diff options
author | Mark Brown <broonie@kernel.org> | 2021-04-23 21:01:00 +0300 |
---|---|---|
committer | Mark Brown <broonie@kernel.org> | 2021-04-23 21:01:00 +0300 |
commit | d143a69fd452a047440391fcbe290ff416b14ab5 (patch) | |
tree | 29e6786b204f13d057637e16b9cb9a11116c92ea /sound/soc | |
parent | bf05bf16c76bb44ab5156223e1e58e26dfe30a88 (diff) | |
parent | a89f3a93cd20f77ac1f84089297258d4b409e280 (diff) | |
download | linux-d143a69fd452a047440391fcbe290ff416b14ab5.tar.xz |
Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/amd/raven/acp3x-i2s.c | 6 | ||||
-rw-r--r-- | sound/soc/amd/raven/acp3x-pcm-dma.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/adau17x1.c | 13 | ||||
-rw-r--r-- | sound/soc/codecs/rt1011.c | 16 | ||||
-rw-r--r-- | sound/soc/codecs/rt1011.h | 1 | ||||
-rw-r--r-- | sound/soc/codecs/rt286.c | 5 | ||||
-rw-r--r-- | sound/soc/codecs/tas2552.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic32x4.c | 12 | ||||
-rw-r--r-- | sound/soc/intel/boards/kbl_da7219_max98927.c | 38 | ||||
-rw-r--r-- | sound/soc/intel/keembay/kmb_platform.c | 5 | ||||
-rw-r--r-- | sound/soc/meson/axg-frddr.c | 27 |
11 files changed, 84 insertions, 49 deletions
diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 5bc028692fcf..2cd93887410c 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -264,8 +264,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = { .playback = { .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rate_min = 8000, @@ -274,8 +273,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = { .capture = { .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 2, .rate_min = 8000, diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 417cda24030c..f22bb2bdf527 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -24,8 +24,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_96000, @@ -45,8 +44,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 546ee8178038..8aae7ab74091 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -553,6 +553,7 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, { struct adau *adau = snd_soc_component_get_drvdata(dai->component); unsigned int ctrl0, ctrl1; + unsigned int ctrl0_mask; int lrclk_pol; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -612,8 +613,16 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, if (lrclk_pol) ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; - regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); - regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + /* Set the mask to update all relevant bits in ADAU17X1_SERIAL_PORT0 */ + ctrl0_mask = ADAU17X1_SERIAL_PORT0_MASTER | + ADAU17X1_SERIAL_PORT0_LRCLK_POL | + ADAU17X1_SERIAL_PORT0_BCLK_POL | + ADAU17X1_SERIAL_PORT0_PULSE_MODE; + + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0_mask, + ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, ctrl1); adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 098ecf13814d..bfe045367db3 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2239,18 +2239,9 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) dc_offset |= (value & 0xffff); dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); - /* check the package info. */ - regmap_read(rt1011->regmap, RT1011_EFUSE_MATCH_DONE, &value); - if (value & 0x4) - rt1011->pack_id = 1; - if (cali_flag) { - if (rt1011->pack_id) - regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x292c); - else - regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925); - + regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925); /* Class D on */ regmap_write(rt1011->regmap, RT1011_CLASS_D_POS, 0x010e); regmap_write(rt1011->regmap, @@ -2376,10 +2367,7 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_r0_load(rt1011); } - if (rt1011->pack_id) - snd_soc_component_write(component, RT1011_ADC_SET_1, 0x292c); - else - snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925); + snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925); } static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index f3a9a96640f1..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -692,7 +692,6 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; - unsigned int pack_id; /* 0: WLCSP; 1: QFN */ }; #endif /* end of _RT1011_H_ */ diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8abe232ca4a4..e16e7237156f 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -171,6 +171,9 @@ static bool rt286_readable_register(struct device *dev, unsigned int reg) case RT286_PROC_COEF: case RT286_SET_AMP_GAIN_ADC_IN1: case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_GPIO_MASK: + case RT286_SET_GPIO_DIRECTION: + case RT286_SET_GPIO_DATA: case RT286_SET_POWER(RT286_DAC_OUT1): case RT286_SET_POWER(RT286_DAC_OUT2): case RT286_SET_POWER(RT286_ADC_IN1): @@ -1204,7 +1207,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); if (!rt286->pdata.gpio2_en) - regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x40); else regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index bd00c35116cd..700baa6314aa 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -730,8 +730,10 @@ static int tas2552_probe(struct i2c_client *client, ret = devm_snd_soc_register_component(&client->dev, &soc_component_dev_tas2552, tas2552_dai, ARRAY_SIZE(tas2552_dai)); - if (ret < 0) + if (ret < 0) { dev_err(&client->dev, "Failed to register component: %d\n", ret); + pm_runtime_get_noresume(&client->dev); + } return ret; } diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f04f88c8d425..b689f26fc4be 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -577,12 +577,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { .window_start = 0, .window_len = 128, .range_min = 0, - .range_max = AIC32X4_RMICPGAVOL, + .range_max = AIC32X4_REFPOWERUP, }, }; const struct regmap_config aic32x4_regmap_config = { - .max_register = AIC32X4_RMICPGAVOL, + .max_register = AIC32X4_REFPOWERUP, .ranges = aic32x4_regmap_pages, .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), }; @@ -1243,6 +1243,10 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) if (ret) goto err_disable_regulators; + ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); + if (ret) + goto err_disable_regulators; + ret = devm_snd_soc_register_component(dev, &soc_component_dev_aic32x4, &aic32x4_dai, 1); if (ret) { @@ -1250,10 +1254,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) goto err_disable_regulators; } - ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); - if (ret) - goto err_disable_regulators; - return 0; err_disable_regulators: diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index cc9a2509ace2..e0149cf6127d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -282,12 +282,34 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ + /* * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE, * where as kblda7219m98927 & kblmax98927 supports S16_LE by default. * Skipping the port wise FE and BE configuration for kblda7219m98373 & @@ -309,9 +331,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -322,7 +344,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 0fd1e8f62c89..96741c7c0fba 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -105,14 +105,15 @@ static unsigned int kmb_pcm_tx_fn(struct kmb_i2s_info *kmb_i2s, void *buf = runtime->dma_area; int i; + if (kmb_i2s->iec958_fmt) + hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr); + /* KMB i2s uses two separate L/R FIFO */ for (i = 0; i < kmb_i2s->fifo_th; i++) { if (kmb_i2s->config.data_width == 16) { writel(((u16(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0)); writel(((u16(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0)); } else { - if (kmb_i2s->iec958_fmt) - hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr); writel(((u32(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0)); writel(((u32(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0)); } diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index c3ae8ac30745..37f4bb3469b5 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -11,6 +11,7 @@ #include <linux/regmap.h> #include <linux/module.h> #include <linux/of_platform.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -46,11 +47,28 @@ static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream, return 0; } +static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int period, depth, val; + + period = params_period_bytes(params); + + /* Trim the FIFO depth if the period is small to improve latency */ + depth = min(period, fifo->depth); + val = (depth / AXG_FIFO_BURST) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, + CTRL1_FRDDR_DEPTH(val)); + + return 0; +} + static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); - unsigned int val; int ret; /* Enable pclk to access registers and clock the fifo ip */ @@ -61,11 +79,6 @@ static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, /* Apply single buffer mode to the interface */ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); - /* Use all fifo depth */ - val = (fifo->depth / AXG_FIFO_BURST) - 1; - regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, - CTRL1_FRDDR_DEPTH(val)); - return 0; } @@ -84,6 +97,7 @@ static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, } static const struct snd_soc_dai_ops axg_frddr_ops = { + .hw_params = axg_frddr_dai_hw_params, .startup = axg_frddr_dai_startup, .shutdown = axg_frddr_dai_shutdown, }; @@ -157,6 +171,7 @@ static const struct axg_fifo_match_data axg_frddr_match_data = { static const struct snd_soc_dai_ops g12a_frddr_ops = { .prepare = g12a_frddr_dai_prepare, + .hw_params = axg_frddr_dai_hw_params, .startup = axg_frddr_dai_startup, .shutdown = axg_frddr_dai_shutdown, }; |