Merge tag 'asoc-fix-v5.12-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.12
Quite a lot of mostly platform specific fixes here, the only one which
is generic is a fix for regressions on devices with more complex
clocking support with simple-card. There's also a few new device IDs
and platform quirks.
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
index 50449b6..4454aca 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
@@ -21,6 +21,10 @@
- fsl,vf610-spdif
- fsl,imx6sx-spdif
- fsl,imx8qm-spdif
+ - fsl,imx8qxp-spdif
+ - fsl,imx8mq-spdif
+ - fsl,imx8mm-spdif
+ - fsl,imx8mn-spdif
reg:
maxItems: 1
diff --git a/drivers/spi/spi-cadence-quadspi.c b/drivers/spi/spi-cadence-quadspi.c
index 442cc7c..52ddb32 100644
--- a/drivers/spi/spi-cadence-quadspi.c
+++ b/drivers/spi/spi-cadence-quadspi.c
@@ -1433,6 +1433,7 @@ static int cqspi_probe(struct platform_device *pdev)
cqspi = spi_master_get_devdata(master);
cqspi->pdev = pdev;
+ platform_set_drvdata(pdev, cqspi);
/* Obtain configuration from OF. */
ret = cqspi_of_get_pdata(cqspi);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e4cf14e..1c87b42 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -186,7 +186,6 @@
imply SND_SOC_SI476X
imply SND_SOC_SIMPLE_AMPLIFIER
imply SND_SOC_SIMPLE_MUX
- imply SND_SOC_SIRF_AUDIO_CODEC
imply SND_SOC_SPDIF
imply SND_SOC_SSM2305
imply SND_SOC_SSM2518
@@ -1279,10 +1278,6 @@
tristate "Simple Audio Mux"
select GPIOLIB
-config SND_SOC_SIRF_AUDIO_CODEC
- tristate "SiRF SoC internal audio codec"
- select REGMAP_MMIO
-
config SND_SOC_SPDIF
tristate "S/PDIF CODEC"
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 472caad..85a1d00 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -812,6 +812,7 @@ static const struct of_device_id ak4458_of_match[] = {
{ .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata},
{ },
};
+MODULE_DEVICE_TABLE(of, ak4458_of_match);
static struct i2c_driver ak4458_i2c_driver = {
.driver = {
diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index 8a32b01..85bdd05 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -419,6 +419,7 @@ static const struct of_device_id ak5558_i2c_dt_ids[] __maybe_unused = {
{ .compatible = "asahi-kasei,ak5558"},
{ }
};
+MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids);
static struct i2c_driver ak5558_i2c_driver = {
.driver = {
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 210fcbe..811b7b1 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = {
};
static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
-static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false);
+static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
static const char * const cs42l42_hpf_freq_text[] = {
"1.86Hz", "120Hz", "235Hz", "466Hz"
@@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
CS42L42_DAC_HPF_EN_SHIFT, true, false),
SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL,
CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT,
- 0x3e, 1, mixer_tlv)
+ 0x3f, 1, mixer_tlv)
};
static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w,
@@ -511,43 +511,6 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = {
{"HP", NULL, "HPDRV"}
};
-static int cs42l42_set_bias_level(struct snd_soc_component *component,
- enum snd_soc_bias_level level)
-{
- struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
- int ret;
-
- switch (level) {
- case SND_SOC_BIAS_ON:
- break;
- case SND_SOC_BIAS_PREPARE:
- break;
- case SND_SOC_BIAS_STANDBY:
- if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
- regcache_cache_only(cs42l42->regmap, false);
- regcache_sync(cs42l42->regmap);
- ret = regulator_bulk_enable(
- ARRAY_SIZE(cs42l42->supplies),
- cs42l42->supplies);
- if (ret != 0) {
- dev_err(component->dev,
- "Failed to enable regulators: %d\n",
- ret);
- return ret;
- }
- }
- break;
- case SND_SOC_BIAS_OFF:
-
- regcache_cache_only(cs42l42->regmap, true);
- regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies),
- cs42l42->supplies);
- break;
- }
-
- return 0;
-}
-
static int cs42l42_component_probe(struct snd_soc_component *component)
{
struct cs42l42_private *cs42l42 =
@@ -560,7 +523,6 @@ static int cs42l42_component_probe(struct snd_soc_component *component)
static const struct snd_soc_component_driver soc_component_dev_cs42l42 = {
.probe = cs42l42_component_probe,
- .set_bias_level = cs42l42_set_bias_level,
.dapm_widgets = cs42l42_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(cs42l42_dapm_widgets),
.dapm_routes = cs42l42_audio_map,
@@ -691,24 +653,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
CS42L42_CLK_OASRC_SEL_MASK,
CS42L42_CLK_OASRC_SEL_12 <<
CS42L42_CLK_OASRC_SEL_SHIFT);
- /* channel 1 on low LRCLK, 32 bit */
- snd_soc_component_update_bits(component,
- CS42L42_ASP_RX_DAI0_CH1_AP_RES,
- CS42L42_ASP_RX_CH_AP_MASK |
- CS42L42_ASP_RX_CH_RES_MASK,
- (CS42L42_ASP_RX_CH_AP_LOW <<
- CS42L42_ASP_RX_CH_AP_SHIFT) |
- (CS42L42_ASP_RX_CH_RES_32 <<
- CS42L42_ASP_RX_CH_RES_SHIFT));
- /* Channel 2 on high LRCLK, 32 bit */
- 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,
- (CS42L42_ASP_RX_CH_AP_HI <<
- CS42L42_ASP_RX_CH_AP_SHIFT) |
- (CS42L42_ASP_RX_CH_RES_32 <<
- CS42L42_ASP_RX_CH_RES_SHIFT));
if (pll_ratio_table[i].mclk_src_sel == 0) {
/* Pass the clock straight through */
snd_soc_component_update_bits(component,
@@ -797,27 +741,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* Bitclock/frame inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
+ asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
break;
case SND_SOC_DAIFMT_NB_IF:
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_LCPOL_IN_SHIFT;
+ asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
+ asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
break;
case SND_SOC_DAIFMT_IB_NF:
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
break;
case SND_SOC_DAIFMT_IB_IF:
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_LCPOL_IN_SHIFT;
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
+ asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
break;
}
- snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG,
- CS42L42_ASP_MODE_MASK |
- CS42L42_ASP_SCPOL_IN_DAC_MASK |
- CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val);
+ snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK |
+ CS42L42_ASP_SCPOL_MASK |
+ CS42L42_ASP_LCPOL_MASK,
+ asp_cfg_val);
return 0;
}
@@ -828,14 +768,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_component *component = dai->component;
struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
- int retval;
+ unsigned int width = (params_width(params) / 8) - 1;
+ unsigned int val = 0;
cs42l42->srate = params_rate(params);
- cs42l42->swidth = params_width(params);
- retval = cs42l42_pll_config(component);
+ switch(substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ val |= width << CS42L42_ASP_RX_CH_RES_SHIFT;
+ /* channel 1 on low LRCLK */
+ snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES,
+ CS42L42_ASP_RX_CH_AP_MASK |
+ CS42L42_ASP_RX_CH_RES_MASK, val);
+ /* Channel 2 on high LRCLK */
+ val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT;
+ 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);
+ break;
+ default:
+ break;
+ }
- return retval;
+ return cs42l42_pll_config(component);
}
static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
@@ -900,9 +855,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction)
return 0;
}
-#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
- SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
- SNDRV_PCM_FMTBIT_S32_LE)
+#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE )
static const struct snd_soc_dai_ops cs42l42_ops = {
@@ -1801,7 +1756,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
}
- mdelay(3);
+ usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
/* Request IRQ */
ret = devm_request_threaded_irq(&i2c_client->dev,
@@ -1926,6 +1881,7 @@ static int cs42l42_runtime_resume(struct device *dev)
}
gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+ usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
regcache_cache_only(cs42l42->regmap, false);
regcache_sync(cs42l42->regmap);
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9e3cc52..866d7c8 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -258,11 +258,12 @@
#define CS42L42_ASP_SLAVE_MODE 0x00
#define CS42L42_ASP_MODE_SHIFT 4
#define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT)
-#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2
-#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT)
-#define CS42L42_ASP_LCPOL_IN_SHIFT 0
-#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT)
-#define CS42L42_ASP_POL_INV 1
+#define CS42L42_ASP_SCPOL_SHIFT 2
+#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT)
+#define CS42L42_ASP_SCPOL_NOR 3
+#define CS42L42_ASP_LCPOL_SHIFT 0
+#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT)
+#define CS42L42_ASP_LCPOL_INV 3
#define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08)
#define CS42L42_ASP_STP_SHIFT 4
@@ -739,6 +740,7 @@
#define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16)
#define CS42L42_NUM_SUPPLIES 5
+#define CS42L42_BOOT_TIME_US 3000
static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
"VA",
@@ -756,7 +758,6 @@ struct cs42l42_private {
struct completion pdn_done;
u32 sclk;
u32 srate;
- u32 swidth;
u8 plug_state;
u8 hs_type;
u8 ts_inv;
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index d632055..067757d 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
1, 1, TLV_DB_SCALE_ITEM(0, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(250, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(450, 0, 0),
- 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0),
- 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0),
- 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0),
- 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0),
- 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0),
- 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0),
- 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0),
+ 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0),
+ 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0),
);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv,
diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index c9c21d22..8c04b3b 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -2895,7 +2895,7 @@ static int rx_macro_enable_echo(struct snd_soc_dapm_widget *w,
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
u16 val, ec_hq_reg;
- int ec_tx;
+ int ec_tx = -1;
val = snd_soc_component_read(component,
CDC_RX_INP_MUX_RX_MIX_CFG4);
diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c
index 91e6890..3d6976a 100644
--- a/sound/soc/codecs/lpass-va-macro.c
+++ b/sound/soc/codecs/lpass-va-macro.c
@@ -189,7 +189,6 @@ struct va_macro {
struct device *dev;
unsigned long active_ch_mask[VA_MACRO_MAX_DAIS];
unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS];
- unsigned long active_decimator[VA_MACRO_MAX_DAIS];
u16 dmic_clk_div;
int dec_mode[VA_MACRO_NUM_DECIMATORS];
@@ -549,11 +548,9 @@ static int va_macro_tx_mixer_put(struct snd_kcontrol *kcontrol,
if (enable) {
set_bit(dec_id, &va->active_ch_mask[dai_id]);
va->active_ch_cnt[dai_id]++;
- va->active_decimator[dai_id] = dec_id;
} else {
clear_bit(dec_id, &va->active_ch_mask[dai_id]);
va->active_ch_cnt[dai_id]--;
- va->active_decimator[dai_id] = -1;
}
snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, update);
@@ -880,18 +877,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
struct va_macro *va = snd_soc_component_get_drvdata(component);
u16 tx_vol_ctl_reg, decimator;
- decimator = va->active_decimator[dai->id];
-
- tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL +
- VA_MACRO_TX_PATH_OFFSET * decimator;
- if (mute)
- snd_soc_component_update_bits(component, tx_vol_ctl_reg,
- CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
- CDC_VA_TX_PATH_PGA_MUTE_EN);
- else
- snd_soc_component_update_bits(component, tx_vol_ctl_reg,
- CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
- CDC_VA_TX_PATH_PGA_MUTE_DISABLE);
+ for_each_set_bit(decimator, &va->active_ch_mask[dai->id],
+ VA_MACRO_DEC_MAX) {
+ tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL +
+ VA_MACRO_TX_PATH_OFFSET * decimator;
+ if (mute)
+ snd_soc_component_update_bits(component, tx_vol_ctl_reg,
+ CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
+ CDC_VA_TX_PATH_PGA_MUTE_EN);
+ else
+ snd_soc_component_update_bits(component, tx_vol_ctl_reg,
+ CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
+ CDC_VA_TX_PATH_PGA_MUTE_DISABLE);
+ }
return 0;
}
diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c
index 5ebcd93..9ca49a1 100644
--- a/sound/soc/codecs/lpass-wsa-macro.c
+++ b/sound/soc/codecs/lpass-wsa-macro.c
@@ -1211,14 +1211,16 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
- u16 gain_reg;
+ u16 path_reg, gain_reg;
int val;
- switch (w->reg) {
- case CDC_WSA_RX0_RX_PATH_MIX_CTL:
+ switch (w->shift) {
+ case WSA_MACRO_RX_MIX0:
+ path_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL;
gain_reg = CDC_WSA_RX0_RX_VOL_MIX_CTL;
break;
- case CDC_WSA_RX1_RX_PATH_MIX_CTL:
+ case WSA_MACRO_RX_MIX1:
+ path_reg = CDC_WSA_RX1_RX_PATH_MIX_CTL;
gain_reg = CDC_WSA_RX1_RX_VOL_MIX_CTL;
break;
default:
@@ -1231,7 +1233,7 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w,
snd_soc_component_write(component, gain_reg, val);
break;
case SND_SOC_DAPM_POST_PMD:
- snd_soc_component_update_bits(component, w->reg,
+ snd_soc_component_update_bits(component, path_reg,
CDC_WSA_RX_PATH_MIX_CLK_EN_MASK,
CDC_WSA_RX_PATH_MIX_CLK_DISABLE);
break;
@@ -2068,14 +2070,14 @@ static const struct snd_soc_dapm_widget wsa_macro_dapm_widgets[] = {
SND_SOC_DAPM_MUX("WSA_RX0 INP0", SND_SOC_NOPM, 0, 0, &rx0_prim_inp0_mux),
SND_SOC_DAPM_MUX("WSA_RX0 INP1", SND_SOC_NOPM, 0, 0, &rx0_prim_inp1_mux),
SND_SOC_DAPM_MUX("WSA_RX0 INP2", SND_SOC_NOPM, 0, 0, &rx0_prim_inp2_mux),
- SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", CDC_WSA_RX0_RX_PATH_MIX_CTL,
- 0, 0, &rx0_mix_mux, wsa_macro_enable_mix_path,
+ SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX0,
+ 0, &rx0_mix_mux, wsa_macro_enable_mix_path,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MUX("WSA_RX1 INP0", SND_SOC_NOPM, 0, 0, &rx1_prim_inp0_mux),
SND_SOC_DAPM_MUX("WSA_RX1 INP1", SND_SOC_NOPM, 0, 0, &rx1_prim_inp1_mux),
SND_SOC_DAPM_MUX("WSA_RX1 INP2", SND_SOC_NOPM, 0, 0, &rx1_prim_inp2_mux),
- SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", CDC_WSA_RX1_RX_PATH_MIX_CTL,
- 0, 0, &rx1_mix_mux, wsa_macro_enable_mix_path,
+ SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX1,
+ 0, &rx1_mix_mux, wsa_macro_enable_mix_path,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER_E("WSA_RX INT0 MIX", SND_SOC_NOPM, 0, 0, NULL, 0,
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 37b5795..844e407 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg)
case RT1015_VENDOR_ID:
case RT1015_DEVICE_ID:
case RT1015_PRO_ALT:
+ case RT1015_MAN_I2C:
case RT1015_DAC3:
case RT1015_VBAT_TEST_OUT1:
case RT1015_VBAT_TEST_OUT2:
@@ -513,6 +514,7 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015)
msleep(300);
regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0008);
regmap_write(regmap, RT1015_SYS_RST1, 0x05F5);
+ regmap_write(regmap, RT1015_CLK_DET, 0x8000);
regcache_cache_bypass(regmap, false);
regcache_mark_dirty(regmap);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 1414ad15..a5674c2 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg)
}
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index d198e19..e59fdc8 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg)
}
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 41e5917..91a4ef7 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int
{
struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
unsigned int reg_val = 0;
+ int ret;
if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src)
return 0;
switch (clk_id) {
case RT5659_SCLK_S_MCLK:
+ ret = clk_set_rate(rt5659->mclk, freq);
+ if (ret)
+ return ret;
+
reg_val |= RT5659_SCLK_SRC_MCLK;
break;
case RT5659_SCLK_S_PLL1:
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index c29317e..4063aac 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -629,21 +629,69 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA,
static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA,
RT5670_IF2_ADC_SEL_SFT, rt5670_data_select);
+/*
+ * For reliable output-mute LED control we need a "DAC1 Playback Switch" control.
+ * We emulate this by only clearing the RT5670_M_DAC1_L/_R AD_DA_MIXER register
+ * bits when both our emulated DAC1 Playback Switch control and the DAC1 MIXL/R
+ * DAPM-mixer DAC1 input are enabled.
+ */
+static void rt5670_update_ad_da_mixer_dac1_m_bits(struct rt5670_priv *rt5670)
+{
+ int val = RT5670_M_DAC1_L | RT5670_M_DAC1_R;
+
+ if (rt5670->dac1_mixl_dac1_switch && rt5670->dac1_playback_switch_l)
+ val &= ~RT5670_M_DAC1_L;
+
+ if (rt5670->dac1_mixr_dac1_switch && rt5670->dac1_playback_switch_r)
+ val &= ~RT5670_M_DAC1_R;
+
+ regmap_update_bits(rt5670->regmap, RT5670_AD_DA_MIXER,
+ RT5670_M_DAC1_L | RT5670_M_DAC1_R, val);
+}
+
+static int rt5670_dac1_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+ struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+ ucontrol->value.integer.value[0] = rt5670->dac1_playback_switch_l;
+ ucontrol->value.integer.value[1] = rt5670->dac1_playback_switch_r;
+
+ return 0;
+}
+
+static int rt5670_dac1_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+ struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+ if (rt5670->dac1_playback_switch_l == ucontrol->value.integer.value[0] &&
+ rt5670->dac1_playback_switch_r == ucontrol->value.integer.value[1])
+ return 0;
+
+ rt5670->dac1_playback_switch_l = ucontrol->value.integer.value[0];
+ rt5670->dac1_playback_switch_r = ucontrol->value.integer.value[1];
+
+ rt5670_update_ad_da_mixer_dac1_m_bits(rt5670);
+
+ return 1;
+}
+
static const struct snd_kcontrol_new rt5670_snd_controls[] = {
/* Headphone Output Volume */
- SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL,
- RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
39, 1, out_vol_tlv),
/* OUTPUT Control */
- SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1,
- RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1),
SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1,
RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv),
/* DAC Digital Volume */
SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL,
RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1),
+ SOC_DOUBLE_EXT("DAC1 Playback Switch", SND_SOC_NOPM, 0, 1, 1, 0,
+ rt5670_dac1_playback_switch_get, rt5670_dac1_playback_switch_put),
SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL,
RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
175, 0, dac_vol_tlv),
@@ -913,18 +961,44 @@ static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = {
RT5670_M_MONO_ADC_R2_SFT, 1, 1),
};
+/* See comment above rt5670_update_ad_da_mixer_dac1_m_bits() */
+static int rt5670_put_dac1_mix_dac1_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol);
+ struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+ int ret;
+
+ if (mc->shift == 0)
+ rt5670->dac1_mixl_dac1_switch = ucontrol->value.integer.value[0];
+ else
+ rt5670->dac1_mixr_dac1_switch = ucontrol->value.integer.value[0];
+
+ /* Apply the update (if any) */
+ ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
+ if (ret == 0)
+ return 0;
+
+ rt5670_update_ad_da_mixer_dac1_m_bits(rt5670);
+
+ return 1;
+}
+
+#define SOC_DAPM_SINGLE_RT5670_DAC1_SW(name, shift) \
+ SOC_SINGLE_EXT(name, SND_SOC_NOPM, shift, 1, 0, \
+ snd_soc_dapm_get_volsw, rt5670_put_dac1_mix_dac1_switch)
+
static const struct snd_kcontrol_new rt5670_dac_l_mix[] = {
SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
RT5670_M_ADCMIX_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
- RT5670_M_DAC1_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 0),
};
static const struct snd_kcontrol_new rt5670_dac_r_mix[] = {
SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
RT5670_M_ADCMIX_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
- RT5670_M_DAC1_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 1),
};
static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = {
@@ -1656,12 +1730,10 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2,
RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL,
- RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix,
- ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
- SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL,
- RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix,
- ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
+ SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0,
+ rt5670_sto1_adc_l_mix, ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
+ SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0,
+ rt5670_sto1_adc_r_mix, ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0,
rt5670_sto2_adc_l_mix,
ARRAY_SIZE(rt5670_sto2_adc_l_mix)),
@@ -2999,6 +3071,16 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
dev_info(&i2c->dev, "quirk JD mode 3\n");
}
+ /*
+ * Enable the emulated "DAC1 Playback Switch" by default to avoid
+ * muting the output with older UCM profiles.
+ */
+ rt5670->dac1_playback_switch_l = true;
+ rt5670->dac1_playback_switch_r = true;
+ /* The Power-On-Reset values for the DAC1 mixer have the DAC1 input enabled. */
+ rt5670->dac1_mixl_dac1_switch = true;
+ rt5670->dac1_mixr_dac1_switch = true;
+
rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap);
if (IS_ERR(rt5670->regmap)) {
ret = PTR_ERR(rt5670->regmap);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index 56b13fe..6fb3c36 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -212,12 +212,8 @@
/* global definition */
#define RT5670_L_MUTE (0x1 << 15)
#define RT5670_L_MUTE_SFT 15
-#define RT5670_VOL_L_MUTE (0x1 << 14)
-#define RT5670_VOL_L_SFT 14
#define RT5670_R_MUTE (0x1 << 7)
#define RT5670_R_MUTE_SFT 7
-#define RT5670_VOL_R_MUTE (0x1 << 6)
-#define RT5670_VOL_R_SFT 6
#define RT5670_L_VOL_MASK (0x3f << 8)
#define RT5670_L_VOL_SFT 8
#define RT5670_R_VOL_MASK (0x3f)
@@ -2017,6 +2013,11 @@ struct rt5670_priv {
int dsp_rate;
int jack_type;
int jack_type_saved;
+
+ bool dac1_mixl_dac1_switch;
+ bool dac1_mixr_dac1_switch;
+ bool dac1_playback_switch_l;
+ bool dac1_playback_switch_r;
};
void rt5670_jack_suspend(struct snd_soc_component *component);
diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 85f7441..047f4e6 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component)
return 0;
}
+static void rt711_remove(struct snd_soc_component *component)
+{
+ struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component);
+
+ regcache_cache_only(rt711->regmap, true);
+}
+
static const struct snd_soc_component_driver soc_codec_dev_rt711 = {
.probe = rt711_probe,
.set_bias_level = rt711_set_bias_level,
@@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = {
.dapm_routes = rt711_audio_map,
.num_dapm_routes = ARRAY_SIZE(rt711_audio_map),
.set_jack = rt711_set_jack_detect,
+ .remove = rt711_remove,
};
static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 73551e3..6d9bb25 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
{ SGTL5000_DAP_EQ_BASS_BAND4, 0x002f },
{ SGTL5000_DAP_MAIN_CHAN, 0x8000 },
{ SGTL5000_DAP_MIX_CHAN, 0x0000 },
- { SGTL5000_DAP_AVC_CTRL, 0x0510 },
+ { SGTL5000_DAP_AVC_CTRL, 0x5100 },
{ SGTL5000_DAP_AVC_THRESHOLD, 0x1473 },
{ SGTL5000_DAP_AVC_ATTACK, 0x0028 },
{ SGTL5000_DAP_AVC_DECAY, 0x0050 },
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
deleted file mode 100644
index a7fe268..0000000
--- a/sound/soc/codecs/sirf-audio-codec.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-or-later */
-/*
- * SiRF inner codec controllers define
- *
- * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
- */
-
-#ifndef _SIRF_AUDIO_CODEC_H
-#define _SIRF_AUDIO_CODEC_H
-
-
-#define AUDIO_IC_CODEC_PWR (0x00E0)
-#define AUDIO_IC_CODEC_CTRL0 (0x00E4)
-#define AUDIO_IC_CODEC_CTRL1 (0x00E8)
-#define AUDIO_IC_CODEC_CTRL2 (0x00EC)
-#define AUDIO_IC_CODEC_CTRL3 (0x00F0)
-
-#define MICBIASEN (1 << 3)
-
-#define IC_RDACEN (1 << 0)
-#define IC_LDACEN (1 << 1)
-#define IC_HSREN (1 << 2)
-#define IC_HSLEN (1 << 3)
-#define IC_SPEN (1 << 4)
-#define IC_CPEN (1 << 5)
-
-#define IC_HPRSELR (1 << 6)
-#define IC_HPLSELR (1 << 7)
-#define IC_HPRSELL (1 << 8)
-#define IC_HPLSELL (1 << 9)
-#define IC_SPSELR (1 << 10)
-#define IC_SPSELL (1 << 11)
-
-#define IC_MONOR (1 << 12)
-#define IC_MONOL (1 << 13)
-
-#define IC_RXOSRSEL (1 << 28)
-#define IC_CPFREQ (1 << 29)
-#define IC_HSINVEN (1 << 30)
-
-#define IC_MICINREN (1 << 0)
-#define IC_MICINLEN (1 << 1)
-#define IC_MICIN1SEL (1 << 2)
-#define IC_MICIN2SEL (1 << 3)
-#define IC_MICDIFSEL (1 << 4)
-#define IC_LINEIN1SEL (1 << 5)
-#define IC_LINEIN2SEL (1 << 6)
-#define IC_RADCEN (1 << 7)
-#define IC_LADCEN (1 << 8)
-#define IC_ALM (1 << 9)
-
-#define IC_DIGMICEN (1 << 22)
-#define IC_DIGMICFREQ (1 << 23)
-#define IC_ADC14B_12 (1 << 24)
-#define IC_FIRDAC_HSL_EN (1 << 25)
-#define IC_FIRDAC_HSR_EN (1 << 26)
-#define IC_FIRDAC_LOUT_EN (1 << 27)
-#define IC_POR (1 << 28)
-#define IC_CODEC_CLK_EN (1 << 29)
-#define IC_HP_3DB_BOOST (1 << 30)
-
-#define IC_ADC_LEFT_GAIN_SHIFT 16
-#define IC_ADC_RIGHT_GAIN_SHIFT 10
-#define IC_ADC_GAIN_MASK 0x3F
-#define IC_MIC_MAX_GAIN 0x39
-
-#define IC_RXPGAR_MASK 0x3F
-#define IC_RXPGAR_SHIFT 14
-#define IC_RXPGAL_MASK 0x3F
-#define IC_RXPGAL_SHIFT 21
-#define IC_RXPGAR 0x7B
-#define IC_RXPGAL 0x7B
-
-#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F
-#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0
-#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10
-#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20
-
-#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
- << AUDIO_PORT_TX_FIFO_SC_OFFSET)
-#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
- << AUDIO_PORT_TX_FIFO_LC_OFFSET)
-#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
- << AUDIO_PORT_TX_FIFO_HC_OFFSET)
-
-#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F
-#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0
-#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10
-#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20
-
-#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
- << AUDIO_PORT_RX_FIFO_SC_OFFSET)
-#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
- << AUDIO_PORT_RX_FIFO_LC_OFFSET)
-#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
- << AUDIO_PORT_RX_FIFO_HC_OFFSET)
-#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4)
-#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8)
-
-#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC)
-#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100)
-#define AUDIO_PORT_IC_TXFIFO_STS (0x0104)
-#define AUDIO_PORT_IC_TXFIFO_INT (0x0108)
-#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C)
-
-#define AUDIO_PORT_IC_RXFIFO_OP (0x0110)
-#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114)
-#define AUDIO_PORT_IC_RXFIFO_STS (0x0118)
-#define AUDIO_PORT_IC_RXFIFO_INT (0x011C)
-#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120)
-
-#define AUDIO_FIFO_START (1 << 0)
-#define AUDIO_FIFO_RESET (1 << 1)
-
-#define AUDIO_FIFO_FULL (1 << 0)
-#define AUDIO_FIFO_EMPTY (1 << 1)
-#define AUDIO_FIFO_OFLOW (1 << 2)
-#define AUDIO_FIFO_UFLOW (1 << 3)
-
-#define IC_TX_ENABLE (0x03)
-#define IC_RX_ENABLE_MONO (0x01)
-#define IC_RX_ENABLE_STEREO (0x03)
-
-#endif /*__SIRF_AUDIO_CODEC_H*/
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 40f682f..d18ae5e 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -1873,6 +1873,12 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai,
wcd = snd_soc_component_get_drvdata(dai->component);
+ if (tx_num > WCD934X_TX_MAX || rx_num > WCD934X_RX_MAX) {
+ dev_err(wcd->dev, "Invalid tx %d or rx %d channel count\n",
+ tx_num, rx_num);
+ return -EINVAL;
+ }
+
if (!tx_slot || !rx_slot) {
dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n",
tx_slot, rx_slot);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 5781174..ad8af3f 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -878,6 +878,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
{
u32 strcr = 0, scr = 0, stcr, srcr, mask;
+ unsigned int slots;
ssi->dai_fmt = fmt;
@@ -909,10 +910,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
return -EINVAL;
}
+ slots = ssi->slots ? : 2;
regmap_update_bits(ssi->regs, REG_SSI_STCCR,
- SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+ SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
regmap_update_bits(ssi->regs, REG_SSI_SRCCR,
- SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+ SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
/* Data on rising edge of bclk, frame low, 1clk before data */
strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS;
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index ab31045..6cada4c 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -172,15 +172,16 @@ int asoc_simple_parse_clk(struct device *dev,
* or device's module clock.
*/
clk = devm_get_clk_from_child(dev, node, NULL);
- if (IS_ERR(clk))
- clk = devm_get_clk_from_child(dev, dlc->of_node, NULL);
-
if (!IS_ERR(clk)) {
- simple_dai->clk = clk;
simple_dai->sysclk = clk_get_rate(clk);
- } else if (!of_property_read_u32(node, "system-clock-frequency",
- &val)) {
+
+ simple_dai->clk = clk;
+ } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) {
simple_dai->sysclk = val;
+ } else {
+ clk = devm_get_clk_from_child(dev, dlc->of_node, NULL);
+ if (!IS_ERR(clk))
+ simple_dai->sysclk = clk_get_rate(clk);
}
if (of_property_read_bool(node, "system-clock-direction-out"))
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 782f2b4..5d48cc3 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -581,7 +581,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
},
.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
BYT_RT5640_JD_SRC_JD1_IN4P |
- BYT_RT5640_OVCD_TH_1500UA |
+ BYT_RT5640_OVCD_TH_2000UA |
BYT_RT5640_OVCD_SF_0P75 |
BYT_RT5640_MCLK_EN),
},
diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
index f5de1d7..f3bebed 100644
--- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
@@ -555,7 +555,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
/* set tdm */
if (tdm_priv->bck_invert)
- tdm_con |= 1 << BCK_INVERSE_SFT;
+ regmap_update_bits(afe->regmap, AUDIO_TOP_CON3,
+ BCK_INVERSE_MASK_SFT,
+ 0x1 << BCK_INVERSE_SFT);
if (tdm_priv->lck_invert)
tdm_con |= 1 << LRCK_INVERSE_SFT;
diff --git a/sound/soc/mediatek/mt8192/mt8192-reg.h b/sound/soc/mediatek/mt8192/mt8192-reg.h
index 562f25c..b9fb80d 100644
--- a/sound/soc/mediatek/mt8192/mt8192-reg.h
+++ b/sound/soc/mediatek/mt8192/mt8192-reg.h
@@ -21,6 +21,11 @@ enum {
/*****************************************************************************
* R E G I S T E R D E F I N I T I O N
*****************************************************************************/
+/* AUDIO_TOP_CON3 */
+#define BCK_INVERSE_SFT 3
+#define BCK_INVERSE_MASK 0x1
+#define BCK_INVERSE_MASK_SFT (0x1 << 3)
+
/* AFE_DAC_CON0 */
#define VUL12_ON_SFT 31
#define VUL12_ON_MASK 0x1
@@ -2079,9 +2084,6 @@ enum {
#define TDM_EN_SFT 0
#define TDM_EN_MASK 0x1
#define TDM_EN_MASK_SFT (0x1 << 0)
-#define BCK_INVERSE_SFT 1
-#define BCK_INVERSE_MASK 0x1
-#define BCK_INVERSE_MASK_SFT (0x1 << 1)
#define LRCK_INVERSE_SFT 2
#define LRCK_INVERSE_MASK 0x1
#define LRCK_INVERSE_MASK_SFT (0x1 << 2)
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index c642e5f..be360a4 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -739,7 +739,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev,
for_each_child_of_node(dev->of_node, node) {
ret = of_property_read_u32(node, "reg", &id);
- if (ret || id < 0 || id >= data->variant->num_dai) {
+ if (ret || id < 0) {
dev_err(dev, "valid dai id not found: %d\n", ret);
continue;
}
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 6c2760e..153e9b2 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -27,18 +27,18 @@
#define SPK_TDM_RX_MASK 0x03
#define NUM_TDM_SLOTS 8
#define SLIM_MAX_TX_PORTS 16
-#define SLIM_MAX_RX_PORTS 16
+#define SLIM_MAX_RX_PORTS 13
#define WCD934X_DEFAULT_MCLK_RATE 9600000
struct sdm845_snd_data {
struct snd_soc_jack jack;
bool jack_setup;
- bool stream_prepared[SLIM_MAX_RX_PORTS];
+ bool stream_prepared[AFE_PORT_MAX];
struct snd_soc_card *card;
uint32_t pri_mi2s_clk_count;
uint32_t sec_mi2s_clk_count;
uint32_t quat_tdm_clk_count;
- struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS];
+ struct sdw_stream_runtime *sruntime[AFE_PORT_MAX];
};
static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f6d4e99..0cffc95 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -31,6 +31,7 @@
#include <linux/of.h>
#include <linux/of_graph.h>
#include <linux/dmi.h>
+#include <linux/acpi.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -1573,6 +1574,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
if (card->long_name)
return 0; /* long name already set by driver or from DMI */
+ if (!is_acpi_device_node(card->dev->fwnode))
+ return 0;
+
/* make up dmi long name as: vendor-product-version-board */
vendor = dmi_get_system_info(DMI_BOARD_VENDOR);
if (!vendor || !is_dmi_valid(vendor)) {
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 5788fe3..c3b757c 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -207,7 +207,7 @@ int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask)
ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
HDA_DSP_REG_ADSPCS, adspcs,
- !(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)),
+ !(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)),
HDA_DSP_REG_POLL_INTERVAL_US,
HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC);
if (ret < 0)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 1d29b1f..0c096db 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -897,6 +897,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev)
/* dsp_unmap: not currently used */
iounmap(sdev->bar[HDA_DSP_BAR]);
hdac_bus_unmap:
+ platform_device_unregister(hdev->dmic_dev);
iounmap(bus->remap_addr);
hda_codec_i915_exit(sdev);
err:
