct snd_soc_component *component, snd_soc_component_write(component, RT5631_INDEX_DATA, value); } -/** +/* * rt5631_read_index - read index register of 2nd layer */ static unsigned int rt5631_read_index(struct snd_soc_component *component, @@ -359,6 +359,7 @@ static int check_adcr_select(struct snd_soc_dapm_widget *source, /** * onebit_depop_power_stage - auto depop in power stage. + * @component: ASoC component * @enable: power on/off * * When power on/off headphone, the depop sequence is done by hardware. @@ -397,6 +398,7 @@ static void onebit_depop_power_stage(struct snd_soc_component *component, int en /** * onebit_depop_mute_stage - auto depop in mute stage. + * @component: ASoC component * @enable: mute/unmute * * When mute/unmute headphone, the depop sequence is done by hardware. @@ -435,6 +437,7 @@ static void onebit_depop_mute_stage(struct snd_soc_component *component, int ena /** * onebit_depop_power_stage - step by step depop sequence in power stage. + * @component: ASoC component * @enable: power on/off * * When power on/off headphone, the depop sequence is done in step by step. @@ -507,6 +510,7 @@ static void depop_seq_power_stage(struct snd_soc_component *component, int enabl /** * depop_seq_mute_stage - step by step depop sequence in mute stage. + * @component: ASoC component * @enable: mute/unmute * * When mute/unmute headphone, the depop sequence is done in step by step. diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 541fc6f1089b..41e5917b16a5 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -1195,43 +1195,6 @@ static const struct snd_kcontrol_new rt5659_if3_dac_swap_mux = static const struct snd_kcontrol_new rt5659_if3_adc_swap_mux = SOC_DAPM_ENUM("IF3 ADC Swap Source", rt5659_if3_adc_enum); -static const char * const rt5659_asrc_clk_src[] = { - "clk_sysy_div_out", "clk_i2s1_track", "clk_i2s2_track", - "clk_i2s3_track", "clk_sys2", "clk_sys3" -}; - -static unsigned int rt5659_asrc_clk_map_values[] = { - 0, 1, 2, 3, 5, 6, -}; - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_da_sto_asrc_enum, RT5659_ASRC_2, RT5659_DA_STO_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_da_monol_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_L_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_da_monor_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_R_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_ad_sto1_asrc_enum, RT5659_ASRC_2, RT5659_AD_STO1_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_ad_sto2_asrc_enum, RT5659_ASRC_3, RT5659_AD_STO2_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_ad_monol_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_L_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - -static SOC_VALUE_ENUM_SINGLE_DECL( - rt5659_ad_monor_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_R_T_SFT, 0x7, - rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); - static int rt5659_hp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 032b39735643..4569bbc08acb 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -131,7 +131,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, return 0; } -/** +/* * aic26_mute - Mute control to reduce noise when changing audio format */ static int aic26_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9fb95c6ee7ba..37053e8f29d0 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -776,8 +776,8 @@ static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, } /* Get valid good bit from interrupt status register */ -static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int fsl_spdif_rx_vbit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); @@ -791,6 +791,35 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, return 0; } +static int fsl_spdif_tx_vbit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val; + + regmap_read(regmap, REG_SPDIF_SCR, &val); + val = (val & SCR_VAL_MASK) >> SCR_VAL_OFFSET; + val = 1 - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} + +static int fsl_spdif_tx_vbit_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val = (1 - ucontrol->value.integer.value[0]) << SCR_VAL_OFFSET; + + regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_VAL_MASK, val); + + return 0; +} + /* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -948,11 +977,21 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { /* Valid bit error controller */ { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 V-Bit Errors", + .name = "IEC958 RX V-Bit Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = fsl_spdif_vbit_info, + .get = fsl_spdif_rx_vbit_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 TX V-Bit", .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = fsl_spdif_vbit_info, - .get = fsl_spdif_vbit_get, + .get = fsl_spdif_tx_vbit_get, + .put = fsl_spdif_tx_vbit_put, }, /* DPLL lock info get controller */ { @@ -985,6 +1024,10 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); + /*Clear the val bit for Tx*/ + regmap_update_bits(spdif_private->regmap, REG_SPDIF_SCR, + SCR_VAL_MASK, SCR_VAL_CLEAR); + return 0; } diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 1470c3de7895..1812b786d33b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,13 +253,17 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot. One board/design, - * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The - * second piggy-backed, output-only codec is inside the keyboard-dock - * (which has extra speakers). Unlike the main rt5672 codec, we cannot - * configure this codec, it is hard coded to use 2 channel 24 bit I2S. - * Since we only support 2 channels anyways, there is no need for TDM - * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. + * The default mode for the cpu-dai is TDM 4 slot. The default mode + * for the codec-dai is I2S. So we need to either set the cpu-dai to + * I2S mode to match the codec-dai, or set the codec-dai to TDM 4 slot + * (or program both to yet another mode). + * One board, the Lenovo Miix 2 10, uses not 1 but 2 codecs connected + * to SSP2. The second piggy-backed, output-only codec is inside the + * keyboard-dock (which has extra speakers). Unlike the main rt5672 + * codec, we cannot configure this codec, it is hard coded to use + * 2 channel 24 bit I2S. For this to work we must use I2S mode on this + * board. Since we only support 2 channels anyways, there is no need + * for TDM on any cht-bsw-rt5672 designs. So we use I2S 2ch everywhere. */ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6a72cc1665b7..d1e09ade0190 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -178,7 +178,7 @@ static int pxa_ssp_resume(struct snd_soc_component *component) #define pxa_ssp_resume NULL #endif -/** +/* * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index e0983970cba9..755062eadcc8 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -912,9 +912,9 @@ err: /** * q6asm_open_write() - Open audio client for writing - * * @ac: audio client pointer * @format: audio sample format + * @codec_profile: compressed format profile * @bits_per_sample: bits per sample * * Return: Will be an negative value on error or zero on success diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2dc2da5d458b..a16adeb7c1e9 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1348,7 +1348,7 @@ struct sti_uniperiph_data { struct sti_uniperiph_dai dai_data; }; -static const struct snd_pcm_hardware uni_tdm_hw = { +static __maybe_unused const struct snd_pcm_hardware uni_tdm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index d0a8d5810c0a..f23ff29e7c1d 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -128,13 +128,21 @@ struct sun4i_i2s; /** * struct sun4i_i2s_quirks - Differences between SoC variants. - * * @has_reset: SoC needs reset deasserted. * @reg_offset_txdata: offset of the tx fifo. * @sun4i_i2s_regmap: regmap config to use. * @field_clkdiv_mclk_en: regmap field to enable mclk output. * @field_fmt_wss: regmap field to set word select size. * @field_fmt_sr: regmap field to set sample resolution. + * @bclk_dividers: bit clock dividers array + * @num_bclk_dividers: number of bit clock dividers + * @mclk_dividers: mclk dividers array + * @num_mclk_dividers: number of mclk dividers + * @get_bclk_parent_rate: callback to get bclk parent rate + * @get_sr: callback to get sample resolution + * @get_wss: callback to get word select size + * @set_chan_cfg: callback to set channel configuration + * @set_fmt: callback to set format */ struct sun4i_i2s_quirks { bool has_reset; diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index 86779a99df75..326dd45e39da 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -167,7 +167,7 @@ /** * struct sun4i_spdif_quirks - Differences between SoC variants. * - * @reg_dac_tx_data: TX FIFO offset for DMA config. + * @reg_dac_txdata: TX FIFO offset for DMA config. * @has_reset: SoC needs reset deasserted. * @val_fctl_ftx: TX FIFO flush bitmask. */ diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 394d8b2a4a16..fd0b88bb7921 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -395,7 +395,7 @@ static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) static void flush_fifo_rx(struct ux500_msp *msp) { - u32 reg_val_DR, reg_val_GCR, reg_val_FLR; + u32 reg_val_GCR, reg_val_FLR; u32 limit = 32; reg_val_GCR = readl(msp->registers + MSP_GCR); @@ -403,7 +403,7 @@ static void flush_fifo_rx(struct ux500_msp *msp) reg_val_FLR = readl(msp->registers + MSP_FLR); while (!(reg_val_FLR & RX_FIFO_EMPTY) && limit--) { - reg_val_DR = readl(msp->registers + MSP_DR); + readl(msp->registers + MSP_DR); reg_val_FLR = readl(msp->registers + MSP_FLR); } @@ -412,7 +412,7 @@ static void flush_fifo_rx(struct ux500_msp *msp) static void flush_fifo_tx(struct ux500_msp *msp) { - u32 reg_val_TSTDR, reg_val_GCR, reg_val_FLR; + u32 reg_val_GCR, reg_val_FLR; u32 limit = 32; reg_val_GCR = readl(msp->registers + MSP_GCR); @@ -421,7 +421,7 @@ static void flush_fifo_tx(struct ux500_msp *msp) reg_val_FLR = readl(msp->registers + MSP_FLR); while (!(reg_val_FLR & TX_FIFO_EMPTY) && limit--) { - reg_val_TSTDR = readl(msp->registers + MSP_TSTDR); + readl(msp->registers + MSP_TSTDR); reg_val_FLR = readl(msp->registers + MSP_FLR); } writel(0x0, msp->registers + MSP_ITCR); |