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ASoC: Merge up fixes

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Needed for the Rockchip driver.
This commit is contained in:
Mark Brown 2022-07-11 15:51:01 +01:00
commit 39c5c44fc4
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0
18 changed files with 149 additions and 114 deletions
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8
Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
View File

@ -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:

2
Documentation/sound/soc/dai.rst
View File

@ -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

4
sound/soc/codecs/arizona.c
View File

@ -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);

8
sound/soc/codecs/cs47l92.c
View File

@ -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,

30
sound/soc/codecs/rt5640.c
View File

@ -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"))

46
sound/soc/codecs/tas2764.c
View File

@ -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 },

6
sound/soc/codecs/tas2764.h
View File

@ -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)

13
sound/soc/codecs/tlv320adcx140.c
View File

@ -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;

17
sound/soc/codecs/wcd9335.c
View File

@ -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;

21
sound/soc/codecs/wm5102.c
View File

@ -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[] = {

21
sound/soc/codecs/wm8998.c
View File

@ -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[] = {

10
sound/soc/intel/boards/sof_rt5682.c
View File

@ -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) {

40
sound/soc/intel/skylake/skl-nhlt.c
View File

@ -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;
}

1
sound/soc/qcom/qdsp6/q6apm.c
View File

@ -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);

1
sound/soc/rockchip/rockchip_i2s.c
View File

@ -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)) {

2
sound/soc/ti/omap-mcbsp-priv.h
View File

@ -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);

14
sound/soc/ti/omap-mcbsp-st.c
View File

@ -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)
{

19
sound/soc/ti/omap-mcbsp.c
View File

@ -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;
}