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-rw-r--r--Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml8
-rw-r--r--Documentation/sound/soc/dai.rst2
-rw-r--r--sound/soc/codecs/arizona.c4
-rw-r--r--sound/soc/codecs/cs47l92.c8
-rw-r--r--sound/soc/codecs/rt5640.c30
-rw-r--r--sound/soc/codecs/tas2764.c46
-rw-r--r--sound/soc/codecs/tas2764.h6
-rw-r--r--sound/soc/codecs/tlv320adcx140.c13
-rw-r--r--sound/soc/codecs/wcd9335.c17
-rw-r--r--sound/soc/codecs/wm5102.c21
-rw-r--r--sound/soc/codecs/wm8998.c21
-rw-r--r--sound/soc/intel/boards/sof_rt5682.c10
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c40
-rw-r--r--sound/soc/qcom/qdsp6/q6apm.c1
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c1
-rw-r--r--sound/soc/ti/omap-mcbsp-priv.h2
-rw-r--r--sound/soc/ti/omap-mcbsp-st.c14
-rw-r--r--sound/soc/ti/omap-mcbsp.c19
18 files changed, 149 insertions, 114 deletions
diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
index e9a533080b32..ef18a572a1ff 100644
--- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
@@ -25,12 +25,12 @@ properties:
- qcom,sc7280-lpass-cpu
reg:
- minItems: 2
+ minItems: 1
maxItems: 6
description: LPAIF core registers
reg-names:
- minItems: 2
+ minItems: 1
maxItems: 6
clocks:
@@ -42,12 +42,12 @@ properties:
maxItems: 10
interrupts:
- minItems: 2
+ minItems: 1
maxItems: 4
description: LPAIF DMA buffer interrupt
interrupt-names:
- minItems: 2
+ minItems: 1
maxItems: 4
qcom,adsp:
diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
index 009b07e5a0f3..bf8431386d26 100644
--- a/Documentation/sound/soc/dai.rst
+++ b/Documentation/sound/soc/dai.rst
@@ -10,7 +10,7 @@ AC97
====
AC97 is a five wire interface commonly found on many PC sound cards. It is
-now also popular in many portable devices. This DAI has a reset line and time
+now also popular in many portable devices. This DAI has a RESET line and time
multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e32871b3f68a..7434aeeda292 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component,
if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
return true;
- val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE);
- if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
+ val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE);
+ if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK))
return true;
val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1);
diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c
index 59da34b480a8..fe576d64e089 100644
--- a/sound/soc/codecs/cs47l92.c
+++ b/sound/soc/codecs/cs47l92.c
@@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol,
end:
snd_soc_dapm_mutex_unlock(dapm);
- return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+ ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+ if (ret < 0) {
+ dev_err(madera->dev, "Failed to update demux power state: %d\n", ret);
+ return ret;
+ }
+
+ return change;
}
static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 5092856a262d..38ab8d4291c2 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component,
snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000);
snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000);
snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000);
- snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000);
+ if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
+ snd_soc_component_write(component, RT5640_PWR_ANLG1,
+ 0x0018);
+ else
+ snd_soc_component_write(component, RT5640_PWR_ANLG1,
+ 0x0000);
snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000);
break;
@@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work)
static irqreturn_t rt5640_irq(int irq, void *data)
{
struct rt5640_priv *rt5640 = data;
+ int delay = 0;
+
+ if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
+ cancel_delayed_work_sync(&rt5640->jack_work);
+ delay = 100;
+ }
if (rt5640->jack)
- queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
+ queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
return IRQ_HANDLED;
}
@@ -2588,6 +2599,12 @@ static void rt5640_enable_hda_jack_detect(
snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0);
+ snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+ RT5640_PWR_VREF2, RT5640_PWR_VREF2);
+ usleep_range(10000, 15000);
+ snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+ RT5640_PWR_FV2, RT5640_PWR_FV2);
+
rt5640->jack = jack;
ret = request_irq(rt5640->irq, rt5640_irq,
@@ -2707,16 +2724,13 @@ static int rt5640_probe(struct snd_soc_component *component)
if (device_property_read_u32(component->dev,
"realtek,jack-detect-source", &val) == 0) {
- if (val <= RT5640_JD_SRC_GPIO4) {
+ if (val <= RT5640_JD_SRC_GPIO4)
rt5640->jd_src = val << RT5640_JD_SFT;
- } else if (val == RT5640_JD_SRC_HDA_HEADER) {
+ else if (val == RT5640_JD_SRC_HDA_HEADER)
rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER;
- snd_soc_component_update_bits(component, RT5640_DUMMY1,
- 0x0300, 0x0);
- } else {
+ else
dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n",
val);
- }
}
if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted"))
diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index 42f0c1e449ba..846d9d3ecc9d 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764)
gpiod_set_value_cansleep(tas2764->reset_gpio, 0);
msleep(20);
gpiod_set_value_cansleep(tas2764->reset_gpio, 1);
+ usleep_range(1000, 2000);
}
snd_soc_component_write(tas2764->component, TAS2764_SW_RST,
TAS2764_RST);
+ usleep_range(1000, 2000);
}
static int tas2764_set_bias_level(struct snd_soc_component *component,
@@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component)
struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
int ret;
- if (tas2764->sdz_gpio)
+ if (tas2764->sdz_gpio) {
gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+ usleep_range(1000, 2000);
+ }
ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL,
TAS2764_PWR_CTRL_MASK,
@@ -131,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = {
};
static SOC_ENUM_SINGLE_DECL(
- tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src);
+ tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT,
+ tas2764_ASI1_src);
static const struct snd_kcontrol_new tas2764_asi1_mux =
SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum);
@@ -329,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_component *component = dai->component;
struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
- u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0;
- int iface;
+ u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
int ret;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_IF:
+ asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+ fallthrough;
case SND_SOC_DAIFMT_NB_NF:
asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
break;
+ case SND_SOC_DAIFMT_IB_IF:
+ asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+ fallthrough;
case SND_SOC_DAIFMT_IB_NF:
asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
break;
- default:
- dev_err(tas2764->dev, "ASI format Inverse is not found\n");
- return -EINVAL;
}
ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
@@ -353,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
+ asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+ fallthrough;
case SND_SOC_DAIFMT_DSP_A:
- iface = TAS2764_TDM_CFG2_SCFG_I2S;
tdm_rx_start_slot = 1;
break;
case SND_SOC_DAIFMT_DSP_B:
case SND_SOC_DAIFMT_LEFT_J:
- iface = TAS2764_TDM_CFG2_SCFG_LEFT_J;
tdm_rx_start_slot = 0;
break;
default:
@@ -368,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return -EINVAL;
}
- ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
- TAS2764_TDM_CFG1_MASK,
- (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
+ ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0,
+ TAS2764_TDM_CFG0_FRAME_START,
+ asi_cfg_0);
if (ret < 0)
return ret;
- ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2,
- TAS2764_TDM_CFG2_SCFG_MASK, iface);
+ ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
+ TAS2764_TDM_CFG1_MASK,
+ (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
if (ret < 0)
return ret;
@@ -501,8 +509,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
tas2764->component = component;
- if (tas2764->sdz_gpio)
+ if (tas2764->sdz_gpio) {
gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+ usleep_range(1000, 2000);
+ }
tas2764_reset(tas2764);
@@ -526,12 +536,12 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
}
static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1);
static const struct snd_kcontrol_new tas2764_snd_controls[] = {
SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
TAS2764_DVC_MAX, 1, tas2764_playback_volume),
- SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0,
+ SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0,
tas2764_digital_tlv),
};
@@ -555,7 +565,7 @@ static const struct reg_default tas2764_reg_defaults[] = {
{ TAS2764_SW_RST, 0x00 },
{ TAS2764_PWR_CTRL, 0x1a },
{ TAS2764_DVC, 0x00 },
- { TAS2764_CHNL_0, 0x00 },
+ { TAS2764_CHNL_0, 0x28 },
{ TAS2764_TDM_CFG0, 0x09 },
{ TAS2764_TDM_CFG1, 0x02 },
{ TAS2764_TDM_CFG2, 0x0a },
diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 67d6fd903c42..f015f22a083b 100644
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -47,6 +47,7 @@
#define TAS2764_TDM_CFG0_MASK GENMASK(3, 1)
#define TAS2764_TDM_CFG0_44_1_48KHZ BIT(3)
#define TAS2764_TDM_CFG0_88_2_96KHZ (BIT(3) | BIT(1))
+#define TAS2764_TDM_CFG0_FRAME_START BIT(0)
/* TDM Configuration Reg1 */
#define TAS2764_TDM_CFG1 TAS2764_REG(0X0, 0x09)
@@ -66,10 +67,7 @@
#define TAS2764_TDM_CFG2_RXS_16BITS 0x0
#define TAS2764_TDM_CFG2_RXS_24BITS BIT(0)
#define TAS2764_TDM_CFG2_RXS_32BITS BIT(1)
-#define TAS2764_TDM_CFG2_SCFG_MASK GENMASK(5, 4)
-#define TAS2764_TDM_CFG2_SCFG_I2S 0x0
-#define TAS2764_TDM_CFG2_SCFG_LEFT_J BIT(4)
-#define TAS2764_TDM_CFG2_SCFG_RIGHT_J BIT(5)
+#define TAS2764_TDM_CFG2_SCFG_SHIFT 4
/* TDM Configuration Reg3 */
#define TAS2764_TDM_CFG3 TAS2764_REG(0X0, 0x0c)
diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index 6618ac4a7d5c..2844a9d2bc4a 100644
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -33,7 +33,6 @@ struct adcx140_priv {
bool micbias_vg;
unsigned int dai_fmt;
- unsigned int tdm_delay;
unsigned int slot_width;
};
@@ -790,12 +789,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
{
struct snd_soc_component *component = codec_dai->component;
struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
- unsigned int lsb;
- /* TDM based on DSP mode requires slots to be adjacent */
- lsb = __ffs(tx_mask);
- if ((lsb + 1) != __fls(tx_mask)) {
- dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+ /*
+ * The chip itself supports arbitrary masks, but the driver currently
+ * only supports adjacent slots beginning at the first slot.
+ */
+ if (tx_mask != GENMASK(__fls(tx_mask), 0)) {
+ dev_err(component->dev, "Only lower adjacent slots are supported\n");
return -EINVAL;
}
@@ -810,7 +810,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- adcx140->tdm_delay = lsb;
adcx140->slot_width = slot_width;
return 0;
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index 3554b95462e8..beeeb35e8032 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -333,7 +333,7 @@ struct wcd9335_codec {
struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY];
unsigned int rx_port_value[WCD9335_RX_MAX];
- unsigned int tx_port_value;
+ unsigned int tx_port_value[WCD9335_TX_MAX];
int hph_l_gain;
int hph_r_gain;
u32 rx_bias_count;
@@ -1325,8 +1325,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc,
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev);
+ struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc);
+ struct soc_mixer_control *mixer =
+ (struct soc_mixer_control *)kc->private_value;
+ int dai_id = widget->shift;
+ int port_id = mixer->shift;
- ucontrol->value.integer.value[0] = wcd->tx_port_value;
+ ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id;
return 0;
}
@@ -1349,12 +1354,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
case AIF2_CAP:
case AIF3_CAP:
/* only add to the list if value not set */
- if (enable && !(wcd->tx_port_value & BIT(port_id))) {
- wcd->tx_port_value |= BIT(port_id);
+ if (enable && wcd->tx_port_value[port_id] != dai_id) {
+ wcd->tx_port_value[port_id] = dai_id;
list_add_tail(&wcd->tx_chs[port_id].list,
&wcd->dai[dai_id].slim_ch_list);
- } else if (!enable && (wcd->tx_port_value & BIT(port_id))) {
- wcd->tx_port_value &= ~BIT(port_id);
+ } else if (!enable && wcd->tx_port_value[port_id] == dai_id) {
+ wcd->tx_port_value[port_id] = -1;
list_del_init(&wcd->tx_chs[port_id].list);
}
break;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 8b1caac65c3a..af7d324e3352 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+ uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+ int ret = 0;
mutex_lock(&arizona->dac_comp_lock);
- arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+ if (arizona->dac_comp_coeff != dac_comp_coeff) {
+ arizona->dac_comp_coeff = dac_comp_coeff;
+ ret = 1;
+ }
mutex_unlock(&arizona->dac_comp_lock);
- return 0;
+ return ret;
}
static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
@@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+ int ret = 0;
+
+ if (ucontrol->value.integer.value[0] > mc->max)
+ return -EINVAL;
mutex_lock(&arizona->dac_comp_lock);
- arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+ if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) {
+ arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+ ret = 1;
+ }
mutex_unlock(&arizona->dac_comp_lock);
- return 0;
+ return ret;
}
static const char * const wm5102_osr_text[] = {
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 328f1946f584..79fc6bbaa3aa 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int mode_reg, mode_index;
unsigned int mux, inmode, src_val, mode_val;
+ int change, ret;
mux = ucontrol->value.enumerated.item[0];
if (mux > 1)
@@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
snd_soc_component_update_bits(component, mode_reg,
ARIZONA_IN1_MODE_MASK, mode_val);
- snd_soc_component_update_bits(component, e->reg,
- ARIZONA_IN1L_SRC_MASK |
- ARIZONA_IN1L_SRC_SE_MASK,
- src_val);
+ change = snd_soc_component_update_bits(component, e->reg,
+ ARIZONA_IN1L_SRC_MASK |
+ ARIZONA_IN1L_SRC_SE_MASK,
+ src_val);
- return snd_soc_dapm_mux_update_power(dapm, kcontrol,
- ucontrol->value.enumerated.item[0],
- e, NULL);
+ ret = snd_soc_dapm_mux_update_power(dapm, kcontrol,
+ ucontrol->value.enumerated.item[0],
+ e, NULL);
+ if (ret < 0) {
+ dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret);
+ return ret;
+ }
+
+ return change;
}
static const char * const wm8998_inmux_texts[] = {
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index a24fb71d5ff3..1384716c6360 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
static int is_legacy_cpu;
-static struct snd_soc_jack sof_hdmi[3];
-
struct sof_hdmi_pcm {
struct list_head head;
struct snd_soc_dai *codec_dai;
+ struct snd_soc_jack hdmi_jack;
int device;
};
@@ -447,7 +446,6 @@ static int sof_card_late_probe(struct snd_soc_card *card)
char jack_name[NAME_SIZE];
struct sof_hdmi_pcm *pcm;
int err;
- int i = 0;
/* HDMI is not supported by SOF on Baytrail/CherryTrail */
if (is_legacy_cpu || !ctx->idisp_codec)
@@ -468,17 +466,15 @@ static int sof_card_late_probe(struct snd_soc_card *card)
snprintf(jack_name, sizeof(jack_name),
"HDMI/DP, pcm=%d Jack", pcm->device);
err = snd_soc_card_jack_new(card, jack_name,
- SND_JACK_AVOUT, &sof_hdmi[i]);
+ SND_JACK_AVOUT, &pcm->hdmi_jack);
if (err)
return err;
err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
- &sof_hdmi[i]);
+ &pcm->hdmi_jack);
if (err < 0)
return err;
-
- i++;
}
if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) {
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 2439a574ac2f..deb7b820325e 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
struct nhlt_fmt_cfg *fmt_cfg;
struct wav_fmt_ext *wav_fmt;
unsigned long rate;
- bool present = false;
int rate_index = 0;
u16 channels, bps;
u8 clk_src;
@@ -112,9 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
if (fmt->fmt_count == 0)
return;
+ fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
for (i = 0; i < fmt->fmt_count; i++) {
- fmt_cfg = &fmt->fmt_config[i];
- wav_fmt = &fmt_cfg->fmt_ext;
+ struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg;
+ bool present = false;
+
+ wav_fmt = &saved_fmt_cfg->fmt_ext;
channels = wav_fmt->fmt.channels;
bps = wav_fmt->fmt.bits_per_sample;
@@ -132,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
* derive the rate.
*/
for (j = i; j < fmt->fmt_count; j++) {
- fmt_cfg = &fmt->fmt_config[j];
- wav_fmt = &fmt_cfg->fmt_ext;
+ struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg;
+
+ wav_fmt = &tmp_fmt_cfg->fmt_ext;
if ((fs == wav_fmt->fmt.samples_per_sec) &&
- (bps == wav_fmt->fmt.bits_per_sample))
+ (bps == wav_fmt->fmt.bits_per_sample)) {
channels = max_t(u16, channels,
wav_fmt->fmt.channels);
+ saved_fmt_cfg = tmp_fmt_cfg;
+ }
+ /* Move to the next nhlt_fmt_cfg */
+ tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps +
+ tmp_fmt_cfg->config.size);
}
rate = channels * bps * fs;
@@ -153,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
/* Fill rate and parent for sclk/sclkfs */
if (!present) {
+ struct nhlt_fmt_cfg *first_fmt_cfg;
+
+ first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
i2s_config_ext = (struct skl_i2s_config_blob_ext *)
- fmt->fmt_config[0].config.caps;
+ first_fmt_cfg->config.caps;
/* MCLK Divider Source Select */
if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -168,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
parent = skl_get_parent_clk(clk_src);
+ /* Move to the next nhlt_fmt_cfg */
+ fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps +
+ fmt_cfg->config.size);
/*
* Do not copy the config data if there is no parent
* clock available for this clock source select
@@ -176,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
continue;
sclk[id].rate_cfg[rate_index].rate = rate;
- sclk[id].rate_cfg[rate_index].config = fmt_cfg;
+ sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg;
sclkfs[id].rate_cfg[rate_index].rate = rate;
- sclkfs[id].rate_cfg[rate_index].config = fmt_cfg;
+ sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg;
sclk[id].parent_name = parent->name;
sclkfs[id].parent_name = parent->name;
@@ -192,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
{
struct skl_i2s_config_blob_ext *i2s_config_ext;
struct skl_i2s_config_blob_legacy *i2s_config;
- struct nhlt_specific_cfg *fmt_cfg;
+ struct nhlt_fmt_cfg *fmt_cfg;
struct skl_clk_parent_src *parent;
u32 clkdiv, div_ratio;
u8 clk_src;
- fmt_cfg = &fmt->fmt_config[0].config;
- i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps;
+ fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
+ i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps;
/* MCLK Divider Source Select and divider */
if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -227,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
return;
mclk[id].rate_cfg[0].rate = parent->rate/div_ratio;
- mclk[id].rate_cfg[0].config = &fmt->fmt_config[0];
+ mclk[id].rate_cfg[0].config = fmt_cfg;
mclk[id].parent_name = parent->name;
}
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index f424d7aa389a..794019286c70 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui
id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL);
if (id < 0) {
dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id);
+ kfree(graph->graph);
kfree(graph);
mutex_unlock(&apm->lock);
return ERR_PTR(id);
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index ee33c5d2e948..f5f3540a9e18 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -803,7 +803,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
i2s->bclk_ratio = 64;
i2s->pinctrl = devm_pinctrl_get(&pdev->dev);
-
if (!IS_ERR(i2s->pinctrl)) {
i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, "bclk_on");
if (!IS_ERR_OR_NULL(i2s->bclk_on)) {
diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h
index 7865cda4bf0a..da519ea1f303 100644
--- a/sound/soc/ti/omap-mcbsp-priv.h
+++ b/sound/soc/ti/omap-mcbsp-priv.h
@@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg,
/* Sidetone specific API */
int omap_mcbsp_st_init(struct platform_device *pdev);
-void omap_mcbsp_st_cleanup(struct platform_device *pdev);
-
int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp);
int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp);
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 0bc7d26c660a..7e8179cae92e 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
if (!st_data)
return -ENOMEM;
- st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick");
+ st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick");
if (IS_ERR(st_data->mcbsp_iclk)) {
dev_warn(mcbsp->dev,
"Failed to get ick, sidetone might be broken\n");
@@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
if (!st_data->io_base_st)
return -ENOMEM;
- ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group);
+ ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group);
if (ret)
return ret;
@@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
return 0;
}
-void omap_mcbsp_st_cleanup(struct platform_device *pdev)
-{
- struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
-
- if (mcbsp->st_data) {
- sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group);
- clk_put(mcbsp->st_data->mcbsp_iclk);
- }
-}
-
static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 76df0e7844f8..c4ac1f30b9fe 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
- ret = sysfs_create_group(&mcbsp->dev->kobj,
- &additional_attr_group);
+ ret = devm_device_add_group(mcbsp->dev, &additional_attr_group);
if (ret) {
dev_err(mcbsp->dev,
"Unable to create additional controls\n");
@@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
}
}
- ret = omap_mcbsp_st_init(pdev);
- if (ret)
- goto err_st;
-
- return 0;
-
-err_st:
- if (mcbsp->pdata->buffer_size)
- sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
- return ret;
+ return omap_mcbsp_st_init(pdev);
}
/*
@@ -1432,11 +1422,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req))
cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
- if (mcbsp->pdata->buffer_size)
- sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-
- omap_mcbsp_st_cleanup(pdev);
-
return 0;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-26 12:21:59 +0000