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-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.yaml2
-rw-r--r--include/sound/soc.h6
-rw-r--r--sound/soc/amd/acp-da7219-max98357a.c5
-rw-r--r--sound/soc/codecs/Kconfig8
-rw-r--r--sound/soc/codecs/rt5631.c2
-rw-r--r--sound/soc/codecs/rt5682.c8
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c2
-rw-r--r--sound/soc/codecs/tlv320aic31xx.h4
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c27
-rw-r--r--sound/soc/codecs/wcd938x.c18
-rw-r--r--sound/soc/codecs/wm_adsp.c6
-rw-r--r--sound/soc/intel/boards/sof_sdw_max98373.c81
-rw-r--r--sound/soc/soc-pcm.c22
-rw-r--r--sound/soc/sof/intel/pci-tgl.c1
-rw-r--r--sound/soc/tegra/tegra_pcm.c30
-rw-r--r--sound/soc/ti/j721e-evm.c18
16 files changed, 152 insertions, 88 deletions
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index ee936d1aa724..c2930d65728e 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -114,7 +114,7 @@ properties:
ports:
$ref: /schemas/graph.yaml#/properties/ports
- properties:
+ patternProperties:
port(@[0-9a-f]+)?:
$ref: audio-graph-port.yaml#
unevaluatedProperties: false
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 675849d07284..8e6dd8a257c5 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -712,6 +712,12 @@ struct snd_soc_dai_link {
/* Do not create a PCM for this DAI link (Backend link) */
unsigned int ignore:1;
+ /* This flag will reorder stop sequence. By enabling this flag
+ * DMA controller stop sequence will be invoked first followed by
+ * CPU DAI driver stop sequence
+ */
+ unsigned int stop_dma_first:1;
+
#ifdef CONFIG_SND_SOC_TOPOLOGY
struct snd_soc_dobj dobj; /* For topology */
#endif
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 84e3906abd4f..9449fb40a956 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
| SND_SOC_DAIFMT_CBM_CFM,
.init = cz_rt5682_init,
.dpcm_playback = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_play_ops,
SND_SOC_DAILINK_REG(designware1, rt5682, platform),
},
@@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_cap_ops,
SND_SOC_DAILINK_REG(designware2, rt5682, platform),
},
@@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_playback = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_max_play_ops,
SND_SOC_DAILINK_REG(designware3, mx, platform),
},
@@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_dmic0_cap_ops,
SND_SOC_DAILINK_REG(designware3, adau, platform),
},
@@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_dmic1_cap_ops,
SND_SOC_DAILINK_REG(designware2, adau, platform),
},
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7ebae3f09435..a3b784ed4f70 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305
high-efficiency mono Class-D audio power amplifiers.
config SND_SOC_SSM2518
- tristate
+ tristate "Analog Devices SSM2518 Class-D Amplifier"
depends on I2C
config SND_SOC_SSM2602
@@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X
Qualcomm SoCs like SDM845.
config SND_SOC_WCD938X
+ depends on SND_SOC_WCD938X_SDW
tristate
config SND_SOC_WCD938X_SDW
@@ -1813,11 +1814,6 @@ config SND_SOC_ZL38060
which consists of a Digital Signal Processor (DSP), several Digital
Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
-config SND_SOC_ZX_AUD96P22
- tristate "ZTE ZX AUD96P22 CODEC"
- depends on I2C
- select REGMAP_I2C
-
# Amp
config SND_SOC_LM4857
tristate
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 3000bc128b5b..38356ea2bd6e 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = {
.reg_defaults = rt5631_reg,
.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
.cache_type = REGCACHE_RBTREE,
+ .use_single_read = true,
+ .use_single_write = true,
};
static int rt5631_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index e4c91571abae..abcd6f483788 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
- if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
- if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 51870d50f419..b504d63385b3 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
ret);
return ret;
}
+ regcache_cache_only(aic31xx->regmap, true);
+
aic31xx->dev = &i2c->dev;
aic31xx->irq = i2c->irq;
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 81952984613d..2513922a0292 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -151,8 +151,8 @@ struct aic31xx_pdata {
#define AIC31XX_WORD_LEN_24BITS 0x02
#define AIC31XX_WORD_LEN_32BITS 0x03
#define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2)
-#define AIC31XX_BCLK_MASTER BIT(2)
-#define AIC31XX_WCLK_MASTER BIT(3)
+#define AIC31XX_BCLK_MASTER BIT(3)
+#define AIC31XX_WCLK_MASTER BIT(2)
/* AIC31XX_DATA_OFFSET */
#define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index c63b717040ed..dcd8aeb45cb3 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
/* -12dB min, 0.5dB steps */
static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
-
-static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
static const char * const lo_cm_text[] = {
@@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
};
static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
- SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
- AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+ SOC_SINGLE_S8_TLV("PCM Playback Volume",
+ AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm),
SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
- SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
- AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
- tlv_driver_gain),
- SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
- AIC32X4_HPLGAIN, 6, 0x01, 1),
- SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+ SOC_SINGLE_TLV("HP Driver Gain Volume",
+ AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
+ SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
- SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
- 0, 0, 117, 1, tlv_spk_vol),
- SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
- 4, 5, 0, tlv_amp_vol),
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+ TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain),
+ SOC_SINGLE_TLV("Speaker Amplifier Playback Volume",
+ TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol),
+
+ SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
};
static const struct snd_kcontrol_new hp_output_mixer_controls[] = {
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index 78b76eceff8f..2fcc97370be2 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component)
(WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0);
}
- ret = wcd938x_irq_init(wcd938x, component->dev);
- if (ret) {
- dev_err(component->dev, "%s: IRQ init failed: %d\n",
- __func__, ret);
- return ret;
- }
-
wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
WCD938X_IRQ_HPHR_PDM_WD_INT);
wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
@@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
- wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
if (!wcd938x->txdev) {
@@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
- wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
if (!wcd938x->tx_sdw_dev) {
dev_err(dev, "could not get txslave with matching of dev\n");
@@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev)
return PTR_ERR(wcd938x->regmap);
}
+ ret = wcd938x_irq_init(wcd938x, dev);
+ if (ret) {
+ dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
+ return ret;
+ }
+
+ wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
+ wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
+
ret = wcd938x_set_micbias_data(wcd938x);
if (ret < 0) {
dev_err(dev, "%s: bad micbias pdata\n", __func__);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 37aa020f23f6..549d98241dae 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -282,6 +282,7 @@
/*
* HALO_CCM_CORE_CONTROL
*/
+#define HALO_CORE_RESET 0x00000200
#define HALO_CORE_EN 0x00000001
/*
@@ -1213,7 +1214,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
mutex_lock(&ctl->dsp->pwr_lock);
- ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size);
+ ret = wm_coeff_read_ctrl(ctl, ctl->cache, size);
if (!ret && copy_to_user(bytes, ctl->cache, size))
ret = -EFAULT;
@@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp)
{
return regmap_update_bits(dsp->regmap,
dsp->base + HALO_CCM_CORE_CONTROL,
- HALO_CORE_EN, HALO_CORE_EN);
+ HALO_CORE_RESET | HALO_CORE_EN,
+ HALO_CORE_RESET | HALO_CORE_EN);
}
static void wm_halo_stop_core(struct wm_adsp *dsp)
diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c
index 0e7ed906b341..25daef910aee 100644
--- a/sound/soc/intel/boards/sof_sdw_max98373.c
+++ b/sound/soc/intel/boards/sof_sdw_max98373.c
@@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd)
+static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable)
{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai;
int ret;
+ int j;
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- /* enable max98373 first */
- ret = max_98373_trigger(substream, cmd);
- if (ret < 0)
- break;
-
- ret = sdw_trigger(substream, cmd);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = sdw_trigger(substream, cmd);
- if (ret < 0)
- break;
-
- ret = max_98373_trigger(substream, cmd);
- break;
- default:
- ret = -EINVAL;
- break;
+ /* set spk pin by playback only */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return 0;
+
+ cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(cpu_dai->component);
+ char pin_name[16];
+
+ snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk",
+ codec_dai->component->name_prefix);
+
+ if (enable)
+ ret = snd_soc_dapm_enable_pin(dapm, pin_name);
+ else
+ ret = snd_soc_dapm_disable_pin(dapm, pin_name);
+
+ if (!ret)
+ snd_soc_dapm_sync(dapm);
}
- return ret;
+ return 0;
+}
+
+static int mx8373_sdw_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ /* according to soc_pcm_prepare dai link prepare is called first */
+ ret = sdw_prepare(substream);
+ if (ret < 0)
+ return ret;
+
+ return mx8373_enable_spk_pin(substream, true);
+}
+
+static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ /* according to soc_pcm_hw_free dai link free is called first */
+ ret = sdw_hw_free(substream);
+ if (ret < 0)
+ return ret;
+
+ return mx8373_enable_spk_pin(substream, false);
}
static const struct snd_soc_ops max_98373_sdw_ops = {
.startup = sdw_startup,
- .prepare = sdw_prepare,
- .trigger = max98373_sdw_trigger,
- .hw_free = sdw_hw_free,
+ .prepare = mx8373_sdw_prepare,
+ .trigger = sdw_trigger,
+ .hw_free = mx8373_sdw_hw_free,
.shutdown = sdw_shutdown,
};
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 46513bb97904..d1c570ca21ea 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1015,6 +1015,7 @@ out:
static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
int ret = -EINVAL, _ret = 0;
int rollback = 0;
@@ -1055,14 +1056,23 @@ start_err:
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
- if (ret < 0)
- break;
+ if (rtd->dai_link->stop_dma_first) {
+ ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
- ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
- if (ret < 0)
- break;
+ ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
+ } else {
+ ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
+ ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
+ }
ret = snd_soc_link_trigger(substream, cmd, rollback);
break;
}
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index a00262184efa..d04ce84fe7cc 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = {
static const struct sof_dev_desc adl_desc = {
.machines = snd_soc_acpi_intel_adl_machines,
.alt_machines = snd_soc_acpi_intel_adl_sdw_machines,
+ .use_acpi_target_states = true,
.resindex_lpe_base = 0,
.resindex_pcicfg_base = -1,
.resindex_imr_base = -1,
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 573374b89b10..d3276b4595af 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component,
}
EXPORT_SYMBOL_GPL(tegra_pcm_pointer);
-static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream,
size_t size)
{
struct snd_pcm_substream *substream = pcm->streams[stream].substream;
struct snd_dma_buffer *buf = &substream->dma_buffer;
- buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL);
+ buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL);
if (!buf->area)
return -ENOMEM;
buf->private_data = NULL;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
+ buf->dev.dev = dev;
buf->bytes = size;
return 0;
@@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
if (!buf->area)
return;
- dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr);
+ dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr);
buf->area = NULL;
}
-static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd,
+static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd,
size_t size)
{
- struct snd_card *card = rtd->card->snd_card;
struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32));
if (ret < 0)
return ret;
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = tegra_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK, size);
+ ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size);
if (ret)
goto err;
}
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = tegra_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE, size);
+ ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size);
if (ret)
goto err_free_play;
}
@@ -284,7 +281,16 @@ err:
int tegra_pcm_construct(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
- return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max);
+ struct device *dev = component->dev;
+
+ /*
+ * Fallback for backwards-compatibility with older device trees that
+ * have the iommus property in the virtual, top-level "sound" node.
+ */
+ if (!of_get_property(dev->of_node, "iommus", NULL))
+ dev = rtd->card->snd_card->dev;
+
+ return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max);
}
EXPORT_SYMBOL_GPL(tegra_pcm_construct);
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
index a7c0484d44ec..265bbc5a2f96 100644
--- a/sound/soc/ti/j721e-evm.c
+++ b/sound/soc/ti/j721e-evm.c
@@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv,
return ret;
}
- if (priv->hsdiv_rates[domain->parent_clk_id] != scki) {
+ if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) {
dev_dbg(priv->dev,
"%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI",
@@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream)
j721e_rule_rate, &priv->rate_range,
SNDRV_PCM_HW_PARAM_RATE, -1);
- mutex_unlock(&priv->mutex);
if (ret)
- return ret;
+ goto out;
/* Reset TDM slots to 32 */
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
if (ret && ret != -ENOTSUPP)
- return ret;
+ goto out;
for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
if (ret && ret != -ENOTSUPP)
- return ret;
+ goto out;
}
- return 0;
+ if (ret == -ENOTSUPP)
+ ret = 0;
+out:
+ if (ret)
+ domain->active--;
+ mutex_unlock(&priv->mutex);
+
+ return ret;
}
static int j721e_audio_hw_params(struct snd_pcm_substream *substream,
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-23 11:04:24 +0000