Maksytka/linux
1
1
Fork 0
mirror of https://github.com/torvalds/linux.git synced 2025-11-30 23:16:01 +07:00
Code Issues Packages Projects Releases Wiki Activity

Merge tag 'sound-6.18-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Browse Source 
Pull sound fixes from Takashi Iwai:
 "A collection of small fixes. It became slightly bigger than usual due
  to timing issues (holidays, etc), but all changes are rather
  device-specific fixes, so not really worrisome.

   - ASoC Cirrus codec fixes for AMD

   - Various fixes for ASoC Intel AVS, Qualcomm, SoundWire, FSL,
     Mediatek, Renesas

   - A few HD-audio quirks, and USB-audio regression fixes for Presonus"

* tag 'sound-6.18-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
  ALSA: hda/realtek: Enable mic on Vaio RPL
  ASoC: dt-bindings: pm4125-sdw: correct number of soundwire ports
  ASoC: renesas: rz-ssi: Use proper dma_buffer_pos after resume
  ASoC: soc_sdw_utils: remove cs42l43 component_name
  ASoC: fsl_sai: Fix sync error in consumer mode
  ASoC: Fix build for sdw_utils
  ALSA: hda/realtek: Fix mute led for HP Victus 15-fa1xxx (MB 8C2D)
  ASoC: rt721: fix prepare clock stop failed
  ALSA: usb-audio: don't log messages meant for 1810c when initializing 1824c
  ASoC: mediatek: Fix double pm_runtime_disable in remove functions
  ASoC: fsl_micfil: correct the endian format for DSD
  ASoC: fsl_sai: fix bit order for DSD format
  ASoC: Intel: avs: Use snd_codec format when initializing probe
  ASoC: Intel: avs: Disable periods-elapsed work when closing PCM
  ASoC: Intel: avs: Unprepare a stream when XRUN occurs
  ASoC: sdw_utils: add name_prefix for rt1321 part id
  ASoC: qdsp6: q6asm: do not sleep while atomic
  ASoC: Intel: soc-acpi-intel-ptl-match: Remove cs42l43 match from sdw link3
  ASOC: max98090/91: fix for filter configuration: AHPF removed DMIC2_HPF added
  ASoC: amd: acp: Add ACP7.0 match entries for cs35l56 and cs42l43
  ...
This commit is contained in:
Linus Torvalds
2025-10-31 07:29:09 -07:00
parent 3ad81aa520 390db60f8e
commit 58fdd8484c
19 changed files with 237 additions and 107 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.yaml
View File

@@ -32,7 +32,7 @@ properties:
$ref: /schemas/types.yaml#/definitions/uint32-array
minItems: 2
maxItems: 2
maxItems: 4
items:
enum: [1, 2, 3, 4]
@@ -48,7 +48,7 @@ properties:
$ref: /schemas/types.yaml#/definitions/uint32-array
minItems: 2
maxItems: 2
maxItems: 5
items:
enum: [1, 2, 3, 4, 5]

14
sound/hda/codecs/realtek/alc269.c
View File

@@ -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),

157
sound/soc/amd/acp/amd-acp70-acpi-match.c
View File

@@ -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);

1
sound/soc/codecs/cs-amp-lib-test.c
View File

@@ -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>

2
sound/soc/codecs/cs530x.c
View File

@@ -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;
}

6
sound/soc/codecs/max98090.c
View File

@@ -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)

4
sound/soc/codecs/rt721-sdca.c
View File

@@ -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)

1
sound/soc/codecs/rt721-sdca.h
View File

@@ -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

4
sound/soc/fsl/fsl_micfil.c
View File

@@ -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:

11
sound/soc/fsl/fsl_sai.c
View File

@@ -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,

3
sound/soc/intel/avs/pcm.c
View File

@@ -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;

18
sound/soc/intel/avs/probes.c
View File

@@ -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);

52
sound/soc/intel/common/soc-acpi-intel-ptl-match.c
View File

@@ -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),
@@ -712,13 +667,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = {
.sof_tplg_filename = "sof-ptl-rt722.tplg",
.get_function_tplg_files = sof_sdw_get_tplg_files,
},
{
.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,

1
sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
View File

@@ -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);
}

1
sound/soc/mediatek/mt8365/mt8365-afe-pcm.c
View File

@@ -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);
}

2
sound/soc/qcom/qdsp6/q6asm.c
View File

@@ -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);
}
/**

25
sound/soc/renesas/rz-ssi.c
View File

@@ -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:

1
sound/soc/sdw_utils/soc_sdw_utils.c
View File

@@ -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,

37
sound/usb/mixer_s1810c.c
View File

@@ -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);
if (ret < 0)
return ret;
} else if (mixer->chip->usb_id == USB_ID(0x194f, 0x010c)) {
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;
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;