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-rw-r--r--sound/soc/codecs/Kconfig9
-rw-r--r--sound/soc/codecs/Makefile5
-rw-r--r--sound/soc/codecs/cs42l42.c104
-rw-r--r--sound/soc/codecs/cs42l42.h3
-rw-r--r--sound/soc/codecs/nau8824.c42
-rw-r--r--sound/soc/codecs/rt5631.c2
-rw-r--r--sound/soc/codecs/rt5682.c9
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c12
-rw-r--r--sound/soc/codecs/tlv320aic31xx.h4
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c60
-rw-r--r--sound/soc/codecs/wcd938x.c18
-rw-r--r--sound/soc/codecs/wm_adsp.c7
12 files changed, 161 insertions, 114 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 9e3e4db3d75d..9ff1600ca823 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1343,7 +1343,7 @@ config SND_SOC_SSM2305
high-efficiency mono Class-D audio power amplifiers.
config SND_SOC_SSM2518
- tristate
+ tristate "Analog Devices SSM2518 Class-D Amplifier"
depends on I2C
config SND_SOC_SSM2602
@@ -1575,7 +1575,9 @@ config SND_SOC_WCD934X
Qualcomm SoCs like SDM845.
config SND_SOC_WCD938X
+ depends on SND_SOC_WCD938X_SDW
tristate
+ depends on SOUNDWIRE || !SOUNDWIRE
config SND_SOC_WCD938X_SDW
tristate "WCD9380/WCD9385 Codec - SDW"
@@ -1831,11 +1833,6 @@ config SND_SOC_ZL38060
which consists of a Digital Signal Processor (DSP), several Digital
Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
-config SND_SOC_ZX_AUD96P22
- tristate "ZTE ZX AUD96P22 CODEC"
- depends on I2C
- select REGMAP_I2C
-
# Amp
config SND_SOC_LM4857
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index d656b1405473..8dcea2c4604a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -585,7 +585,10 @@ obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o
obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o
obj-$(CONFIG_SND_SOC_WCD934X) += snd-soc-wcd934x.o
obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x.o
-obj-$(CONFIG_SND_SOC_WCD938X_SDW) += snd-soc-wcd938x-sdw.o
+ifdef CONFIG_SND_SOC_WCD938X_SDW
+# avoid link failure by forcing sdw code built-in when needed
+obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o
+endif
obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o
obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o
obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index eff013f295be..99c022be94a6 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = {
.use_single_write = true,
};
-static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
+static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true);
static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
static const char * const cs42l42_hpf_freq_text[] = {
@@ -425,34 +425,23 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
CS42L42_ADC_WNF_CF_SHIFT,
cs42l42_wnf3_freq_text);
-static const char * const cs42l42_wnf05_freq_text[] = {
- "280Hz", "315Hz", "350Hz", "385Hz",
- "420Hz", "455Hz", "490Hz", "525Hz"
-};
-
-static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
- CS42L42_ADC_WNF_CF_SHIFT,
- cs42l42_wnf05_freq_text);
-
static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
/* ADC Volume and Filter Controls */
SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL,
- CS42L42_ADC_NOTCH_DIS_SHIFT, true, false),
+ CS42L42_ADC_NOTCH_DIS_SHIFT, true, true),
SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL,
CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false),
SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL,
CS42L42_ADC_INV_SHIFT, true, false),
SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL,
CS42L42_ADC_DIG_BOOST_SHIFT, true, false),
- SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME,
- CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv),
+ SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv),
SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL,
CS42L42_ADC_WNF_EN_SHIFT, true, false),
SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL,
CS42L42_ADC_HPF_EN_SHIFT, true, false),
SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum),
SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum),
- SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum),
/* DAC Volume and Filter Controls */
SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1,
@@ -471,8 +460,8 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HP"),
SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1),
SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0),
- SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH1_SHIFT, 0),
- SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH2_SHIFT, 0),
+ SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, SND_SOC_NOPM, 0, 0),
/* Playback Requirements */
SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0),
@@ -630,6 +619,8 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
if (pll_ratio_table[i].sclk == clk) {
+ cs42l42->pll_config = i;
+
/* Configure the internal sample rate */
snd_soc_component_update_bits(component, CS42L42_MCLK_CTL,
CS42L42_INTERNAL_FS_MASK,
@@ -638,14 +629,9 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
(pll_ratio_table[i].mclk_int !=
24000000)) <<
CS42L42_INTERNAL_FS_SHIFT);
- /* Set the MCLK src (PLL or SCLK) and the divide
- * ratio
- */
+
snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL,
- CS42L42_MCLK_SRC_SEL_MASK |
CS42L42_MCLKDIV_MASK,
- (pll_ratio_table[i].mclk_src_sel
- << CS42L42_MCLK_SRC_SEL_SHIFT) |
(pll_ratio_table[i].mclk_div <<
CS42L42_MCLKDIV_SHIFT));
/* Set up the LRCLK */
@@ -681,15 +667,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
CS42L42_FSYNC_PULSE_WIDTH_MASK,
CS42L42_FRAC1_VAL(fsync - 1) <<
CS42L42_FSYNC_PULSE_WIDTH_SHIFT);
- snd_soc_component_update_bits(component,
- CS42L42_ASP_FRM_CFG,
- CS42L42_ASP_5050_MASK,
- CS42L42_ASP_5050_MASK);
- /* Set the frame delay to 1.0 SCLK clocks */
- snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG,
- CS42L42_ASP_FSD_MASK,
- CS42L42_ASP_FSD_1_0 <<
- CS42L42_ASP_FSD_SHIFT);
/* Set the sample rates (96k or lower) */
snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN,
CS42L42_FS_EN_MASK,
@@ -789,7 +766,18 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- case SND_SOC_DAIFMT_LEFT_J:
+ /*
+ * 5050 mode, frame starts on falling edge of LRCLK,
+ * frame delayed by 1.0 SCLKs
+ */
+ snd_soc_component_update_bits(component,
+ CS42L42_ASP_FRM_CFG,
+ CS42L42_ASP_STP_MASK |
+ CS42L42_ASP_5050_MASK |
+ CS42L42_ASP_FSD_MASK,
+ CS42L42_ASP_5050_MASK |
+ (CS42L42_ASP_FSD_1_0 <<
+ CS42L42_ASP_FSD_SHIFT));
break;
default:
return -EINVAL;
@@ -819,6 +807,25 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return 0;
}
+static int cs42l42_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
+
+ /*
+ * Sample rates < 44.1 kHz would produce an out-of-range SCLK with
+ * a standard I2S frame. If the machine driver sets SCLK it must be
+ * legal.
+ */
+ if (cs42l42->sclk)
+ return 0;
+
+ /* Machine driver has not set a SCLK, limit bottom end to 44.1 kHz */
+ return snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ 44100, 192000);
+}
+
static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -832,6 +839,10 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
cs42l42->srate = params_rate(params);
cs42l42->bclk = snd_soc_params_to_bclk(params);
+ /* I2S frame always has 2 channels even for mono audio */
+ if (channels == 1)
+ cs42l42->bclk *= 2;
+
switch(substream->stream) {
case SNDRV_PCM_STREAM_CAPTURE:
if (channels == 2) {
@@ -855,6 +866,17 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
CS42L42_ASP_RX_CH_AP_MASK |
CS42L42_ASP_RX_CH_RES_MASK, val);
+
+ /* Channel B comes from the last active channel */
+ snd_soc_component_update_bits(component, CS42L42_SP_RX_CH_SEL,
+ CS42L42_SP_RX_CHB_SEL_MASK,
+ (channels - 1) << CS42L42_SP_RX_CHB_SEL_SHIFT);
+
+ /* Both LRCLK slots must be enabled */
+ snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN,
+ CS42L42_ASP_RX0_CH_EN_MASK,
+ BIT(CS42L42_ASP_RX0_CH1_SHIFT) |
+ BIT(CS42L42_ASP_RX0_CH2_SHIFT));
break;
default:
break;
@@ -900,13 +922,21 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
*/
regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq,
ARRAY_SIZE(cs42l42_to_osc_seq));
+
+ /* Must disconnect PLL before stopping it */
+ snd_soc_component_update_bits(component,
+ CS42L42_MCLK_SRC_SEL,
+ CS42L42_MCLK_SRC_SEL_MASK,
+ 0);
+ usleep_range(100, 200);
+
snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
CS42L42_PLL_START_MASK, 0);
}
} else {
if (!cs42l42->stream_use) {
/* SCLK must be running before codec unmute */
- if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) {
+ if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) {
snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
CS42L42_PLL_START_MASK, 1);
@@ -927,6 +957,12 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
CS42L42_PLL_LOCK_TIMEOUT_US);
if (ret < 0)
dev_warn(component->dev, "PLL failed to lock: %d\n", ret);
+
+ /* PLL must be running to drive glitchless switch logic */
+ snd_soc_component_update_bits(component,
+ CS42L42_MCLK_SRC_SEL,
+ CS42L42_MCLK_SRC_SEL_MASK,
+ CS42L42_MCLK_SRC_SEL_MASK);
}
/* Mark SCLK as present, turn off internal oscillator */
@@ -960,8 +996,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
SNDRV_PCM_FMTBIT_S24_LE |\
SNDRV_PCM_FMTBIT_S32_LE )
-
static const struct snd_soc_dai_ops cs42l42_ops = {
+ .startup = cs42l42_dai_startup,
.hw_params = cs42l42_pcm_hw_params,
.set_fmt = cs42l42_set_dai_fmt,
.set_sysclk = cs42l42_set_sysclk,
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 206b3c81d3e0..8734f6828f3e 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -653,6 +653,8 @@
/* Page 0x25 Audio Port Registers */
#define CS42L42_SP_RX_CH_SEL (CS42L42_PAGE_25 + 0x01)
+#define CS42L42_SP_RX_CHB_SEL_SHIFT 2
+#define CS42L42_SP_RX_CHB_SEL_MASK (3 << CS42L42_SP_RX_CHB_SEL_SHIFT)
#define CS42L42_SP_RX_ISOC_CTL (CS42L42_PAGE_25 + 0x02)
#define CS42L42_SP_RX_RSYNC_SHIFT 6
@@ -775,6 +777,7 @@ struct cs42l42_private {
struct gpio_desc *reset_gpio;
struct completion pdn_done;
struct snd_soc_jack *jack;
+ int pll_config;
int bclk;
u32 sclk;
u32 srate;
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 15bd8335f667..db88be48c998 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap)
}
}
-static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin)
-{
- struct snd_soc_dapm_context *dapm = nau8824->dapm;
- const char *prefix = dapm->component->name_prefix;
- char prefixed_pin[80];
-
- if (prefix) {
- snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
- prefix, pin);
- snd_soc_dapm_disable_pin(dapm, prefixed_pin);
- } else {
- snd_soc_dapm_disable_pin(dapm, pin);
- }
-}
-
-static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin)
-{
- struct snd_soc_dapm_context *dapm = nau8824->dapm;
- const char *prefix = dapm->component->name_prefix;
- char prefixed_pin[80];
-
- if (prefix) {
- snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
- prefix, pin);
- snd_soc_dapm_force_enable_pin(dapm, prefixed_pin);
- } else {
- snd_soc_dapm_force_enable_pin(dapm, pin);
- }
-}
-
static void nau8824_eject_jack(struct nau8824 *nau8824)
{
struct snd_soc_dapm_context *dapm = nau8824->dapm;
@@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824)
/* Clear all interruption status */
nau8824_int_status_clear_all(regmap);
- nau8824_dapm_disable_pin(nau8824, "SAR");
- nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+ snd_soc_dapm_disable_pin(dapm, "SAR");
+ snd_soc_dapm_disable_pin(dapm, "MICBIAS");
snd_soc_dapm_sync(dapm);
/* Enable the insertion interruption, disable the ejection
@@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work)
struct regmap *regmap = nau8824->regmap;
int adc_value, event = 0, event_mask = 0;
- nau8824_dapm_enable_pin(nau8824, "MICBIAS");
- nau8824_dapm_enable_pin(nau8824, "SAR");
+ snd_soc_dapm_enable_pin(dapm, "MICBIAS");
+ snd_soc_dapm_enable_pin(dapm, "SAR");
snd_soc_dapm_sync(dapm);
msleep(100);
@@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work)
if (adc_value < HEADSET_SARADC_THD) {
event |= SND_JACK_HEADPHONE;
- nau8824_dapm_disable_pin(nau8824, "SAR");
- nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+ snd_soc_dapm_disable_pin(dapm, "SAR");
+ snd_soc_dapm_disable_pin(dapm, "MICBIAS");
snd_soc_dapm_sync(dapm);
} else {
event |= SND_JACK_HEADSET;
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 3000bc128b5b..38356ea2bd6e 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = {
.reg_defaults = rt5631_reg,
.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
.cache_type = REGCACHE_RBTREE,
+ .use_single_read = true,
+ .use_single_write = true,
};
static int rt5631_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index e4c91571abae..51ecaa2abcd1 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = {
{RT5682_I2C_CTRL, 0x000f},
{RT5682_PLL2_INTERNAL, 0x8266},
{RT5682_SAR_IL_CMD_3, 0x8365},
+ {RT5682_SAR_IL_CMD_6, 0x0180},
};
void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev)
@@ -973,10 +974,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
- if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
- if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 51870d50f419..52d2c968b5c0 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -35,6 +35,9 @@
#include "tlv320aic31xx.h"
+static int aic31xx_set_jack(struct snd_soc_component *component,
+ struct snd_soc_jack *jack, void *data);
+
static const struct reg_default aic31xx_reg_defaults[] = {
{ AIC31XX_CLKMUX, 0x00 },
{ AIC31XX_PLLPR, 0x11 },
@@ -1256,6 +1259,13 @@ static int aic31xx_power_on(struct snd_soc_component *component)
return ret;
}
+ /*
+ * The jack detection configuration is in the same register
+ * that is used to report jack detect status so is volatile
+ * and not covered by the cache sync, restore it separately.
+ */
+ aic31xx_set_jack(component, aic31xx->jack, NULL);
+
return 0;
}
@@ -1604,6 +1614,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
ret);
return ret;
}
+ regcache_cache_only(aic31xx->regmap, true);
+
aic31xx->dev = &i2c->dev;
aic31xx->irq = i2c->irq;
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 81952984613d..2513922a0292 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -151,8 +151,8 @@ struct aic31xx_pdata {
#define AIC31XX_WORD_LEN_24BITS 0x02
#define AIC31XX_WORD_LEN_32BITS 0x03
#define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2)
-#define AIC31XX_BCLK_MASTER BIT(2)
-#define AIC31XX_WCLK_MASTER BIT(3)
+#define AIC31XX_BCLK_MASTER BIT(3)
+#define AIC31XX_WCLK_MASTER BIT(2)
/* AIC31XX_DATA_OFFSET */
#define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index c63b717040ed..2e9175b37dc9 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
/* -12dB min, 0.5dB steps */
static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
-
-static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
static const char * const lo_cm_text[] = {
@@ -682,11 +682,20 @@ static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr)
static int aic32x4_set_processing_blocks(struct snd_soc_component *component,
u8 r_block, u8 p_block)
{
- if (r_block > 18 || p_block > 25)
- return -EINVAL;
+ struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
+
+ if (aic32x4->type == AIC32X4_TYPE_TAS2505) {
+ if (r_block || p_block > 3)
+ return -EINVAL;
- snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
- snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+ snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+ } else { /* AIC32x4 */
+ if (r_block > 18 || p_block > 25)
+ return -EINVAL;
+
+ snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
+ snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+ }
return 0;
}
@@ -695,6 +704,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
unsigned int sample_rate, unsigned int channels,
unsigned int bit_depth)
{
+ struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
u8 aosr;
u16 dosr;
u8 adc_resource_class, dac_resource_class;
@@ -721,19 +731,28 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
adc_resource_class = 6;
dac_resource_class = 8;
dosr_increment = 8;
- aic32x4_set_processing_blocks(component, 1, 1);
+ if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+ aic32x4_set_processing_blocks(component, 0, 1);
+ else
+ aic32x4_set_processing_blocks(component, 1, 1);
} else if (sample_rate <= 96000) {
aosr = 64;
adc_resource_class = 6;
dac_resource_class = 8;
dosr_increment = 4;
- aic32x4_set_processing_blocks(component, 1, 9);
+ if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+ aic32x4_set_processing_blocks(component, 0, 1);
+ else
+ aic32x4_set_processing_blocks(component, 1, 9);
} else if (sample_rate == 192000) {
aosr = 32;
adc_resource_class = 3;
dac_resource_class = 4;
dosr_increment = 2;
- aic32x4_set_processing_blocks(component, 13, 19);
+ if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+ aic32x4_set_processing_blocks(component, 0, 1);
+ else
+ aic32x4_set_processing_blocks(component, 13, 19);
} else {
dev_err(component->dev, "Sampling rate not supported\n");
return -EINVAL;
@@ -1063,21 +1082,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
};
static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
- SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
- AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+ SOC_SINGLE_S8_TLV("PCM Playback Volume",
+ AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm),
SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
- SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
- AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
- tlv_driver_gain),
- SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
- AIC32X4_HPLGAIN, 6, 0x01, 1),
- SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+ SOC_SINGLE_TLV("HP Driver Gain Volume",
+ AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
+ SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
- SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
- 0, 0, 117, 1, tlv_spk_vol),
- SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
- 4, 5, 0, tlv_amp_vol),
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+ TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain),
+ SOC_SINGLE_TLV("Speaker Amplifier Playback Volume",
+ TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol),
+
+ SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
};
static const struct snd_kcontrol_new hp_output_mixer_controls[] = {
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index 78b76eceff8f..2fcc97370be2 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component)
(WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0);
}
- ret = wcd938x_irq_init(wcd938x, component->dev);
- if (ret) {
- dev_err(component->dev, "%s: IRQ init failed: %d\n",
- __func__, ret);
- return ret;
- }
-
wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
WCD938X_IRQ_HPHR_PDM_WD_INT);
wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
@@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
- wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
if (!wcd938x->txdev) {
@@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
- wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
if (!wcd938x->tx_sdw_dev) {
dev_err(dev, "could not get txslave with matching of dev\n");
@@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev)
return PTR_ERR(wcd938x->regmap);
}
+ ret = wcd938x_irq_init(wcd938x, dev);
+ if (ret) {
+ dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
+ return ret;
+ }
+
+ wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
+ wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
+
ret = wcd938x_set_micbias_data(wcd938x);
if (ret < 0) {
dev_err(dev, "%s: bad micbias pdata\n", __func__);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 37aa020f23f6..fe15cbc7bcaf 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -282,6 +282,7 @@
/*
* HALO_CCM_CORE_CONTROL
*/
+#define HALO_CORE_RESET 0x00000200
#define HALO_CORE_EN 0x00000001
/*
@@ -746,7 +747,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp)
{
wm_adsp_debugfs_clear(dsp);
- debugfs_remove_recursive(dsp->debugfs_root);
}
#else
static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
@@ -1213,7 +1213,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
mutex_lock(&ctl->dsp->pwr_lock);
- ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size);
+ ret = wm_coeff_read_ctrl(ctl, ctl->cache, size);
if (!ret && copy_to_user(bytes, ctl->cache, size))
ret = -EFAULT;
@@ -3333,7 +3333,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp)
{
return regmap_update_bits(dsp->regmap,
dsp->base + HALO_CCM_CORE_CONTROL,
- HALO_CORE_EN, HALO_CORE_EN);
+ HALO_CORE_RESET | HALO_CORE_EN,
+ HALO_CORE_RESET | HALO_CORE_EN);
}
static void wm_halo_stop_core(struct wm_adsp *dsp)