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-rw-r--r--sound/hda/codecs/realtek/alc269.c14
-rw-r--r--sound/soc/amd/acp/amd-acp70-acpi-match.c157
-rw-r--r--sound/soc/codecs/cs-amp-lib-test.c1
-rw-r--r--sound/soc/codecs/cs530x.c2
-rw-r--r--sound/soc/codecs/max98090.c6
-rw-r--r--sound/soc/codecs/rt721-sdca.c4
-rw-r--r--sound/soc/codecs/rt721-sdca.h1
-rw-r--r--sound/soc/fsl/fsl_micfil.c4
-rw-r--r--sound/soc/fsl/fsl_sai.c11
-rw-r--r--sound/soc/intel/avs/pcm.c3
-rw-r--r--sound/soc/intel/avs/probes.c18
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-ptl-match.c52
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-afe-pcm.c1
-rw-r--r--sound/soc/mediatek/mt8365/mt8365-afe-pcm.c1
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c2
-rw-r--r--sound/soc/renesas/rz-ssi.c25
-rw-r--r--sound/soc/sdw_utils/soc_sdw_utils.c1
-rw-r--r--sound/usb/mixer_s1810c.c35
18 files changed, 234 insertions, 104 deletions
diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c
index 8ad5febd822a..4aec5067c59d 100644
--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -3736,6 +3736,7 @@ enum {
ALC285_FIXUP_ASUS_GA605K_I2C_SPEAKER2_TO_DAC1,
ALC269_FIXUP_POSITIVO_P15X_HEADSET_MIC,
ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK,
+ ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE,
};
/* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -6172,6 +6173,16 @@ static const struct hda_fixup alc269_fixups[] = {
{ 0x1e, 0x90170150 }, /* Internal Speaker */
{ }
},
+ },
+ [ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */
+ { 0x1a, 0x22a190a0 }, /* dock mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
}
};
@@ -6578,6 +6589,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX),
SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS),
+ SND_PCI_QUIRK(0x103c, 0x8c2d, "HP Victus 15-fa1xxx (MB 8C2D)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
SND_PCI_QUIRK(0x103c, 0x8c30, "HP Victus 15-fb1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
@@ -6959,6 +6971,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0xa554, "VAIO VJFH52", ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xa559, "VAIO RPL", ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
@@ -7080,6 +7093,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2),
+ HDA_CODEC_QUIRK(0x17aa, 0x391c, "Lenovo Yoga 7 2-in-1 14AKP10", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2),
diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c
index dcecac792e6d..871b4f054a84 100644
--- a/sound/soc/amd/acp/amd-acp70-acpi-match.c
+++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c
@@ -30,6 +30,20 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
.group_id = 1
};
+static const struct snd_soc_acpi_endpoint spk_2_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 2,
+ .group_id = 1
+};
+
+static const struct snd_soc_acpi_endpoint spk_3_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 3,
+ .group_id = 1
+};
+
static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = {
{
.adr = 0x000030025D071101ull,
@@ -112,6 +126,134 @@ static const struct snd_soc_acpi_adr_device rt1320_1_single_adr[] = {
}
};
+static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
+ { /* Jack Playback Endpoint */
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* DMIC Capture Endpoint */
+ .num = 1,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Jack Capture Endpoint */
+ .num = 2,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Speaker Playback Endpoint */
+ .num = 3,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+};
+
+static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = {
+ {
+ .adr = 0x00003001FA424301ull,
+ .num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
+ .endpoints = cs42l43_endpoints,
+ .name_prefix = "cs42l43"
+ }
+};
+
+static const struct snd_soc_acpi_adr_device cs42l43_1_cs35l56x4_1_adr[] = {
+ {
+ .adr = 0x00013001FA424301ull,
+ .num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
+ .endpoints = cs42l43_endpoints,
+ .name_prefix = "cs42l43"
+ },
+ {
+ .adr = 0x00013001FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_l_endpoint,
+ .name_prefix = "AMP1"
+ },
+ {
+ .adr = 0x00013101FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_r_endpoint,
+ .name_prefix = "AMP2"
+ },
+ {
+ .adr = 0x00013201FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_2_endpoint,
+ .name_prefix = "AMP3"
+ },
+ {
+ .adr = 0x00013301FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_3_endpoint,
+ .name_prefix = "AMP4"
+ },
+};
+
+static const struct snd_soc_acpi_adr_device cs35l56x4_1_adr[] = {
+ {
+ .adr = 0x00013301FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_l_endpoint,
+ .name_prefix = "AMP1"
+ },
+ {
+ .adr = 0x00013201FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_r_endpoint,
+ .name_prefix = "AMP2"
+ },
+ {
+ .adr = 0x00013101FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_2_endpoint,
+ .name_prefix = "AMP3"
+ },
+ {
+ .adr = 0x00013001FA355601ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_3_endpoint,
+ .name_prefix = "AMP4"
+ },
+};
+
+static const struct snd_soc_acpi_link_adr acp70_cs42l43_l1_cs35l56x4_l1[] = {
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(cs42l43_1_cs35l56x4_1_adr),
+ .adr_d = cs42l43_1_cs35l56x4_1_adr,
+ },
+ {}
+};
+
+static const struct snd_soc_acpi_link_adr acp70_cs42l43_l0_cs35l56x4_l1[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(cs42l43_0_adr),
+ .adr_d = cs42l43_0_adr,
+ },
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(cs35l56x4_1_adr),
+ .adr_d = cs35l56x4_1_adr,
+ },
+ {}
+};
+
+static const struct snd_soc_acpi_link_adr acp70_cs35l56x4_l1[] = {
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(cs35l56x4_1_adr),
+ .adr_d = cs35l56x4_1_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_link_adr acp70_rt722_only[] = {
{
.mask = BIT(0),
@@ -151,6 +293,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = {
.links = acp70_4_in_1_sdca,
.drv_name = "amd_sdw",
},
+ {
+ .link_mask = BIT(0) | BIT(1),
+ .links = acp70_cs42l43_l0_cs35l56x4_l1,
+ .drv_name = "amd_sdw",
+ },
+ {
+ .link_mask = BIT(1),
+ .links = acp70_cs42l43_l1_cs35l56x4_l1,
+ .drv_name = "amd_sdw",
+ },
+ {
+ .link_mask = BIT(1),
+ .links = acp70_cs35l56x4_l1,
+ .drv_name = "amd_sdw",
+ },
{},
};
EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sdw_machines);
diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c
index 2fde84309338..3406887cdfa2 100644
--- a/sound/soc/codecs/cs-amp-lib-test.c
+++ b/sound/soc/codecs/cs-amp-lib-test.c
@@ -7,6 +7,7 @@
#include <kunit/resource.h>
#include <kunit/test.h>
+#include <kunit/test-bug.h>
#include <kunit/static_stub.h>
#include <linux/device/faux.h>
#include <linux/firmware/cirrus/cs_dsp.h>
diff --git a/sound/soc/codecs/cs530x.c b/sound/soc/codecs/cs530x.c
index b9eff240b929..535387cd7aa3 100644
--- a/sound/soc/codecs/cs530x.c
+++ b/sound/soc/codecs/cs530x.c
@@ -793,7 +793,7 @@ static int cs530x_set_sysclk(struct snd_soc_component *component, int clk_id,
case CS530X_SYSCLK_SRC_PLL:
break;
default:
- dev_err(component->dev, "Invalid clock id %d\n", clk_id);
+ dev_err(component->dev, "Invalid sysclk source: %d\n", source);
return -EINVAL;
}
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index cb1508fc99f8..5aff5a459a43 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1239,6 +1239,8 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE,
M98090_DIGMIC4_SHIFT, 0, max98090_shdn_event,
SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SUPPLY("DMIC34_HPF", M98090_REG_FILTER_CONFIG,
+ M98090_FLT_DMIC34HPF_SHIFT, 0, NULL, 0),
};
static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
@@ -1427,8 +1429,8 @@ static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
/* DMIC inputs */
{"DMIC3", NULL, "DMIC3_ENA"},
{"DMIC4", NULL, "DMIC4_ENA"},
- {"DMIC3", NULL, "AHPF"},
- {"DMIC4", NULL, "AHPF"},
+ {"DMIC3", NULL, "DMIC34_HPF"},
+ {"DMIC4", NULL, "DMIC34_HPF"},
};
static int max98090_add_widgets(struct snd_soc_component *component)
diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c
index a4bd29d7220b..5f7b505d5414 100644
--- a/sound/soc/codecs/rt721-sdca.c
+++ b/sound/soc/codecs/rt721-sdca.c
@@ -281,6 +281,10 @@ static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721)
rt_sdca_index_write(rt721->mbq_regmap, RT721_BOOST_CTRL,
RT721_BST_4CH_TOP_GATING_CTRL1, 0x002a);
regmap_write(rt721->regmap, 0x2f58, 0x07);
+
+ regmap_write(rt721->regmap, 0x2f51, 0x00);
+ rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT,
+ RT721_MISC_CTL, 0x0004);
}
static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721)
diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h
index 71fac9cd8739..24ce188562ba 100644
--- a/sound/soc/codecs/rt721-sdca.h
+++ b/sound/soc/codecs/rt721-sdca.h
@@ -137,6 +137,7 @@ struct rt721_sdca_dmic_kctrl_priv {
#define RT721_HDA_LEGACY_UAJ_CTL 0x02
#define RT721_HDA_LEGACY_CTL1 0x05
#define RT721_HDA_LEGACY_RESET_CTL 0x06
+#define RT721_MISC_CTL 0x07
#define RT721_XU_REL_CTRL 0x0c
#define RT721_GE_REL_CTRL1 0x0d
#define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index aabd90a8b3ec..cac26ba0aa4b 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -131,7 +131,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx943 = {
.fifos = 8,
.fifo_depth = 32,
.dataline = 0xf,
- .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE,
.use_edma = true,
.use_verid = true,
.volume_sx = false,
@@ -823,7 +823,7 @@ static int fsl_micfil_hw_params(struct snd_pcm_substream *substream,
break;
}
- if (format == SNDRV_PCM_FORMAT_DSD_U32_BE) {
+ if (format == SNDRV_PCM_FORMAT_DSD_U32_LE) {
micfil->dec_bypass = true;
/*
* According to equation 29 in RM:
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 757e7868e322..72bfc91e21b9 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -353,7 +353,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
break;
case SND_SOC_DAIFMT_PDM:
val_cr2 |= FSL_SAI_CR2_BCP;
- val_cr4 &= ~FSL_SAI_CR4_MF;
sai->is_pdm_mode = true;
break;
case SND_SOC_DAIFMT_RIGHT_J:
@@ -638,7 +637,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
val_cr5 |= FSL_SAI_CR5_WNW(slot_width);
val_cr5 |= FSL_SAI_CR5_W0W(slot_width);
- if (sai->is_lsb_first || sai->is_pdm_mode)
+ if (sai->is_lsb_first)
val_cr5 |= FSL_SAI_CR5_FBT(0);
else
val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
@@ -653,12 +652,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
val_cr4 |= FSL_SAI_CR4_CHMOD;
/*
- * For SAI provider mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will
- * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4),
- * RCR5(TCR5) for playback(capture), or there will be sync error.
+ * When Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will provide bclk and
+ * frame clock for Tx(Rx). We should set RCR4(TCR4), RCR5(TCR5)
+ * for playback(capture), or there will be sync error.
*/
- if (!sai->is_consumer_mode[tx] && fsl_sai_dir_is_synced(sai, adir)) {
+ if (fsl_sai_dir_is_synced(sai, adir)) {
regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs),
FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK |
FSL_SAI_CR4_CHMOD_MASK,
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index d31058e2de5b..80c001120cdd 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -651,6 +651,7 @@ static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_
data = snd_soc_dai_get_dma_data(dai, substream);
+ disable_work_sync(&data->period_elapsed_work);
snd_hdac_ext_stream_release(data->host_stream, HDAC_EXT_STREAM_TYPE_HOST);
avs_dai_shutdown(substream, dai);
}
@@ -754,6 +755,8 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
data = snd_soc_dai_get_dma_data(dai, substream);
host_stream = data->host_stream;
+ if (runtime->state == SNDRV_PCM_STATE_XRUN)
+ hdac_stream(host_stream)->prepared = false;
if (hdac_stream(host_stream)->prepared)
return 0;
diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c
index 693ecfe68fd0..74096236984a 100644
--- a/sound/soc/intel/avs/probes.c
+++ b/sound/soc/intel/avs/probes.c
@@ -14,8 +14,8 @@
#include "debug.h"
#include "messages.h"
-static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id node_id,
- size_t buffer_size)
+static int avs_dsp_init_probe(struct avs_dev *adev, struct snd_compr_params *params, int bps,
+ union avs_connector_node_id node_id, size_t buffer_size)
{
struct avs_probe_cfg cfg = {{0}};
struct avs_module_entry mentry;
@@ -27,12 +27,16 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
return ret;
/*
- * Probe module uses no cycles, audio data format and input and output
- * frame sizes are unused. It is also not owned by any pipeline.
+ * Probe module uses no cycles, input and output frame sizes are unused.
+ * It is also not owned by any pipeline.
*/
cfg.base.ibs = 1;
/* BSS module descriptor is always segment of index=2. */
cfg.base.is_pages = mentry.segments[2].flags.length;
+ cfg.base.audio_fmt.sampling_freq = params->codec.sample_rate;
+ cfg.base.audio_fmt.bit_depth = bps;
+ cfg.base.audio_fmt.num_channels = params->codec.ch_out;
+ cfg.base.audio_fmt.valid_bit_depth = bps;
cfg.gtw_cfg.node_id = node_id;
cfg.gtw_cfg.dma_buffer_size = buffer_size;
@@ -128,8 +132,6 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream,
struct hdac_ext_stream *host_stream = avs_compr_get_host_stream(cstream);
struct snd_compr_runtime *rtd = cstream->runtime;
struct avs_dev *adev = to_avs_dev(dai->dev);
- /* compr params do not store bit depth, default to S32_LE. */
- snd_pcm_format_t format = SNDRV_PCM_FORMAT_S32_LE;
unsigned int format_val;
int bps, ret;
@@ -142,7 +144,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream,
ret = snd_compr_malloc_pages(cstream, rtd->buffer_size);
if (ret < 0)
return ret;
- bps = snd_pcm_format_physical_width(format);
+ bps = snd_pcm_format_physical_width(params->codec.format);
if (bps < 0)
return bps;
format_val = snd_hdac_stream_format(params->codec.ch_out, bps, params->codec.sample_rate);
@@ -166,7 +168,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream,
node_id.vindex = hdac_stream(host_stream)->stream_tag - 1;
node_id.dma_type = AVS_DMA_HDA_HOST_INPUT;
- ret = avs_dsp_init_probe(adev, node_id, rtd->dma_bytes);
+ ret = avs_dsp_init_probe(adev, params, bps, node_id, rtd->dma_bytes);
if (ret < 0) {
dev_err(dai->dev, "probe init failed: %d\n", ret);
avs_dsp_enable_d0ix(adev);
diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c
index 3c8b10e21ceb..4853f4f31786 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c
@@ -227,33 +227,6 @@ static const struct snd_soc_acpi_endpoint cs42l43_amp_spkagg_endpoints[] = {
},
};
-static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
- { /* Jack Playback Endpoint */
- .num = 0,
- .aggregated = 0,
- .group_position = 0,
- .group_id = 0,
- },
- { /* DMIC Capture Endpoint */
- .num = 1,
- .aggregated = 0,
- .group_position = 0,
- .group_id = 0,
- },
- { /* Jack Capture Endpoint */
- .num = 2,
- .aggregated = 0,
- .group_position = 0,
- .group_id = 0,
- },
- { /* Speaker Playback Endpoint */
- .num = 3,
- .aggregated = 0,
- .group_position = 0,
- .group_id = 0,
- },
-};
-
static const struct snd_soc_acpi_adr_device cs42l43_2_adr[] = {
{
.adr = 0x00023001fa424301ull,
@@ -305,15 +278,6 @@ static const struct snd_soc_acpi_adr_device cs35l56_3_3amp_adr[] = {
}
};
-static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = {
- {
- .adr = 0x00033001FA424301ull,
- .num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
- .endpoints = cs42l43_endpoints,
- .name_prefix = "cs42l43"
- }
-};
-
static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = {
{
.adr = 0x000030025D071101ull,
@@ -486,15 +450,6 @@ static const struct snd_soc_acpi_link_adr ptl_cs42l43_l2_cs35l56x6_l13[] = {
{}
};
-static const struct snd_soc_acpi_link_adr ptl_cs42l43_l3[] = {
- {
- .mask = BIT(3),
- .num_adr = ARRAY_SIZE(cs42l43_3_adr),
- .adr_d = cs42l43_3_adr,
- },
- {}
-};
-
static const struct snd_soc_acpi_link_adr ptl_rt721_l0[] = {
{
.mask = BIT(0),
@@ -714,13 +669,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = {
},
{
.link_mask = BIT(3),
- .links = ptl_cs42l43_l3,
- .drv_name = "sof_sdw",
- .sof_tplg_filename = "sof-ptl-cs42l43-l3.tplg",
- .get_function_tplg_files = sof_sdw_get_tplg_files,
- },
- {
- .link_mask = BIT(3),
.links = ptl_sdw_rt712_vb_l3_rt1320_l3,
.drv_name = "sof_sdw",
.machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb,
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
index 5d025ad72263..c63b3444bc17 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
@@ -3176,7 +3176,6 @@ err_pm_put:
static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev)
{
- pm_runtime_disable(&pdev->dev);
if (!pm_runtime_status_suspended(&pdev->dev))
mt8195_afe_runtime_suspend(&pdev->dev);
}
diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c
index 10793bbe9275..d48252cd96ac 100644
--- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c
+++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c
@@ -2238,7 +2238,6 @@ static void mt8365_afe_pcm_dev_remove(struct platform_device *pdev)
mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_AFE);
- pm_runtime_disable(&pdev->dev);
if (!pm_runtime_status_suspended(&pdev->dev))
mt8365_afe_runtime_suspend(&pdev->dev);
}
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 06a802f9dba5..67e9ca18883c 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -377,9 +377,9 @@ static void q6asm_audio_client_free_buf(struct audio_client *ac,
spin_lock_irqsave(&ac->lock, flags);
port->num_periods = 0;
+ spin_unlock_irqrestore(&ac->lock, flags);
kfree(port->buf);
port->buf = NULL;
- spin_unlock_irqrestore(&ac->lock, flags);
}
/**
diff --git a/sound/soc/renesas/rz-ssi.c b/sound/soc/renesas/rz-ssi.c
index e00940814157..81b883e8ac92 100644
--- a/sound/soc/renesas/rz-ssi.c
+++ b/sound/soc/renesas/rz-ssi.c
@@ -85,6 +85,7 @@ struct rz_ssi_stream {
struct snd_pcm_substream *substream;
int fifo_sample_size; /* sample capacity of SSI FIFO */
int dma_buffer_pos; /* The address for the next DMA descriptor */
+ int completed_dma_buf_pos; /* The address of the last completed DMA descriptor. */
int period_counter; /* for keeping track of periods transferred */
int sample_width;
int buffer_pos; /* current frame position in the buffer */
@@ -215,6 +216,7 @@ static void rz_ssi_stream_init(struct rz_ssi_stream *strm,
rz_ssi_set_substream(strm, substream);
strm->sample_width = samples_to_bytes(runtime, 1);
strm->dma_buffer_pos = 0;
+ strm->completed_dma_buf_pos = 0;
strm->period_counter = 0;
strm->buffer_pos = 0;
@@ -437,6 +439,10 @@ static void rz_ssi_pointer_update(struct rz_ssi_stream *strm, int frames)
snd_pcm_period_elapsed(strm->substream);
strm->period_counter = current_period;
}
+
+ strm->completed_dma_buf_pos += runtime->period_size;
+ if (strm->completed_dma_buf_pos >= runtime->buffer_size)
+ strm->completed_dma_buf_pos = 0;
}
static int rz_ssi_pio_recv(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm)
@@ -778,10 +784,14 @@ no_dma:
return -ENODEV;
}
-static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi)
+static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm)
{
+ struct snd_pcm_substream *substream = strm->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
int ret;
+ strm->dma_buffer_pos = strm->completed_dma_buf_pos + runtime->period_size;
+
if (rz_ssi_is_stream_running(&ssi->playback) ||
rz_ssi_is_stream_running(&ssi->capture))
return 0;
@@ -794,16 +804,6 @@ static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi)
ssi->hw_params_cache.channels);
}
-static void rz_ssi_streams_suspend(struct rz_ssi_priv *ssi)
-{
- if (rz_ssi_is_stream_running(&ssi->playback) ||
- rz_ssi_is_stream_running(&ssi->capture))
- return;
-
- ssi->playback.dma_buffer_pos = 0;
- ssi->capture.dma_buffer_pos = 0;
-}
-
static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
@@ -813,7 +813,7 @@ static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
- ret = rz_ssi_trigger_resume(ssi);
+ ret = rz_ssi_trigger_resume(ssi, strm);
if (ret)
return ret;
@@ -852,7 +852,6 @@ static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_SUSPEND:
rz_ssi_stop(ssi, strm);
- rz_ssi_streams_suspend(ssi);
break;
case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
index 270c66b90228..f7c8c16308de 100644
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -638,7 +638,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
{
.direction = {true, false},
.dai_name = "cs42l43-dp6",
- .component_name = "cs42l43",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
.init = asoc_sdw_cs42l43_spk_init,
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
index 15960d25e748..6e09e074c0e7 100644
--- a/sound/usb/mixer_s1810c.c
+++ b/sound/usb/mixer_s1810c.c
@@ -178,7 +178,7 @@ snd_sc1810c_get_status_field(struct usb_device *dev,
pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1;
pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2;
- ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+ ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
SC1810C_SET_STATE_REQ,
SC1810C_SET_STATE_REQTYPE,
(*seqnum), 0, &pkt_out, sizeof(pkt_out));
@@ -597,15 +597,6 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
if (!list_empty(&chip->mixer_list))
return 0;
- dev_info(&dev->dev,
- "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
- if (chip->setup == 1)
- dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
- else if (chip->setup == 2)
- dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
- else
- dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
-
ret = snd_s1810c_init_mixer_maps(chip);
if (ret < 0)
return ret;
@@ -634,16 +625,28 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
if (ret < 0)
return ret;
- // The 1824c has a Mono Main switch instead of a
- // A/B select switch.
- if (mixer->chip->usb_id == USB_ID(0x194f, 0x010d)) {
- ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw);
+ switch (chip->usb_id) {
+ case USB_ID(0x194f, 0x010c): /* Presonus Studio 1810c */
+ dev_info(&dev->dev,
+ "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
+ if (chip->setup == 1)
+ dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
+ else if (chip->setup == 2)
+ dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
+ else
+ dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
if (ret < 0)
return ret;
- } else if (mixer->chip->usb_id == USB_ID(0x194f, 0x010c)) {
- ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
+
+ break;
+ case USB_ID(0x194f, 0x010d): /* Presonus Studio 1824c */
+ ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw);
if (ret < 0)
return ret;
+
+ break;
}
return ret;
generated by cgit (git 2.34.1) at 2025-11-01 17:31:25 +0000