Commit 7f04f3ed authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes over several drivers, but all are driver-
  specific and nothing looks scary.

  Slightly large changes are seen in ASoC qcom driver for the bugs that
  were revealed by the recent ASoC core change to report the invalid
  register access errors. Also ASoC fsl got a slight intensive change
  for the distortion fix.

  Others are only trivial fixes or device-specific quirks"

* tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
  ALSA: hda: avoid reset of sdo_limit
  ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
  ALSA: usb-audio: ignore broken processing/extension unit
  ASoC: intel: Fix memleak in sst_media_open
  ASoC: wm8994: Avoid attempts to read unreadable registers
  ASoC: msm8916-wcd-analog: fix register Interrupt offset
  ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
  ALS...
parents 43d387a4 b90b925f
......@@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
INIT_LIST_HEAD(&bus->hlink_list);
init_waitqueue_head(&bus->rirb_wq);
bus->irq = -1;
/*
* Default value of '8' is as per the HD audio specification (Rev 1.0a).
* Following relation is used to derive STRIPE control value.
* For sample rate <= 48K:
* { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
* For sample rate > 48K:
* { ((num_channels * bits_per_sample * rate/48000) /
* number of SDOs) >= 8 }
*/
bus->sdo_limit = 8;
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_init);
......
......@@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
bus->chip_init = true;
/*
* Default value of '8' is as per the HD audio specification (Rev 1.0a).
* Following relation is used to derive STRIPE control value.
* For sample rate <= 48K:
* { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
* For sample rate > 48K:
* { ((num_channels * bits_per_sample * rate/48000) /
* number of SDOs) >= 8 }
*/
bus->sdo_limit = 8;
return true;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);
......
......@@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu)
}
/*
* This is apparently the standard way to initailise an MPU-401
* This is apparently the standard way to initialise an MPU-401
*/
static inline void initialise_mpu401(const struct snd_mpu401 *mpu)
{
......@@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c)
}
/*
* Tell the SoundScape to begin a DMA tranfer using the given channel.
* Tell the SoundScape to begin a DMA transfer using the given channel.
* All locking issues are left to the caller.
*/
static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
......@@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu)
}
/*
* Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
* Initialise an MPU-401 subdevice for MIDI support on the SoundScape.
*/
static int create_mpu401(struct snd_card *card, int devnum,
unsigned long port, int irq)
......
......@@ -7694,6 +7694,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
......@@ -7955,6 +7957,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
{}
};
#define ALC225_STANDARD_PINS \
......
......@@ -138,7 +138,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream,
srate = params_rate(params);
for_each_rtd_codec_dais(rtd, i, codec_dai) {
if (strcmp(codec_dai->component->name, "rt1015-aif"))
if (strcmp(codec_dai->name, "rt1015-aif"))
continue;
ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);
if (ret < 0)
......
......@@ -314,40 +314,30 @@ static int acp_pdm_dma_close(struct snd_soc_component *component,
return 0;
}
static int acp_pdm_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
struct pdm_stream_instance *rtd;
int ret;
bool pdm_status;
unsigned int ch_mask;
rtd = substream->runtime->private_data;
switch (params_channels(params)) {
ret = 0;
switch (substream->runtime->channels) {
case TWO_CH:
ch_mask = 0x00;
break;
default:
return -EINVAL;
}
rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS);
rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base +
ACP_WOV_PDM_DECIMATION_FACTOR);
return 0;
}
static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
struct pdm_stream_instance *rtd;
int ret;
bool pdm_status;
rtd = substream->runtime->private_data;
ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS);
rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base +
ACP_WOV_PDM_DECIMATION_FACTOR);
rtd->bytescount = acp_pdm_get_byte_count(rtd,
substream->stream);
pdm_status = check_pdm_dma_status(rtd->acp_base);
......@@ -369,7 +359,6 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream,
}
static struct snd_soc_dai_ops acp_pdm_dai_ops = {
.hw_params = acp_pdm_dai_hw_params,
.trigger = acp_pdm_dai_trigger,
};
......
......@@ -19,8 +19,8 @@
#define CDC_D_REVISION1 (0xf000)
#define CDC_D_PERPH_SUBTYPE (0xf005)
#define CDC_D_INT_EN_SET (0x015)
#define CDC_D_INT_EN_CLR (0x016)
#define CDC_D_INT_EN_SET (0xf015)
#define CDC_D_INT_EN_CLR (0xf016)
#define MBHC_SWITCH_INT BIT(7)
#define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6)
#define MBHC_BUTTON_PRESS_DET BIT(5)
......
......@@ -412,8 +412,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct wm8994 *control = dev_get_drvdata(component->dev->parent);
int i;
if (control->type != WM8958)
return 0;
switch (event) {
case SND_SOC_DAPM_POST_PMU:
case SND_SOC_DAPM_PRE_PMU:
......
......@@ -151,7 +151,6 @@ static const struct reg_default wm8962_reg[] = {
{ 40, 0x0000 }, /* R40 - SPKOUTL volume */
{ 41, 0x0000 }, /* R41 - SPKOUTR volume */
{ 48, 0x0000 }, /* R48 - Additional control(4) */
{ 49, 0x0010 }, /* R49 - Class D Control 1 */
{ 51, 0x0003 }, /* R51 - Class D Control 2 */
......@@ -842,6 +841,7 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg)
case WM8962_SPKOUTL_VOLUME:
case WM8962_SPKOUTR_VOLUME:
case WM8962_THERMAL_SHUTDOWN_STATUS:
case WM8962_ADDITIONAL_CONTROL_4:
case WM8962_CLASS_D_CONTROL_1:
case WM8962_CLASS_D_CONTROL_2:
case WM8962_CLOCKING_4:
......
......@@ -43,10 +43,12 @@
#define WM8994_NUM_DRC 3
#define WM8994_NUM_EQ 3
static struct {
struct wm8994_reg_mask {
unsigned int reg;
unsigned int mask;
} wm8994_vu_bits[] = {
};
static struct wm8994_reg_mask wm8994_vu_bits[] = {
{ WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU },
{ WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU },
{ WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU },
......@@ -60,14 +62,10 @@ static struct {
{ WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU },
{ WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU },
{ WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU },
{ WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU },
{ WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU },
{ WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU },
{ WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU },
{ WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU },
......@@ -76,6 +74,14 @@ static struct {
{ WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU },
};
/* VU bitfields for ADC2, DAC2 not available on WM1811 */
static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = {
{ WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
};
static int wm8994_drc_base[] = {
WM8994_AIF1_DRC1_1,
WM8994_AIF1_DRC2_1,
......@@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component)
return true;
}
static void wm8994_update_vu_bits(struct snd_soc_component *component)
{
struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component);
struct wm8994 *control = wm8994->wm8994;
int i;
for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
snd_soc_component_write(component, wm8994_vu_bits[i].reg,
snd_soc_component_read(component,
wm8994_vu_bits[i].reg));
if (control->type == WM1811)
return;
for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++)
snd_soc_component_write(component,
wm8994_adc2_dac2_vu_bits[i].reg,
snd_soc_component_read(component,
wm8994_adc2_dac2_vu_bits[i].reg));
}
static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable)
{
struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component);
......@@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component);
struct wm8994 *control = wm8994->wm8994;
int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
int ret, i;
int ret;
int dac;
int adc;
int val;
......@@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
break;
case SND_SOC_DAPM_POST_PMU:
for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
snd_soc_component_write(component, wm8994_vu_bits[i].reg,
snd_soc_component_read(component,
wm8994_vu_bits[i].reg));
wm8994_update_vu_bits(component);
break;
case SND_SOC_DAPM_PRE_PMD:
......@@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
int ret, i;
int ret;
int dac;
int adc;
int val;
......@@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w,
break;
case SND_SOC_DAPM_POST_PMU:
for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
snd_soc_component_write(component, wm8994_vu_bits[i].reg,
snd_soc_component_read(component,
wm8994_vu_bits[i].reg));
wm8994_update_vu_bits(component);
break;
case SND_SOC_DAPM_PRE_PMD:
......@@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component)
wm8994_vu_bits[i].mask,
wm8994_vu_bits[i].mask);
if (control->type != WM1811) {
for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++)
snd_soc_component_update_bits(component,
wm8994_adc2_dac2_vu_bits[i].reg,
wm8994_adc2_dac2_vu_bits[i].mask,
wm8994_adc2_dac2_vu_bits[i].mask);
}
/* Set the low bit of the 3D stereo depth so TLV matches */
snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2,
1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT,
......
......@@ -73,6 +73,7 @@ struct cpu_priv {
* @codec_priv: CODEC private data
* @cpu_priv: CPU private data
* @card: ASoC card structure
* @streams: Mask of current active streams
* @sample_rate: Current sample rate
* @sample_format: Current sample format
* @asrc_rate: ASRC sample rate used by Back-Ends
......@@ -89,6 +90,7 @@ struct fsl_asoc_card_priv {
struct codec_priv codec_priv;
struct cpu_priv cpu_priv;
struct snd_soc_card card;
u8 streams;
u32 sample_rate;
snd_pcm_format_t sample_format;
u32 asrc_rate;
......@@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct codec_priv *codec_priv = &priv->codec_priv;
struct cpu_priv *cpu_priv = &priv->cpu_priv;
struct device *dev = rtd->card->dev;
unsigned int pll_out;
int ret;
priv->sample_rate = params_rate(params);
priv->sample_format = params_format(params);
priv->streams |= BIT(substream->stream);
/*
* If codec-dai is DAI Master and all configurations are already in the
* set_bias_level(), bypass the remaining settings in hw_params().
* Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
*/
if ((priv->card.set_bias_level &&
priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
fsl_asoc_card_is_ac97(priv))
if (fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
......@@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set sysclk for cpu dai\n");
return ret;
goto fail;
}
if (cpu_priv->slot_width) {
......@@ -182,6 +180,68 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
cpu_priv->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set TDM slot for cpu dai\n");
goto fail;
}
}
/* Specific configuration for PLL */
if (codec_priv->pll_id && codec_priv->fll_id) {
if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
pll_out = priv->sample_rate * 384;
else
pll_out = priv->sample_rate * 256;
ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
codec_priv->pll_id,
codec_priv->mclk_id,
codec_priv->mclk_freq, pll_out);
if (ret) {
dev_err(dev, "failed to start FLL: %d\n", ret);
goto fail;
}
ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
codec_priv->fll_id,
pll_out, SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set SYSCLK: %d\n", ret);
goto fail;
}
}
return 0;
fail:
priv->streams &= ~BIT(substream->stream);
return ret;
}
static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = rtd->card->dev;
int ret;
priv->streams &= ~BIT(substream->stream);
if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
/* Force freq to be 0 to avoid error message in codec */
ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
codec_priv->mclk_id,
0,
SND_SOC_CLOCK_IN);
if (ret) {
dev_err(dev, "failed to switch away from FLL: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
codec_priv->pll_id, 0, 0, 0);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to stop FLL: %d\n", ret);
return ret;
}
}
......@@ -191,6 +251,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
static const struct snd_soc_ops fsl_asoc_card_ops = {
.hw_params = fsl_asoc_card_hw_params,
.hw_free = fsl_asoc_card_hw_free,
};
static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
......@@ -254,75 +315,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
},
};
static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai *codec_dai;
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = card->dev;
unsigned int pll_out;
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
break;
if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
pll_out = priv->sample_rate * 384;
else
pll_out = priv->sample_rate * 256;
ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
codec_priv->mclk_id,
codec_priv->mclk_freq, pll_out);
if (ret) {
dev_err(dev, "failed to start FLL: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
pll_out, SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set SYSCLK: %d\n", ret);
return ret;
}
break;
case SND_SOC_BIAS_STANDBY:
if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
break;
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
codec_priv->mclk_freq,
SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to switch away from FLL: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
if (ret) {
dev_err(dev, "failed to stop FLL: %d\n", ret);
return ret;
}
break;
default:
break;
}
return 0;
}
static int fsl_asoc_card_audmux_init(struct device_node *np,
struct fsl_asoc_card_priv *priv)
{
......@@ -611,7 +603,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name = "cs42888";
priv->card.set_bias_level = NULL;
priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
......@@ -628,26 +619,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
codec_dai_name = "wm8962";
priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
priv->codec_priv.pll_id = WM8962_FLL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
codec_dai_name = "wm8960-hifi";
priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name = "ac97-hifi";
priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_AC97;
priv->card.dapm_routes = audio_map_ac97;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
codec_dai_name = "fsl-mqs-dai";
priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
SND_SOC_DAIFMT_CBS_CFS |
SND_SOC_DAIFMT_NB_NF;
......@@ -657,7 +644,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
codec_dai_name = "wm8524-hifi";
priv->card.set_bias_level = NULL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
priv->dai_link[1].dpcm_capture = 0;
priv->dai_link[2].dpcm_capture = 0;
......
......@@ -339,7 +339,6 @@ static int psc_dma_new(struct snd_soc_component *component,
static void psc_dma_free(struct snd_soc_component *component,
struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;
struct snd_pcm_substream *substream;
int stream;
......
......@@ -331,7 +331,7 @@ static int sst_media_open(struct snd_pcm_substream *substream,
ret_val = power_up_sst(stream);
if (ret_val < 0)
return ret_val;
goto out_power_up;
/* Make sure, that the period size is always even */
snd_pcm_hw_constraint_step(substream->runtime, 0,
......@@ -340,8 +340,9 @@ static int sst_media_open(struct snd_pcm_substream *substream,
return snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
out_ops:
kfree(stream);
mutex_unlock(&sst_lock);
out_power_up:
kfree(stream);
return ret_val;
}
......
......@@ -1150,206 +1150,206 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
}
static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0),