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-rw-r--r--sound/soc/amd/ps/pci-ps.c3
-rw-r--r--sound/soc/amd/ps/ps-pdm-dma.c10
-rw-r--r--sound/soc/amd/yc/acp6x-mach.c14
-rw-r--r--sound/soc/codecs/cs35l41-lib.c6
-rw-r--r--sound/soc/codecs/cs35l56.c6
-rw-r--r--sound/soc/codecs/lpass-tx-macro.c5
-rw-r--r--sound/soc/codecs/max98363.c4
-rw-r--r--sound/soc/codecs/nau8824.c24
-rw-r--r--sound/soc/codecs/rt5682-i2c.c4
-rw-r--r--sound/soc/codecs/rt5682.c6
-rw-r--r--sound/soc/codecs/rt5682.h1
-rw-r--r--sound/soc/codecs/ssm2602.c15
-rw-r--r--sound/soc/codecs/wcd938x-sdw.c1
-rw-r--r--sound/soc/codecs/wsa881x.c1
-rw-r--r--sound/soc/codecs/wsa883x.c1
-rw-r--r--sound/soc/dwc/dwc-i2s.c45
-rw-r--r--sound/soc/fsl/fsl_micfil.c14
-rw-r--r--sound/soc/fsl/fsl_sai.c11
-rw-r--r--sound/soc/fsl/fsl_sai.h1
-rw-r--r--sound/soc/generic/simple-card-utils.c2
-rw-r--r--sound/soc/generic/simple-card.c1
-rw-r--r--sound/soc/intel/avs/apl.c6
-rw-r--r--sound/soc/intel/avs/avs.h4
-rw-r--r--sound/soc/intel/avs/board_selection.c2
-rw-r--r--sound/soc/intel/avs/control.c22
-rw-r--r--sound/soc/intel/avs/dsp.c4
-rw-r--r--sound/soc/intel/avs/messages.h2
-rw-r--r--sound/soc/intel/avs/path.h2
-rw-r--r--sound/soc/intel/avs/pcm.c23
-rw-r--r--sound/soc/intel/avs/probes.c2
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c54
-rw-r--r--sound/soc/mediatek/mt8186/mt8186-afe-clk.c6
-rw-r--r--sound/soc/mediatek/mt8186/mt8186-afe-clk.h1
-rw-r--r--sound/soc/mediatek/mt8186/mt8186-afe-pcm.c4
-rw-r--r--sound/soc/mediatek/mt8186/mt8186-audsys-clk.c46
-rw-r--r--sound/soc/mediatek/mt8186/mt8186-audsys-clk.h1
-rw-r--r--sound/soc/mediatek/mt8188/mt8188-afe-clk.c7
-rw-r--r--sound/soc/mediatek/mt8188/mt8188-afe-clk.h1
-rw-r--r--sound/soc/mediatek/mt8188/mt8188-afe-pcm.c4
-rw-r--r--sound/soc/mediatek/mt8188/mt8188-audsys-clk.c47
-rw-r--r--sound/soc/mediatek/mt8188/mt8188-audsys-clk.h1
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-afe-clk.c5
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-afe-clk.h1
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-afe-pcm.c4
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-audsys-clk.c47
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-audsys-clk.h1
-rw-r--r--sound/soc/soc-pcm.c20
-rw-r--r--sound/soc/sof/amd/acp-ipc.c7
-rw-r--r--sound/soc/sof/debug.c4
-rw-r--r--sound/soc/sof/intel/hda-mlink.c96
-rw-r--r--sound/soc/sof/ipc3-topology.c7
-rw-r--r--sound/soc/sof/ipc4-topology.c4
-rw-r--r--sound/soc/sof/pcm.c11
-rw-r--r--sound/soc/sof/pm.c14
-rw-r--r--sound/soc/sof/sof-client-probes.c14
-rw-r--r--sound/soc/sof/topology.c6
-rw-r--r--sound/soc/tegra/tegra_pcm.c3
57 files changed, 442 insertions, 216 deletions
diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c
index afddb9a77ba4..b1337b96ea8d 100644
--- a/sound/soc/amd/ps/pci-ps.c
+++ b/sound/soc/amd/ps/pci-ps.c
@@ -211,8 +211,7 @@ static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data
case ACP63_PDM_DEV_MASK:
adata->pdm_dev_index = 0;
acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma",
- 0, adata->res, 1, &adata->acp_lock,
- sizeof(adata->acp_lock));
+ 0, adata->res, 1, NULL, 0);
acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec",
0, NULL, 0, NULL, 0);
acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach",
diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c
index 46b91327168f..3a83dc178e7d 100644
--- a/sound/soc/amd/ps/ps-pdm-dma.c
+++ b/sound/soc/amd/ps/ps-pdm-dma.c
@@ -361,12 +361,12 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev)
{
struct resource *res;
struct pdm_dev_data *adata;
+ struct acp63_dev_data *acp_data;
+ struct device *parent;
int status;
- if (!pdev->dev.platform_data) {
- dev_err(&pdev->dev, "platform_data not retrieved\n");
- return -ENODEV;
- }
+ parent = pdev->dev.parent;
+ acp_data = dev_get_drvdata(parent);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n");
@@ -382,7 +382,7 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev)
return -ENOMEM;
adata->capture_stream = NULL;
- adata->acp_lock = pdev->dev.platform_data;
+ adata->acp_lock = &acp_data->acp_lock;
dev_set_drvdata(&pdev->dev, adata);
status = devm_snd_soc_register_component(&pdev->dev,
&acp63_pdm_component,
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 0bc6e4066d0f..246299a178f9 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -175,6 +175,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
.driver_data = &acp6x_card,
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "21EF"),
+ }
+ },
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_NAME, "21EM"),
}
},
@@ -311,6 +318,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_BOARD_NAME, "8A22"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "System76"),
+ DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"),
+ }
+ },
{}
};
diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c
index 8538e2871c5f..1e4205295a0d 100644
--- a/sound/soc/codecs/cs35l41-lib.c
+++ b/sound/soc/codecs/cs35l41-lib.c
@@ -46,7 +46,7 @@ static const struct reg_default cs35l41_reg[] = {
{ CS35L41_DSP1_RX5_SRC, 0x00000020 },
{ CS35L41_DSP1_RX6_SRC, 0x00000021 },
{ CS35L41_DSP1_RX7_SRC, 0x0000003A },
- { CS35L41_DSP1_RX8_SRC, 0x00000001 },
+ { CS35L41_DSP1_RX8_SRC, 0x0000003B },
{ CS35L41_NGATE1_SRC, 0x00000008 },
{ CS35L41_NGATE2_SRC, 0x00000009 },
{ CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 },
@@ -58,8 +58,8 @@ static const struct reg_default cs35l41_reg[] = {
{ CS35L41_IRQ1_MASK2, 0xFFFFFFFF },
{ CS35L41_IRQ1_MASK3, 0xFFFF87FF },
{ CS35L41_IRQ1_MASK4, 0xFEFFFFFF },
- { CS35L41_GPIO1_CTRL1, 0xE1000001 },
- { CS35L41_GPIO2_CTRL1, 0xE1000001 },
+ { CS35L41_GPIO1_CTRL1, 0x81000001 },
+ { CS35L41_GPIO2_CTRL1, 0x81000001 },
{ CS35L41_MIXER_NGATE_CFG, 0x00000000 },
{ CS35L41_MIXER_NGATE_CH1_CFG, 0x00000303 },
{ CS35L41_MIXER_NGATE_CH2_CFG, 0x00000303 },
diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c
index 46762f7f1449..e0d2b9bb2326 100644
--- a/sound/soc/codecs/cs35l56.c
+++ b/sound/soc/codecs/cs35l56.c
@@ -704,9 +704,6 @@ static int cs35l56_sdw_dai_hw_free(struct snd_pcm_substream *substream,
static int cs35l56_sdw_dai_set_stream(struct snd_soc_dai *dai,
void *sdw_stream, int direction)
{
- if (!sdw_stream)
- return 0;
-
snd_soc_dai_dma_data_set(dai, direction, sdw_stream);
return 0;
@@ -852,10 +849,11 @@ static void cs35l56_dsp_work(struct work_struct *work)
*/
if (cs35l56->sdw_peripheral) {
cs35l56->sdw_irq_no_unmask = true;
- cancel_work_sync(&cs35l56->sdw_irq_work);
+ flush_work(&cs35l56->sdw_irq_work);
sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0);
sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1);
sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF);
+ flush_work(&cs35l56->sdw_irq_work);
}
ret = cs35l56_mbox_send(cs35l56, CS35L56_MBOX_CMD_SHUTDOWN);
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index da6fcf7f0991..de978c3d70b7 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol,
struct tx_macro *tx = snd_soc_component_get_drvdata(component);
val = ucontrol->value.enumerated.item[0];
+ if (val >= e->items)
+ return -EINVAL;
switch (e->reg) {
case CDC_TX_INP_MUX_ADC_MUX0_CFG0:
@@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol,
case CDC_TX_INP_MUX_ADC_MUX7_CFG0:
mic_sel_reg = CDC_TX7_TX_PATH_CFG0;
break;
+ default:
+ dev_err(component->dev, "Error in configuration!!\n");
+ return -EINVAL;
}
if (val != 0) {
diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c
index dcce06bff756..e6b84e222b50 100644
--- a/sound/soc/codecs/max98363.c
+++ b/sound/soc/codecs/max98363.c
@@ -211,7 +211,7 @@ static int max98363_io_init(struct sdw_slave *slave)
}
#define MAX98363_RATES SNDRV_PCM_RATE_8000_192000
-#define MAX98363_FORMATS (SNDRV_PCM_FMTBIT_S32_LE)
+#define MAX98363_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
@@ -246,7 +246,7 @@ static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream,
stream_config.frame_rate = params_rate(params);
stream_config.bps = snd_pcm_format_width(params_format(params));
stream_config.direction = direction;
- stream_config.ch_count = params_channels(params);
+ stream_config.ch_count = 1;
if (stream_config.ch_count > runtime->hw.channels_max) {
stream_config.ch_count = runtime->hw.channels_max;
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 4f19fd9b65d1..5a4db8944d06 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -1903,6 +1903,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = {
},
.driver_data = (void *)(NAU8824_MONO_SPEAKER),
},
+ {
+ /* Positivo CW14Q01P */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"),
+ DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"),
+ },
+ .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+ },
+ {
+ /* Positivo K1424G */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"),
+ DMI_MATCH(DMI_BOARD_NAME, "K1424G"),
+ },
+ .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+ },
+ {
+ /* Positivo N14ZP74G */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"),
+ DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"),
+ },
+ .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+ },
{}
};
diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c
index 2935c1bb81f3..5bc46b041786 100644
--- a/sound/soc/codecs/rt5682-i2c.c
+++ b/sound/soc/codecs/rt5682-i2c.c
@@ -267,7 +267,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c)
ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
| IRQF_ONESHOT, "rt5682", rt5682);
- if (ret)
+ if (!ret)
+ rt5682->irq = i2c->irq;
+ else
dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
}
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index f6c798b65c08..5d992543b791 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component)
if (rt5682->is_sdw)
return 0;
+ if (rt5682->irq)
+ disable_irq(rt5682->irq);
+
cancel_delayed_work_sync(&rt5682->jack_detect_work);
cancel_delayed_work_sync(&rt5682->jd_check_work);
if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) {
@@ -3027,6 +3030,9 @@ static int rt5682_resume(struct snd_soc_component *component)
mod_delayed_work(system_power_efficient_wq,
&rt5682->jack_detect_work, msecs_to_jiffies(0));
+ if (rt5682->irq)
+ enable_irq(rt5682->irq);
+
return 0;
}
#else
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index d568c6993c33..e8efd8a84a6c 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1462,6 +1462,7 @@ struct rt5682_priv {
int pll_out[RT5682_PLLS];
int jack_type;
+ int irq;
int irq_work_delay_time;
};
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 00b60369b029..c29324403e9d 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = {
{ .reg = 0x09, .def = 0x0000 }
};
+/*
+ * ssm2602 register patch
+ * Workaround for playback distortions after power up: activates digital
+ * core, and then powers on output, DAC, and whole chip at the same time
+ */
+
+static const struct reg_sequence ssm2602_patch[] = {
+ { SSM2602_ACTIVE, 0x01 },
+ { SSM2602_PWR, 0x07 },
+ { SSM2602_RESET, 0x00 },
+};
+
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
@@ -598,6 +610,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component)
return ret;
}
+ regmap_register_patch(ssm2602->regmap, ssm2602_patch,
+ ARRAY_SIZE(ssm2602_patch));
+
/* set the update bits */
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c
index 402286dfaea4..9c10200ff34b 100644
--- a/sound/soc/codecs/wcd938x-sdw.c
+++ b/sound/soc/codecs/wcd938x-sdw.c
@@ -1190,7 +1190,6 @@ static const struct regmap_config wcd938x_regmap_config = {
.readable_reg = wcd938x_readable_register,
.writeable_reg = wcd938x_writeable_register,
.volatile_reg = wcd938x_volatile_register,
- .can_multi_write = true,
};
static const struct sdw_slave_ops wcd9380_slave_ops = {
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index f709231b1277..97f6873a0a8c 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -645,7 +645,6 @@ static struct regmap_config wsa881x_regmap_config = {
.readable_reg = wsa881x_readable_register,
.reg_format_endian = REGMAP_ENDIAN_NATIVE,
.val_format_endian = REGMAP_ENDIAN_NATIVE,
- .can_multi_write = true,
};
enum {
diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c
index c609cb63dae6..e80b53143569 100644
--- a/sound/soc/codecs/wsa883x.c
+++ b/sound/soc/codecs/wsa883x.c
@@ -946,7 +946,6 @@ static struct regmap_config wsa883x_regmap_config = {
.writeable_reg = wsa883x_writeable_register,
.reg_format_endian = REGMAP_ENDIAN_NATIVE,
.val_format_endian = REGMAP_ENDIAN_NATIVE,
- .can_multi_write = true,
.use_single_read = true,
};
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index acdf98b2ee9c..399a489f24f2 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
/* Error Handling: TX */
if (isr[i] & ISR_TXFO) {
- dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i);
+ dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i);
irq_valid = true;
}
/* Error Handling: TX */
if (isr[i] & ISR_RXFO) {
- dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i);
+ dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i);
irq_valid = true;
}
}
@@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev,
}
}
-static int dw_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
- union dw_i2s_snd_dma_data *dma_data = NULL;
-
- if (!(dev->capability & DWC_I2S_RECORD) &&
- (substream->stream == SNDRV_PCM_STREAM_CAPTURE))
- return -EINVAL;
-
- if (!(dev->capability & DWC_I2S_PLAY) &&
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
- return -EINVAL;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dma_data = &dev->play_dma_data;
- else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
- dma_data = &dev->capture_dma_data;
-
- snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data);
-
- return 0;
-}
-
static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
{
u32 ch_reg;
@@ -305,12 +281,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- snd_soc_dai_set_dma_data(dai, substream, NULL);
-}
-
static int dw_i2s_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -382,8 +352,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
}
static const struct snd_soc_dai_ops dw_i2s_dai_ops = {
- .startup = dw_i2s_startup,
- .shutdown = dw_i2s_shutdown,
.hw_params = dw_i2s_hw_params,
.prepare = dw_i2s_prepare,
.trigger = dw_i2s_trigger,
@@ -625,6 +593,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev,
}
+static int dw_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data);
+ return 0;
+}
+
static int dw_i2s_probe(struct platform_device *pdev)
{
const struct i2s_platform_data *pdata = pdev->dev.platform_data;
@@ -643,6 +619,7 @@ static int dw_i2s_probe(struct platform_device *pdev)
return -ENOMEM;
dw_i2s_dai->ops = &dw_i2s_dai_ops;
+ dw_i2s_dai->probe = dw_i2s_dai_probe;
dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res);
if (IS_ERR(dev->i2s_base))
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 94341e4352b3..3f08082a55be 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -1159,7 +1159,7 @@ static int fsl_micfil_probe(struct platform_device *pdev)
ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
if (ret) {
dev_err(&pdev->dev, "failed to pcm register\n");
- return ret;
+ goto err_pm_disable;
}
fsl_micfil_dai.capture.formats = micfil->soc->formats;
@@ -1169,9 +1169,20 @@ static int fsl_micfil_probe(struct platform_device *pdev)
if (ret) {
dev_err(&pdev->dev, "failed to register component %s\n",
fsl_micfil_component.name);
+ goto err_pm_disable;
}
return ret;
+
+err_pm_disable:
+ pm_runtime_disable(&pdev->dev);
+
+ return ret;
+}
+
+static void fsl_micfil_remove(struct platform_device *pdev)
+{
+ pm_runtime_disable(&pdev->dev);
}
static int __maybe_unused fsl_micfil_runtime_suspend(struct device *dev)
@@ -1232,6 +1243,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = {
static struct platform_driver fsl_micfil_driver = {
.probe = fsl_micfil_probe,
+ .remove_new = fsl_micfil_remove,
.driver = {
.name = "fsl-micfil-dai",
.pm = &fsl_micfil_pm_ops,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index abdaffb00fbd..e3105d48fb65 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -491,14 +491,21 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_MSEL_MASK,
FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
- if (savediv == 1)
+ if (savediv == 1) {
regmap_update_bits(sai->regmap, reg,
FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP,
FSL_SAI_CR2_BYP);
- else
+ if (fsl_sai_dir_is_synced(sai, adir))
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs),
+ FSL_SAI_CR2_BCI, FSL_SAI_CR2_BCI);
+ else
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs),
+ FSL_SAI_CR2_BCI, 0);
+ } else {
regmap_update_bits(sai->regmap, reg,
FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP,
savediv / 2 - 1);
+ }
if (sai->soc_data->max_register >= FSL_SAI_MCTL) {
/* SAI is in master mode at this point, so enable MCLK */
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 197748a888d5..a53c4f0e25fa 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -116,6 +116,7 @@
/* SAI Transmit and Receive Configuration 2 Register */
#define FSL_SAI_CR2_SYNC BIT(30)
+#define FSL_SAI_CR2_BCI BIT(28)
#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26)
#define FSL_SAI_CR2_MSEL_BUS 0
#define FSL_SAI_CR2_MSEL_MCLK1 BIT(26)
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 467edd96eae5..e5ff61c1e9d1 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -314,7 +314,7 @@ int asoc_simple_startup(struct snd_pcm_substream *substream)
}
ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE,
fixed_rate, fixed_rate);
- if (ret)
+ if (ret < 0)
goto codec_err;
}
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6f044cc8357e..5a5e4ecd0f61 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -416,6 +416,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv,
if (ret < 0) {
of_node_put(codec);
+ of_node_put(plat);
of_node_put(np);
goto error;
}
diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c
index 02683dce277a..1860099c782a 100644
--- a/sound/soc/intel/avs/apl.c
+++ b/sound/soc/intel/avs/apl.c
@@ -169,6 +169,7 @@ static bool apl_lp_streaming(struct avs_dev *adev)
{
struct avs_path *path;
+ spin_lock(&adev->path_list_lock);
/* Any gateway without buffer allocated in LP area disqualifies D0IX. */
list_for_each_entry(path, &adev->path_list, node) {
struct avs_path_pipeline *ppl;
@@ -188,11 +189,14 @@ static bool apl_lp_streaming(struct avs_dev *adev)
if (cfg->copier.dma_type == INVALID_OBJECT_ID)
continue;
- if (!mod->gtw_attrs.lp_buffer_alloc)
+ if (!mod->gtw_attrs.lp_buffer_alloc) {
+ spin_unlock(&adev->path_list_lock);
return false;
+ }
}
}
}
+ spin_unlock(&adev->path_list_lock);
return true;
}
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index d7fccdcb9c16..0cf38c9e768e 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -283,8 +283,8 @@ void avs_release_firmwares(struct avs_dev *adev);
int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id,
u8 core_id, u8 domain, void *param, u32 param_size,
- u16 *instance_id);
-void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id,
+ u8 *instance_id);
+void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id,
u8 ppl_instance_id, u8 core_id);
int avs_dsp_create_pipeline(struct avs_dev *adev, u16 req_size, u8 priority,
bool lp, u16 attributes, u8 *instance_id);
diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c
index b2823c2107f7..60f8fb0bff95 100644
--- a/sound/soc/intel/avs/board_selection.c
+++ b/sound/soc/intel/avs/board_selection.c
@@ -443,7 +443,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev)
}
for (mach = boards->machs; mach->id[0]; mach++) {
- if (!acpi_dev_present(mach->id, NULL, -1))
+ if (!acpi_dev_present(mach->id, mach->uid, -1))
continue;
if (mach->machine_quirk)
diff --git a/sound/soc/intel/avs/control.c b/sound/soc/intel/avs/control.c
index a8b14b784f8a..3dfa2e9816db 100644
--- a/sound/soc/intel/avs/control.c
+++ b/sound/soc/intel/avs/control.c
@@ -21,17 +21,25 @@ static struct avs_dev *avs_get_kcontrol_adev(struct snd_kcontrol *kcontrol)
return to_avs_dev(w->dapm->component->dev);
}
-static struct avs_path_module *avs_get_kcontrol_module(struct avs_dev *adev, u32 id)
+static struct avs_path_module *avs_get_volume_module(struct avs_dev *adev, u32 id)
{
struct avs_path *path;
struct avs_path_pipeline *ppl;
struct avs_path_module *mod;
- list_for_each_entry(path, &adev->path_list, node)
- list_for_each_entry(ppl, &path->ppl_list, node)
- list_for_each_entry(mod, &ppl->mod_list, node)
- if (mod->template->ctl_id && mod->template->ctl_id == id)
+ spin_lock(&adev->path_list_lock);
+ list_for_each_entry(path, &adev->path_list, node) {
+ list_for_each_entry(ppl, &path->ppl_list, node) {
+ list_for_each_entry(mod, &ppl->mod_list, node) {
+ if (guid_equal(&mod->template->cfg_ext->type, &AVS_PEAKVOL_MOD_UUID)
+ && mod->template->ctl_id == id) {
+ spin_unlock(&adev->path_list_lock);
return mod;
+ }
+ }
+ }
+ }
+ spin_unlock(&adev->path_list_lock);
return NULL;
}
@@ -49,7 +57,7 @@ int avs_control_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
/* prevent access to modules while path is being constructed */
mutex_lock(&adev->path_mutex);
- active_module = avs_get_kcontrol_module(adev, ctl_data->id);
+ active_module = avs_get_volume_module(adev, ctl_data->id);
if (active_module) {
ret = avs_ipc_peakvol_get_volume(adev, active_module->module_id,
active_module->instance_id, &dspvols,
@@ -89,7 +97,7 @@ int avs_control_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
changed = 1;
}
- active_module = avs_get_kcontrol_module(adev, ctl_data->id);
+ active_module = avs_get_volume_module(adev, ctl_data->id);
if (active_module) {
dspvol.channel_id = AVS_ALL_CHANNELS_MASK;
dspvol.target_volume = *volume;
diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c
index b881100d3e02..aa03af4473e9 100644
--- a/sound/soc/intel/avs/dsp.c
+++ b/sound/soc/intel/avs/dsp.c
@@ -225,7 +225,7 @@ err:
int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id,
u8 core_id, u8 domain, void *param, u32 param_size,
- u16 *instance_id)
+ u8 *instance_id)
{
struct avs_module_entry mentry;
bool was_loaded = false;
@@ -272,7 +272,7 @@ err_mod_entry:
return ret;
}
-void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id,
+void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id,
u8 ppl_instance_id, u8 core_id)
{
struct avs_module_entry mentry;
diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h
index d3b60ae7d743..7f23a304b4a9 100644
--- a/sound/soc/intel/avs/messages.h
+++ b/sound/soc/intel/avs/messages.h
@@ -619,7 +619,7 @@ enum avs_channel_config {
AVS_CHANNEL_CONFIG_DUAL_MONO = 9,
AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_0 = 10,
AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_1 = 11,
- AVS_CHANNEL_CONFIG_4_CHANNEL = 12,
+ AVS_CHANNEL_CONFIG_7_1 = 12,
AVS_CHANNEL_CONFIG_INVALID
};
diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h
index 197222c5e008..657f7b093e80 100644
--- a/sound/soc/intel/avs/path.h
+++ b/sound/soc/intel/avs/path.h
@@ -37,7 +37,7 @@ struct avs_path_pipeline {
struct avs_path_module {
u16 module_id;
- u16 instance_id;
+ u8 instance_id;
union avs_gtw_attributes gtw_attrs;
struct avs_tplg_module *template;
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 31c032a0f7e4..1fbb2c2fadb5 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -468,21 +468,34 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST);
if (!host_stream) {
- kfree(data);
- return -EBUSY;
+ ret = -EBUSY;
+ goto err;
}
data->host_stream = host_stream;
- snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto err;
+
/* avoid wrap-around with wall-clock */
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
- snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+ ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
+ if (ret < 0)
+ goto err;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+ if (ret < 0)
+ goto err;
+
snd_pcm_set_sync(substream);
dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p",
__func__, hdac_stream(host_stream)->stream_tag, substream);
return 0;
+
+err:
+ kfree(data);
+ return ret;
}
static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c
index 70a94201d6a5..275928281c6c 100644
--- a/sound/soc/intel/avs/probes.c
+++ b/sound/soc/intel/avs/probes.c
@@ -18,7 +18,7 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
{
struct avs_probe_cfg cfg = {{0}};
struct avs_module_entry mentry;
- u16 dummy;
+ u8 dummy;
avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 6d9cfe0a5041..d0f6c945d9ae 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
+static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate,
+ unsigned long i2sdiv_max)
+{
+ unsigned long div, rate1, rate2, err1, err2;
+
+ div = mclk / (64 * rate);
+ if (div == 0)
+ div = 1;
+
+ rate1 = mclk / (64 * div);
+ rate2 = mclk / (64 * (div + 1));
+
+ err1 = abs(rate1 - rate);
+ err2 = abs(rate2 - rate);
+
+ /*
+ * Choose the divider that produces the smallest error in the
+ * output rate and reject dividers with a 5% or higher error.
+ * In the event that both dividers are outside the acceptable
+ * error margin, reject the rate to prevent distorted audio.
+ * (The number 5% is arbitrary.)
+ */
+ if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20)
+ return div;
+ if (div < i2sdiv_max && err2 < rate/20)
+ return div + 1;
+
+ return -EINVAL;
+}
+
static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
struct regmap_field *div_field;
+ unsigned long i2sdiv_max;
unsigned int sample_size;
- uint32_t ctrl;
- int div;
+ uint32_t ctrl, conf;
+ int div = 1;
regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl);
-
- div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
+ regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
@@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
div_field = i2s->field_i2sdiv_playback;
+ i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb,
+ i2s->soc_info->field_i2sdiv_playback.lsb);
} else {
ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE;
ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size);
div_field = i2s->field_i2sdiv_capture;
+ i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb,
+ i2s->soc_info->field_i2sdiv_capture.lsb);
+ }
+
+ /*
+ * Only calculate I2SDIV if we're supplying the bit or frame clock.
+ * If the codec is supplying both clocks then the divider output is
+ * unused, and we don't want it to limit the allowed sample rates.
+ */
+ if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) {
+ div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s),
+ params_rate(params), i2sdiv_max);
+ if (div < 0)
+ return div;
}
regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl);
diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c
index a6b4f29049bb..539e3a023bc4 100644
--- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c
+++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c
@@ -644,9 +644,3 @@ int mt8186_init_clock(struct mtk_base_afe *afe)
return 0;
}
-
-void mt8186_deinit_clock(void *priv)
-{
- struct mtk_base_afe *afe = priv;
- mt8186_audsys_clk_unregister(afe);
-}
diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h
index d5988717d8f2..a9d59e506d9a 100644
--- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h
+++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h
@@ -81,7 +81,6 @@ enum {
struct mtk_base_afe;
int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, int clk_id);
int mt8186_init_clock(struct mtk_base_afe *afe);
-void mt8186_deinit_clock(void *priv);
int mt8186_afe_enable_cgs(struct mtk_base_afe *afe);
void mt8186_afe_disable_cgs(struct mtk_base_afe *afe);
int mt8186_afe_enable_clock(struct mtk_base_afe *afe);
diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
index 41172a82103e..a868a04ed4e7 100644
--- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
+++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
@@ -2848,10 +2848,6 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev)
return ret;
}
- ret = devm_add_action_or_reset(dev, mt8186_deinit_clock, (void *)afe);
- if (ret)
- return ret;
-
/* init memif */
afe->memif_32bit_supported = 0;
afe->memif_size = MT8186_MEMIF_NUM;
diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c
index 578969ca91c8..5666be6b1bd2 100644
--- a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c
+++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c
@@ -84,6 +84,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
GATE_AUD2(CLK_AUD_ETDM_OUT1_BCLK, "aud_etdm_out1_bclk", "top_audio", 24),
};
+static void mt8186_audsys_clk_unregister(void *data)
+{
+ struct mtk_base_afe *afe = data;
+ struct mt8186_afe_private *afe_priv = afe->platform_priv;
+ struct clk *clk;
+ struct clk_lookup *cl;
+ int i;
+
+ if (!afe_priv)
+ return;
+
+ for (i = 0; i < CLK_AUD_NR_CLK; i++) {
+ cl = afe_priv->lookup[i];
+ if (!cl)
+ continue;
+
+ clk = cl->clk;
+ clk_unregister_gate(clk);
+
+ clkdev_drop(cl);
+ }
+}
+
int mt8186_audsys_clk_register(struct mtk_base_afe *afe)
{
struct mt8186_afe_private *afe_priv = afe->platform_priv;
@@ -124,27 +147,6 @@ int mt8186_audsys_clk_register(struct mtk_base_afe *afe)
afe_priv->lookup[i] = cl;
}
- return 0;
+ return devm_add_action_or_reset(afe->dev, mt8186_audsys_clk_unregister, afe);
}
-void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe)
-{
- struct mt8186_afe_private *afe_priv = afe->platform_priv;
- struct clk *clk;
- struct clk_lookup *cl;
- int i;
-
- if (!afe_priv)
- return;
-
- for (i = 0; i < CLK_AUD_NR_CLK; i++) {
- cl = afe_priv->lookup[i];
- if (!cl)
- continue;
-
- clk = cl->clk;
- clk_unregister_gate(clk);
-
- clkdev_drop(cl);
- }
-}
diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h
index b8d6a06e11e8..897a2914dc19 100644
--- a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h
+++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h
@@ -10,6 +10,5 @@
#define _MT8186_AUDSYS_CLK_H_
int mt8186_audsys_clk_register(struct mtk_base_afe *afe);
-void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe);
#endif
diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-clk.c b/sound/soc/mediatek/mt8188/mt8188-afe-clk.c
index 743d6a162cb9..0fb97517f82c 100644
--- a/sound/soc/mediatek/mt8188/mt8188-afe-clk.c
+++ b/sound/soc/mediatek/mt8188/mt8188-afe-clk.c
@@ -418,13 +418,6 @@ int mt8188_afe_init_clock(struct mtk_base_afe *afe)
return 0;
}
-void mt8188_afe_deinit_clock(void *priv)
-{
- struct mtk_base_afe *afe = priv;
-
- mt8188_audsys_clk_unregister(afe);
-}
-
int mt8188_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk)
{
int ret;
diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-clk.h b/sound/soc/mediatek/mt8188/mt8188-afe-clk.h
index 084fdfb1d877..a4203a87a1e3 100644
--- a/sound/soc/mediatek/mt8188/mt8188-afe-clk.h
+++ b/sound/soc/mediatek/mt8188/mt8188-afe-clk.h
@@ -100,7 +100,6 @@ int mt8188_afe_get_mclk_source_clk_id(int sel);
int mt8188_afe_get_mclk_source_rate(struct mtk_base_afe *afe, int apll);
int mt8188_afe_get_default_mclk_source_by_rate(int rate);
int mt8188_afe_init_clock(struct mtk_base_afe *afe);
-void mt8188_afe_deinit_clock(void *priv);
int mt8188_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk);
void mt8188_afe_disable_clk(struct mtk_base_afe *afe, struct clk *clk);
int mt8188_afe_set_clk_rate(struct mtk_base_afe *afe, struct clk *clk,
diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
index e5f9373bed56..bcf7025886df 100644
--- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
+++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
@@ -3185,10 +3185,6 @@ static int mt8188_afe_pcm_dev_probe(struct platform_device *pdev)
if (ret)
return dev_err_probe(dev, ret, "init clock error");
- ret = devm_add_action_or_reset(dev, mt8188_afe_deinit_clock, (void *)afe);
- if (ret)
- return ret;
-
spin_lock_init(&afe_priv->afe_ctrl_lock);
mutex_init(&afe->irq_alloc_lock);
diff --git a/sound/soc/mediatek/mt8188/mt8188-audsys-clk.c b/sound/soc/mediatek/mt8188/mt8188-audsys-clk.c
index be1c53bf4729..c796ad8b62ee 100644
--- a/sound/soc/mediatek/mt8188/mt8188-audsys-clk.c
+++ b/sound/soc/mediatek/mt8188/mt8188-audsys-clk.c
@@ -138,6 +138,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
GATE_AUD6(CLK_AUD_GASRC11, "aud_gasrc11", "top_asm_h", 11),
};
+static void mt8188_audsys_clk_unregister(void *data)
+{
+ struct mtk_base_afe *afe = data;
+ struct mt8188_afe_private *afe_priv = afe->platform_priv;
+ struct clk *clk;
+ struct clk_lookup *cl;
+ int i;
+
+ if (!afe_priv)
+ return;
+
+ for (i = 0; i < CLK_AUD_NR_CLK; i++) {
+ cl = afe_priv->lookup[i];
+ if (!cl)
+ continue;
+
+ clk = cl->clk;
+ clk_unregister_gate(clk);
+
+ clkdev_drop(cl);
+ }
+}
+
int mt8188_audsys_clk_register(struct mtk_base_afe *afe)
{
struct mt8188_afe_private *afe_priv = afe->platform_priv;
@@ -179,27 +202,5 @@ int mt8188_audsys_clk_register(struct mtk_base_afe *afe)
afe_priv->lookup[i] = cl;
}
- return 0;
-}
-
-void mt8188_audsys_clk_unregister(struct mtk_base_afe *afe)
-{
- struct mt8188_afe_private *afe_priv = afe->platform_priv;
- struct clk *clk;
- struct clk_lookup *cl;
- int i;
-
- if (!afe_priv)
- return;
-
- for (i = 0; i < CLK_AUD_NR_CLK; i++) {
- cl = afe_priv->lookup[i];
- if (!cl)
- continue;
-
- clk = cl->clk;
- clk_unregister_gate(clk);
-
- clkdev_drop(cl);
- }
+ return devm_add_action_or_reset(afe->dev, mt8188_audsys_clk_unregister, afe);
}
diff --git a/sound/soc/mediatek/mt8188/mt8188-audsys-clk.h b/sound/soc/mediatek/mt8188/mt8188-audsys-clk.h
index 6c5f463ad7e4..45b0948c4a06 100644
--- a/sound/soc/mediatek/mt8188/mt8188-audsys-clk.h
+++ b/sound/soc/mediatek/mt8188/mt8188-audsys-clk.h
@@ -10,6 +10,5 @@
#define _MT8188_AUDSYS_CLK_H_
int mt8188_audsys_clk_register(struct mtk_base_afe *afe);
-void mt8188_audsys_clk_unregister(struct mtk_base_afe *afe);
#endif
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c
index 9ca2cb8c8a9c..f35318ae0739 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c
@@ -410,11 +410,6 @@ int mt8195_afe_init_clock(struct mtk_base_afe *afe)
return 0;
}
-void mt8195_afe_deinit_clock(struct mtk_base_afe *afe)
-{
- mt8195_audsys_clk_unregister(afe);
-}
-
int mt8195_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk)
{
int ret;
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.h b/sound/soc/mediatek/mt8195/mt8195-afe-clk.h
index 40663e31becd..a08c0ee6c860 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-clk.h
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.h
@@ -101,7 +101,6 @@ int mt8195_afe_get_mclk_source_clk_id(int sel);
int mt8195_afe_get_mclk_source_rate(struct mtk_base_afe *afe, int apll);
int mt8195_afe_get_default_mclk_source_by_rate(int rate);
int mt8195_afe_init_clock(struct mtk_base_afe *afe);
-void mt8195_afe_deinit_clock(struct mtk_base_afe *afe);
int mt8195_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk);
void mt8195_afe_disable_clk(struct mtk_base_afe *afe, struct clk *clk);
int mt8195_afe_prepare_clk(struct mtk_base_afe *afe, struct clk *clk);
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
index 9e45efeada55..03dabc056b91 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
@@ -3255,15 +3255,11 @@ err_pm_put:
static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev)
{
- struct mtk_base_afe *afe = platform_get_drvdata(pdev);
-
snd_soc_unregister_component(&pdev->dev);
pm_runtime_disable(&pdev->dev);
if (!pm_runtime_status_suspended(&pdev->dev))
mt8195_afe_runtime_suspend(&pdev->dev);
-
- mt8195_afe_deinit_clock(afe);
}
static const struct of_device_id mt8195_afe_pcm_dt_match[] = {
diff --git a/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c b/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c
index e0670e0dbd5b..38594bc3f2f7 100644
--- a/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c
+++ b/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c
@@ -148,6 +148,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
GATE_AUD6(CLK_AUD_GASRC19, "aud_gasrc19", "top_asm_h", 19),
};
+static void mt8195_audsys_clk_unregister(void *data)
+{
+ struct mtk_base_afe *afe = data;
+ struct mt8195_afe_private *afe_priv = afe->platform_priv;
+ struct clk *clk;
+ struct clk_lookup *cl;
+ int i;
+
+ if (!afe_priv)
+ return;
+
+ for (i = 0; i < CLK_AUD_NR_CLK; i++) {
+ cl = afe_priv->lookup[i];
+ if (!cl)
+ continue;
+
+ clk = cl->clk;
+ clk_unregister_gate(clk);
+
+ clkdev_drop(cl);
+ }
+}
+
int mt8195_audsys_clk_register(struct mtk_base_afe *afe)
{
struct mt8195_afe_private *afe_priv = afe->platform_priv;
@@ -188,27 +211,5 @@ int mt8195_audsys_clk_register(struct mtk_base_afe *afe)
afe_priv->lookup[i] = cl;
}
- return 0;
-}
-
-void mt8195_audsys_clk_unregister(struct mtk_base_afe *afe)
-{
- struct mt8195_afe_private *afe_priv = afe->platform_priv;
- struct clk *clk;
- struct clk_lookup *cl;
- int i;
-
- if (!afe_priv)
- return;
-
- for (i = 0; i < CLK_AUD_NR_CLK; i++) {
- cl = afe_priv->lookup[i];
- if (!cl)
- continue;
-
- clk = cl->clk;
- clk_unregister_gate(clk);
-
- clkdev_drop(cl);
- }
+ return devm_add_action_or_reset(afe->dev, mt8195_audsys_clk_unregister, afe);
}
diff --git a/sound/soc/mediatek/mt8195/mt8195-audsys-clk.h b/sound/soc/mediatek/mt8195/mt8195-audsys-clk.h
index 239d31016ba7..69db2dd1c9e0 100644
--- a/sound/soc/mediatek/mt8195/mt8195-audsys-clk.h
+++ b/sound/soc/mediatek/mt8195/mt8195-audsys-clk.h
@@ -10,6 +10,5 @@
#define _MT8195_AUDSYS_CLK_H_
int mt8195_audsys_clk_register(struct mtk_base_afe *afe);
-void mt8195_audsys_clk_unregister(struct mtk_base_afe *afe);
#endif
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index adb69d7820a8..4fb1ac8e1c4a 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2405,6 +2405,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
if (!snd_soc_dpcm_be_can_update(fe, be, stream))
continue;
+ if (!snd_soc_dpcm_can_be_prepared(fe, be, stream))
+ continue;
+
if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) &&
@@ -3042,3 +3045,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
+
+/*
+ * We can only prepare a BE DAI if any of it's FE are not prepared,
+ * running or paused for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ const enum snd_soc_dpcm_state state[] = {
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_PREPARE,
+ };
+
+ return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared);
diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c
index 4e0c48a36159..749e856dc601 100644
--- a/sound/soc/sof/amd/acp-ipc.c
+++ b/sound/soc/sof/amd/acp-ipc.c
@@ -209,7 +209,12 @@ int acp_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sp
acp_mailbox_read(sdev, offset, p, sz);
} else {
struct snd_pcm_substream *substream = sps->substream;
- struct acp_dsp_stream *stream = substream->runtime->private_data;
+ struct acp_dsp_stream *stream;
+
+ if (!substream || !substream->runtime)
+ return -ESTRPIPE;
+
+ stream = substream->runtime->private_data;
if (!stream)
return -ESTRPIPE;
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index b42b5982cbbc..d547318e0d32 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -438,8 +438,8 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg)
/* should we prevent DSP entering D3 ? */
if (!sdev->ipc_dump_printed)
dev_info(sdev->dev,
- "preventing DSP entering D3 state to preserve context\n");
- pm_runtime_get_noresume(sdev->dev);
+ "Attempting to prevent DSP from entering D3 state to preserve context\n");
+ pm_runtime_get_if_in_use(sdev->dev);
}
/* dump vital information to the logs */
diff --git a/sound/soc/sof/intel/hda-mlink.c b/sound/soc/sof/intel/hda-mlink.c
index 775582ab7494..b7cbf66badf5 100644
--- a/sound/soc/sof/intel/hda-mlink.c
+++ b/sound/soc/sof/intel/hda-mlink.c
@@ -19,6 +19,9 @@
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_MLINK)
+/* worst-case number of sublinks is used for sublink refcount array allocation only */
+#define HDAML_MAX_SUBLINKS (AZX_ML_LCTL_CPA_SHIFT - AZX_ML_LCTL_SPA_SHIFT)
+
/**
* struct hdac_ext2_link - HDAudio extended+alternate link
*
@@ -33,6 +36,7 @@
* @leptr: extended link pointer
* @eml_lock: mutual exclusion to access shared registers e.g. CPA/SPA bits
* in LCTL register
+ * @sublink_ref_count: array of refcounts, required to power-manage sublinks independently
* @base_ptr: pointer to shim/ip/shim_vs space
* @instance_offset: offset between each of @slcount instances managed by link
* @shim_offset: offset to SHIM register base
@@ -53,6 +57,7 @@ struct hdac_ext2_link {
u32 leptr;
struct mutex eml_lock; /* prevent concurrent access to e.g. CPA/SPA */
+ int sublink_ref_count[HDAML_MAX_SUBLINKS];
/* internal values computed from LCAP contents */
void __iomem *base_ptr;
@@ -68,6 +73,7 @@ struct hdac_ext2_link {
#define AZX_REG_SDW_SHIM_OFFSET 0x0
#define AZX_REG_SDW_IP_OFFSET 0x100
#define AZX_REG_SDW_VS_SHIM_OFFSET 0x6000
+#define AZX_REG_SDW_SHIM_PCMSyCM(y) (0x16 + 0x4 * (y))
/* only one instance supported */
#define AZX_REG_INTEL_DMIC_SHIM_OFFSET 0x0
@@ -91,7 +97,7 @@ struct hdac_ext2_link {
*/
static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
- void __iomem *ml_addr, int link_idx)
+ void __iomem *remap_addr, void __iomem *ml_addr, int link_idx)
{
struct hdac_ext_link *hlink = &h2link->hext_link;
u32 base_offset;
@@ -126,15 +132,16 @@ static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
link_idx, h2link->slcount);
/* find IP ID and offsets */
- h2link->leptr = readl(hlink->ml_addr + AZX_REG_ML_LEPTR);
+ h2link->leptr = readl(ml_addr + AZX_REG_ML_LEPTR);
h2link->elid = FIELD_GET(AZX_REG_ML_LEPTR_ID, h2link->leptr);
base_offset = FIELD_GET(AZX_REG_ML_LEPTR_PTR, h2link->leptr);
- h2link->base_ptr = hlink->ml_addr + base_offset;
+ h2link->base_ptr = remap_addr + base_offset;
switch (h2link->elid) {
case AZX_REG_ML_LEPTR_ID_SDW:
+ h2link->instance_offset = AZX_REG_SDW_INSTANCE_OFFSET;
h2link->shim_offset = AZX_REG_SDW_SHIM_OFFSET;
h2link->ip_offset = AZX_REG_SDW_IP_OFFSET;
h2link->shim_vs_offset = AZX_REG_SDW_VS_SHIM_OFFSET;
@@ -149,6 +156,7 @@ static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
link_idx, base_offset);
break;
case AZX_REG_ML_LEPTR_ID_INTEL_SSP:
+ h2link->instance_offset = AZX_REG_INTEL_SSP_INSTANCE_OFFSET;
h2link->shim_offset = AZX_REG_INTEL_SSP_SHIM_OFFSET;
h2link->ip_offset = AZX_REG_INTEL_SSP_IP_OFFSET;
h2link->shim_vs_offset = AZX_REG_INTEL_SSP_VS_SHIM_OFFSET;
@@ -333,6 +341,21 @@ static void hdaml_link_set_lsdiid(u32 __iomem *lsdiid, int dev_num)
writel(val, lsdiid);
}
+static void hdaml_shim_map_stream_ch(u16 __iomem *pcmsycm, int lchan, int hchan,
+ int stream_id, int dir)
+{
+ u16 val;
+
+ val = readw(pcmsycm);
+
+ u16p_replace_bits(&val, lchan, GENMASK(3, 0));
+ u16p_replace_bits(&val, hchan, GENMASK(7, 4));
+ u16p_replace_bits(&val, stream_id, GENMASK(13, 8));
+ u16p_replace_bits(&val, dir, BIT(15));
+
+ writew(val, pcmsycm);
+}
+
static void hdaml_lctl_offload_enable(u32 __iomem *lctl, bool enable)
{
u32 val = readl(lctl);
@@ -364,7 +387,7 @@ static int hda_ml_alloc_h2link(struct hdac_bus *bus, int index)
hlink->bus = bus;
hlink->ml_addr = bus->mlcap + AZX_ML_BASE + (AZX_ML_INTERVAL * index);
- ret = hdaml_lnk_enum(bus->dev, h2link, hlink->ml_addr, index);
+ ret = hdaml_lnk_enum(bus->dev, h2link, bus->remap_addr, hlink->ml_addr, index);
if (ret < 0) {
kfree(h2link);
return ret;
@@ -641,8 +664,13 @@ static int hdac_bus_eml_power_up_base(struct hdac_bus *bus, bool alt, int elid,
if (eml_lock)
mutex_lock(&h2link->eml_lock);
- if (++hlink->ref_count > 1)
- goto skip_init;
+ if (!alt) {
+ if (++hlink->ref_count > 1)
+ goto skip_init;
+ } else {
+ if (++h2link->sublink_ref_count[sublink] > 1)
+ goto skip_init;
+ }
ret = hdaml_link_init(hlink->ml_addr + AZX_REG_ML_LCTL, sublink);
@@ -684,9 +712,13 @@ static int hdac_bus_eml_power_down_base(struct hdac_bus *bus, bool alt, int elid
if (eml_lock)
mutex_lock(&h2link->eml_lock);
- if (--hlink->ref_count > 0)
- goto skip_shutdown;
-
+ if (!alt) {
+ if (--hlink->ref_count > 0)
+ goto skip_shutdown;
+ } else {
+ if (--h2link->sublink_ref_count[sublink] > 0)
+ goto skip_shutdown;
+ }
ret = hdaml_link_shutdown(hlink->ml_addr + AZX_REG_ML_LCTL, sublink);
skip_shutdown:
@@ -740,6 +772,40 @@ int hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num)
return 0;
} EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_set_lsdiid, SND_SOC_SOF_HDA_MLINK);
+/*
+ * the 'y' parameter comes from the PCMSyCM hardware register naming. 'y' refers to the
+ * PDI index, i.e. the FIFO used for RX or TX
+ */
+int hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
+ int channel_mask, int stream_id, int dir)
+{
+ struct hdac_ext2_link *h2link;
+ u16 __iomem *pcmsycm;
+ u16 val;
+
+ h2link = find_ext2_link(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
+ if (!h2link)
+ return -ENODEV;
+
+ pcmsycm = h2link->base_ptr + h2link->shim_offset +
+ h2link->instance_offset * sublink +
+ AZX_REG_SDW_SHIM_PCMSyCM(y);
+
+ mutex_lock(&h2link->eml_lock);
+
+ hdaml_shim_map_stream_ch(pcmsycm, 0, hweight32(channel_mask),
+ stream_id, dir);
+
+ mutex_unlock(&h2link->eml_lock);
+
+ val = readw(pcmsycm);
+
+ dev_dbg(bus->dev, "channel_mask %#x stream_id %d dir %d pcmscm %#x\n",
+ channel_mask, stream_id, dir, val);
+
+ return 0;
+} EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_map_stream_ch, SND_SOC_SOF_HDA_MLINK);
+
void hda_bus_ml_put_all(struct hdac_bus *bus)
{
struct hdac_ext_link *hlink;
@@ -836,6 +902,18 @@ struct hdac_ext_link *hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus)
}
EXPORT_SYMBOL_NS(hdac_bus_eml_dmic_get_hlink, SND_SOC_SOF_HDA_MLINK);
+struct hdac_ext_link *hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus)
+{
+ struct hdac_ext2_link *h2link;
+
+ h2link = find_ext2_link(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
+ if (!h2link)
+ return NULL;
+
+ return &h2link->hext_link;
+}
+EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_get_hlink, SND_SOC_SOF_HDA_MLINK);
+
int hdac_bus_eml_enable_offload(struct hdac_bus *bus, bool alt, int elid, bool enable)
{
struct hdac_ext2_link *h2link;
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index fc1eb8e2de2c..ba4ef290b634 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -2103,10 +2103,13 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
* For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is.
*/
- if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
+ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+ /* Clear stale command */
+ config->flags &= ~SOF_DAI_CONFIG_FLAGS_CMD_MASK;
config->flags |= flags;
- else
+ } else {
config->flags = flags;
+ }
/* only send the IPC if the widget is set up in the DSP */
if (swidget->use_count > 0) {
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 059eebf0a687..5abe616a2054 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -59,7 +59,7 @@ static const struct sof_topology_token ipc4_in_audio_format_tokens[] = {
audio_fmt.interleaving_style)},
{SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)},
- {SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
+ {SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, pin_index)},
{SOF_TKN_CAVS_AUDIO_FORMAT_IBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, buffer_size)},
@@ -79,7 +79,7 @@ static const struct sof_topology_token ipc4_out_audio_format_tokens[] = {
audio_fmt.interleaving_style)},
{SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)},
- {SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
+ {SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, pin_index)},
{SOF_TKN_CAVS_AUDIO_FORMAT_OBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, buffer_size)},
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 567db32173a8..d0ab6f390734 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -643,16 +643,17 @@ static int sof_pcm_probe(struct snd_soc_component *component)
"%s/%s",
plat_data->tplg_filename_prefix,
plat_data->tplg_filename);
- if (!tplg_filename)
- return -ENOMEM;
+ if (!tplg_filename) {
+ ret = -ENOMEM;
+ goto pm_error;
+ }
ret = snd_sof_load_topology(component, tplg_filename);
- if (ret < 0) {
+ if (ret < 0)
dev_err(component->dev, "error: failed to load DSP topology %d\n",
ret);
- return ret;
- }
+pm_error:
pm_runtime_mark_last_busy(component->dev);
pm_runtime_put_autosuspend(component->dev);
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 2fdbc53ca715..2b232442e84b 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -164,7 +164,7 @@ static int sof_resume(struct device *dev, bool runtime_resume)
ret = tplg_ops->set_up_all_pipelines(sdev, false);
if (ret < 0) {
dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret);
- return ret;
+ goto setup_fail;
}
}
@@ -178,6 +178,18 @@ static int sof_resume(struct device *dev, bool runtime_resume)
dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret);
}
+setup_fail:
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE)
+ if (ret < 0) {
+ /*
+ * Debugfs cannot be read in runtime suspend, so cache
+ * the contents upon failure. This allows to capture
+ * possible DSP coredump information.
+ */
+ sof_cache_debugfs(sdev);
+ }
+#endif
+
return ret;
}
diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c
index fff126808bc0..8d9e9d5f40e4 100644
--- a/sound/soc/sof/sof-client-probes.c
+++ b/sound/soc/sof/sof-client-probes.c
@@ -218,12 +218,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
ret = ipc->points_info(cdev, &desc, &num_desc);
if (ret < 0)
- goto exit;
-
- pm_runtime_mark_last_busy(dev);
- err = pm_runtime_put_autosuspend(dev);
- if (err < 0)
- dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
+ goto pm_error;
for (i = 0; i < num_desc; i++) {
offset = strlen(buf);
@@ -241,6 +236,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf));
kfree(desc);
+
+pm_error:
+ pm_runtime_mark_last_busy(dev);
+ err = pm_runtime_put_autosuspend(dev);
+ if (err < 0)
+ dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
+
exit:
kfree(buf);
return ret;
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index d3d536b0a8f5..f160dc454b44 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -586,6 +586,10 @@ static int sof_copy_tuples(struct snd_sof_dev *sdev, struct snd_soc_tplg_vendor_
if (*num_copied_tuples == tuples_size)
return 0;
}
+
+ /* stop when we've found the required token instances */
+ if (found == num_tokens * token_instance_num)
+ return 0;
}
/* next array */
@@ -1261,7 +1265,7 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s
if (num_sets > 1) {
struct snd_sof_tuple *new_tuples;
- num_tuples += token_list[object_token_list[i]].count * num_sets;
+ num_tuples += token_list[object_token_list[i]].count * (num_sets - 1);
new_tuples = krealloc(swidget->tuples,
sizeof(*new_tuples) * num_tuples, GFP_KERNEL);
if (!new_tuples) {
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 468c8e77de21..0b69cebc9a33 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -117,6 +117,9 @@ int tegra_pcm_open(struct snd_soc_component *component,
return ret;
}
+ /* Set wait time to 500ms by default */
+ substream->wait_time = 500;
+
return 0;
}
EXPORT_SYMBOL_GPL(tegra_pcm_open);