Linux kernel mirror (for testing)
git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
kernel
os
linux
Merge tag 'sound-6.18-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
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
...
+237
-107
+2
-2
Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.yaml
+2
-2
Documentation/devicetree/bindings/sound/qcom,pm4125-sdw.yaml
···
32
32
33
33
$ref: /schemas/types.yaml#/definitions/uint32-array
34
34
minItems: 2
35
-
maxItems: 2
35
+
maxItems: 4
36
36
items:
37
37
enum: [1, 2, 3, 4]
38
38
···
48
48
49
49
$ref: /schemas/types.yaml#/definitions/uint32-array
50
50
minItems: 2
51
-
maxItems: 2
51
+
maxItems: 5
52
52
items:
53
53
enum: [1, 2, 3, 4, 5]
54
54
+14
sound/hda/codecs/realtek/alc269.c
+14
sound/hda/codecs/realtek/alc269.c
···
3736
3736
ALC285_FIXUP_ASUS_GA605K_I2C_SPEAKER2_TO_DAC1,
3737
3737
ALC269_FIXUP_POSITIVO_P15X_HEADSET_MIC,
3738
3738
ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK,
3739
+
ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE,
3739
3740
};
3740
3741
3741
3742
/* A special fixup for Lenovo C940 and Yoga Duet 7;
···
6173
6172
{ 0x1e, 0x90170150 }, /* Internal Speaker */
6174
6173
{ }
6175
6174
},
6175
+
},
6176
+
[ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE] = {
6177
+
.type = HDA_FIXUP_PINS,
6178
+
.v.pins = (const struct hda_pintbl[]) {
6179
+
{ 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */
6180
+
{ 0x1a, 0x22a190a0 }, /* dock mic */
6181
+
{ }
6182
+
},
6183
+
.chained = true,
6184
+
.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
6176
6185
}
6177
6186
};
6178
6187
···
6589
6578
SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX),
6590
6579
SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2),
6591
6580
SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS),
6581
+
SND_PCI_QUIRK(0x103c, 0x8c2d, "HP Victus 15-fa1xxx (MB 8C2D)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
6592
6582
SND_PCI_QUIRK(0x103c, 0x8c30, "HP Victus 15-fb1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
6593
6583
SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
6594
6584
SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
···
6971
6959
SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
6972
6960
SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
6973
6961
SND_PCI_QUIRK(0x1558, 0xa554, "VAIO VJFH52", ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE),
6962
+
SND_PCI_QUIRK(0x1558, 0xa559, "VAIO RPL", ALC256_FIXUP_VAIO_RPL_MIC_NO_PRESENCE),
6974
6963
SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
6975
6964
SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
6976
6965
SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
···
7093
7080
SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD),
7094
7081
SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD),
7095
7082
SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2),
7083
+
HDA_CODEC_QUIRK(0x17aa, 0x391c, "Lenovo Yoga 7 2-in-1 14AKP10", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
7096
7084
SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2),
7097
7085
SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2),
7098
7086
SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2),
+157
sound/soc/amd/acp/amd-acp70-acpi-match.c
+157
sound/soc/amd/acp/amd-acp70-acpi-match.c
···
30
30
.group_id = 1
31
31
};
32
32
33
+
static const struct snd_soc_acpi_endpoint spk_2_endpoint = {
34
+
.num = 0,
35
+
.aggregated = 1,
36
+
.group_position = 2,
37
+
.group_id = 1
38
+
};
39
+
40
+
static const struct snd_soc_acpi_endpoint spk_3_endpoint = {
41
+
.num = 0,
42
+
.aggregated = 1,
43
+
.group_position = 3,
44
+
.group_id = 1
45
+
};
46
+
33
47
static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = {
34
48
{
35
49
.adr = 0x000030025D071101ull,
···
126
112
}
127
113
};
128
114
115
+
static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
116
+
{ /* Jack Playback Endpoint */
117
+
.num = 0,
118
+
.aggregated = 0,
119
+
.group_position = 0,
120
+
.group_id = 0,
121
+
},
122
+
{ /* DMIC Capture Endpoint */
123
+
.num = 1,
124
+
.aggregated = 0,
125
+
.group_position = 0,
126
+
.group_id = 0,
127
+
},
128
+
{ /* Jack Capture Endpoint */
129
+
.num = 2,
130
+
.aggregated = 0,
131
+
.group_position = 0,
132
+
.group_id = 0,
133
+
},
134
+
{ /* Speaker Playback Endpoint */
135
+
.num = 3,
136
+
.aggregated = 0,
137
+
.group_position = 0,
138
+
.group_id = 0,
139
+
},
140
+
};
141
+
142
+
static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = {
143
+
{
144
+
.adr = 0x00003001FA424301ull,
145
+
.num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
146
+
.endpoints = cs42l43_endpoints,
147
+
.name_prefix = "cs42l43"
148
+
}
149
+
};
150
+
151
+
static const struct snd_soc_acpi_adr_device cs42l43_1_cs35l56x4_1_adr[] = {
152
+
{
153
+
.adr = 0x00013001FA424301ull,
154
+
.num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
155
+
.endpoints = cs42l43_endpoints,
156
+
.name_prefix = "cs42l43"
157
+
},
158
+
{
159
+
.adr = 0x00013001FA355601ull,
160
+
.num_endpoints = 1,
161
+
.endpoints = &spk_l_endpoint,
162
+
.name_prefix = "AMP1"
163
+
},
164
+
{
165
+
.adr = 0x00013101FA355601ull,
166
+
.num_endpoints = 1,
167
+
.endpoints = &spk_r_endpoint,
168
+
.name_prefix = "AMP2"
169
+
},
170
+
{
171
+
.adr = 0x00013201FA355601ull,
172
+
.num_endpoints = 1,
173
+
.endpoints = &spk_2_endpoint,
174
+
.name_prefix = "AMP3"
175
+
},
176
+
{
177
+
.adr = 0x00013301FA355601ull,
178
+
.num_endpoints = 1,
179
+
.endpoints = &spk_3_endpoint,
180
+
.name_prefix = "AMP4"
181
+
},
182
+
};
183
+
184
+
static const struct snd_soc_acpi_adr_device cs35l56x4_1_adr[] = {
185
+
{
186
+
.adr = 0x00013301FA355601ull,
187
+
.num_endpoints = 1,
188
+
.endpoints = &spk_l_endpoint,
189
+
.name_prefix = "AMP1"
190
+
},
191
+
{
192
+
.adr = 0x00013201FA355601ull,
193
+
.num_endpoints = 1,
194
+
.endpoints = &spk_r_endpoint,
195
+
.name_prefix = "AMP2"
196
+
},
197
+
{
198
+
.adr = 0x00013101FA355601ull,
199
+
.num_endpoints = 1,
200
+
.endpoints = &spk_2_endpoint,
201
+
.name_prefix = "AMP3"
202
+
},
203
+
{
204
+
.adr = 0x00013001FA355601ull,
205
+
.num_endpoints = 1,
206
+
.endpoints = &spk_3_endpoint,
207
+
.name_prefix = "AMP4"
208
+
},
209
+
};
210
+
211
+
static const struct snd_soc_acpi_link_adr acp70_cs42l43_l1_cs35l56x4_l1[] = {
212
+
{
213
+
.mask = BIT(1),
214
+
.num_adr = ARRAY_SIZE(cs42l43_1_cs35l56x4_1_adr),
215
+
.adr_d = cs42l43_1_cs35l56x4_1_adr,
216
+
},
217
+
{}
218
+
};
219
+
220
+
static const struct snd_soc_acpi_link_adr acp70_cs42l43_l0_cs35l56x4_l1[] = {
221
+
{
222
+
.mask = BIT(0),
223
+
.num_adr = ARRAY_SIZE(cs42l43_0_adr),
224
+
.adr_d = cs42l43_0_adr,
225
+
},
226
+
{
227
+
.mask = BIT(1),
228
+
.num_adr = ARRAY_SIZE(cs35l56x4_1_adr),
229
+
.adr_d = cs35l56x4_1_adr,
230
+
},
231
+
{}
232
+
};
233
+
234
+
static const struct snd_soc_acpi_link_adr acp70_cs35l56x4_l1[] = {
235
+
{
236
+
.mask = BIT(1),
237
+
.num_adr = ARRAY_SIZE(cs35l56x4_1_adr),
238
+
.adr_d = cs35l56x4_1_adr,
239
+
},
240
+
{}
241
+
};
242
+
129
243
static const struct snd_soc_acpi_link_adr acp70_rt722_only[] = {
130
244
{
131
245
.mask = BIT(0),
···
291
149
{
292
150
.link_mask = BIT(0) | BIT(1),
293
151
.links = acp70_4_in_1_sdca,
152
+
.drv_name = "amd_sdw",
153
+
},
154
+
{
155
+
.link_mask = BIT(0) | BIT(1),
156
+
.links = acp70_cs42l43_l0_cs35l56x4_l1,
157
+
.drv_name = "amd_sdw",
158
+
},
159
+
{
160
+
.link_mask = BIT(1),
161
+
.links = acp70_cs42l43_l1_cs35l56x4_l1,
162
+
.drv_name = "amd_sdw",
163
+
},
164
+
{
165
+
.link_mask = BIT(1),
166
+
.links = acp70_cs35l56x4_l1,
294
167
.drv_name = "amd_sdw",
295
168
},
296
169
{},
+1
sound/soc/codecs/cs-amp-lib-test.c
+1
sound/soc/codecs/cs-amp-lib-test.c
+1
-1
sound/soc/codecs/cs530x.c
+1
-1
sound/soc/codecs/cs530x.c
+4
-2
sound/soc/codecs/max98090.c
+4
-2
sound/soc/codecs/max98090.c
···
1239
1239
SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE,
1240
1240
M98090_DIGMIC4_SHIFT, 0, max98090_shdn_event,
1241
1241
SND_SOC_DAPM_POST_PMU),
1242
+
SND_SOC_DAPM_SUPPLY("DMIC34_HPF", M98090_REG_FILTER_CONFIG,
1243
+
M98090_FLT_DMIC34HPF_SHIFT, 0, NULL, 0),
1242
1244
};
1243
1245
1244
1246
static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
···
1429
1427
/* DMIC inputs */
1430
1428
{"DMIC3", NULL, "DMIC3_ENA"},
1431
1429
{"DMIC4", NULL, "DMIC4_ENA"},
1432
-
{"DMIC3", NULL, "AHPF"},
1433
-
{"DMIC4", NULL, "AHPF"},
1430
+
{"DMIC3", NULL, "DMIC34_HPF"},
1431
+
{"DMIC4", NULL, "DMIC34_HPF"},
1434
1432
};
1435
1433
1436
1434
static int max98090_add_widgets(struct snd_soc_component *component)
+4
sound/soc/codecs/rt721-sdca.c
+4
sound/soc/codecs/rt721-sdca.c
···
281
281
rt_sdca_index_write(rt721->mbq_regmap, RT721_BOOST_CTRL,
282
282
RT721_BST_4CH_TOP_GATING_CTRL1, 0x002a);
283
283
regmap_write(rt721->regmap, 0x2f58, 0x07);
284
+
285
+
regmap_write(rt721->regmap, 0x2f51, 0x00);
286
+
rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT,
287
+
RT721_MISC_CTL, 0x0004);
284
288
}
285
289
286
290
static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721)
+1
sound/soc/codecs/rt721-sdca.h
+1
sound/soc/codecs/rt721-sdca.h
···
137
137
#define RT721_HDA_LEGACY_UAJ_CTL 0x02
138
138
#define RT721_HDA_LEGACY_CTL1 0x05
139
139
#define RT721_HDA_LEGACY_RESET_CTL 0x06
140
+
#define RT721_MISC_CTL 0x07
140
141
#define RT721_XU_REL_CTRL 0x0c
141
142
#define RT721_GE_REL_CTRL1 0x0d
142
143
#define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e
+2
-2
sound/soc/fsl/fsl_micfil.c
+2
-2
sound/soc/fsl/fsl_micfil.c
···
131
131
.fifos = 8,
132
132
.fifo_depth = 32,
133
133
.dataline = 0xf,
134
-
.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE,
134
+
.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE,
135
135
.use_edma = true,
136
136
.use_verid = true,
137
137
.volume_sx = false,
···
823
823
break;
824
824
}
825
825
826
-
if (format == SNDRV_PCM_FORMAT_DSD_U32_BE) {
826
+
if (format == SNDRV_PCM_FORMAT_DSD_U32_LE) {
827
827
micfil->dec_bypass = true;
828
828
/*
829
829
* According to equation 29 in RM:
+5
-6
sound/soc/fsl/fsl_sai.c
+5
-6
sound/soc/fsl/fsl_sai.c
···
353
353
break;
354
354
case SND_SOC_DAIFMT_PDM:
355
355
val_cr2 |= FSL_SAI_CR2_BCP;
356
-
val_cr4 &= ~FSL_SAI_CR4_MF;
357
356
sai->is_pdm_mode = true;
358
357
break;
359
358
case SND_SOC_DAIFMT_RIGHT_J:
···
637
638
val_cr5 |= FSL_SAI_CR5_WNW(slot_width);
638
639
val_cr5 |= FSL_SAI_CR5_W0W(slot_width);
639
640
640
-
if (sai->is_lsb_first || sai->is_pdm_mode)
641
+
if (sai->is_lsb_first)
641
642
val_cr5 |= FSL_SAI_CR5_FBT(0);
642
643
else
643
644
val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
···
652
653
val_cr4 |= FSL_SAI_CR4_CHMOD;
653
654
654
655
/*
655
-
* For SAI provider mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will
656
-
* generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4),
657
-
* RCR5(TCR5) for playback(capture), or there will be sync error.
656
+
* When Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will provide bclk and
657
+
* frame clock for Tx(Rx). We should set RCR4(TCR4), RCR5(TCR5)
658
+
* for playback(capture), or there will be sync error.
658
659
*/
659
660
660
-
if (!sai->is_consumer_mode[tx] && fsl_sai_dir_is_synced(sai, adir)) {
661
+
if (fsl_sai_dir_is_synced(sai, adir)) {
661
662
regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs),
662
663
FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK |
663
664
FSL_SAI_CR4_CHMOD_MASK,
+3
sound/soc/intel/avs/pcm.c
+3
sound/soc/intel/avs/pcm.c
···
651
651
652
652
data = snd_soc_dai_get_dma_data(dai, substream);
653
653
654
+
disable_work_sync(&data->period_elapsed_work);
654
655
snd_hdac_ext_stream_release(data->host_stream, HDAC_EXT_STREAM_TYPE_HOST);
655
656
avs_dai_shutdown(substream, dai);
656
657
}
···
755
754
data = snd_soc_dai_get_dma_data(dai, substream);
756
755
host_stream = data->host_stream;
757
756
757
+
if (runtime->state == SNDRV_PCM_STATE_XRUN)
758
+
hdac_stream(host_stream)->prepared = false;
758
759
if (hdac_stream(host_stream)->prepared)
759
760
return 0;
760
761
+10
-8
sound/soc/intel/avs/probes.c
+10
-8
sound/soc/intel/avs/probes.c
···
14
14
#include "debug.h"
15
15
#include "messages.h"
16
16
17
-
static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id node_id,
18
-
size_t buffer_size)
17
+
static int avs_dsp_init_probe(struct avs_dev *adev, struct snd_compr_params *params, int bps,
18
+
union avs_connector_node_id node_id, size_t buffer_size)
19
19
{
20
20
struct avs_probe_cfg cfg = {{0}};
21
21
struct avs_module_entry mentry;
···
27
27
return ret;
28
28
29
29
/*
30
-
* Probe module uses no cycles, audio data format and input and output
31
-
* frame sizes are unused. It is also not owned by any pipeline.
30
+
* Probe module uses no cycles, input and output frame sizes are unused.
31
+
* It is also not owned by any pipeline.
32
32
*/
33
33
cfg.base.ibs = 1;
34
34
/* BSS module descriptor is always segment of index=2. */
35
35
cfg.base.is_pages = mentry.segments[2].flags.length;
36
+
cfg.base.audio_fmt.sampling_freq = params->codec.sample_rate;
37
+
cfg.base.audio_fmt.bit_depth = bps;
38
+
cfg.base.audio_fmt.num_channels = params->codec.ch_out;
39
+
cfg.base.audio_fmt.valid_bit_depth = bps;
36
40
cfg.gtw_cfg.node_id = node_id;
37
41
cfg.gtw_cfg.dma_buffer_size = buffer_size;
38
42
···
132
128
struct hdac_ext_stream *host_stream = avs_compr_get_host_stream(cstream);
133
129
struct snd_compr_runtime *rtd = cstream->runtime;
134
130
struct avs_dev *adev = to_avs_dev(dai->dev);
135
-
/* compr params do not store bit depth, default to S32_LE. */
136
-
snd_pcm_format_t format = SNDRV_PCM_FORMAT_S32_LE;
137
131
unsigned int format_val;
138
132
int bps, ret;
139
133
···
144
142
ret = snd_compr_malloc_pages(cstream, rtd->buffer_size);
145
143
if (ret < 0)
146
144
return ret;
147
-
bps = snd_pcm_format_physical_width(format);
145
+
bps = snd_pcm_format_physical_width(params->codec.format);
148
146
if (bps < 0)
149
147
return bps;
150
148
format_val = snd_hdac_stream_format(params->codec.ch_out, bps, params->codec.sample_rate);
···
168
166
node_id.vindex = hdac_stream(host_stream)->stream_tag - 1;
169
167
node_id.dma_type = AVS_DMA_HDA_HOST_INPUT;
170
168
171
-
ret = avs_dsp_init_probe(adev, node_id, rtd->dma_bytes);
169
+
ret = avs_dsp_init_probe(adev, params, bps, node_id, rtd->dma_bytes);
172
170
if (ret < 0) {
173
171
dev_err(dai->dev, "probe init failed: %d\n", ret);
174
172
avs_dsp_enable_d0ix(adev);
-52
sound/soc/intel/common/soc-acpi-intel-ptl-match.c
-52
sound/soc/intel/common/soc-acpi-intel-ptl-match.c
···
227
227
},
228
228
};
229
229
230
-
static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
231
-
{ /* Jack Playback Endpoint */
232
-
.num = 0,
233
-
.aggregated = 0,
234
-
.group_position = 0,
235
-
.group_id = 0,
236
-
},
237
-
{ /* DMIC Capture Endpoint */
238
-
.num = 1,
239
-
.aggregated = 0,
240
-
.group_position = 0,
241
-
.group_id = 0,
242
-
},
243
-
{ /* Jack Capture Endpoint */
244
-
.num = 2,
245
-
.aggregated = 0,
246
-
.group_position = 0,
247
-
.group_id = 0,
248
-
},
249
-
{ /* Speaker Playback Endpoint */
250
-
.num = 3,
251
-
.aggregated = 0,
252
-
.group_position = 0,
253
-
.group_id = 0,
254
-
},
255
-
};
256
-
257
230
static const struct snd_soc_acpi_adr_device cs42l43_2_adr[] = {
258
231
{
259
232
.adr = 0x00023001fa424301ull,
···
275
302
.num_endpoints = 1,
276
303
.endpoints = &spk_6_endpoint,
277
304
.name_prefix = "AMP6"
278
-
}
279
-
};
280
-
281
-
static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = {
282
-
{
283
-
.adr = 0x00033001FA424301ull,
284
-
.num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
285
-
.endpoints = cs42l43_endpoints,
286
-
.name_prefix = "cs42l43"
287
305
}
288
306
};
289
307
···
446
482
.mask = BIT(3),
447
483
.num_adr = ARRAY_SIZE(cs35l56_3_3amp_adr),
448
484
.adr_d = cs35l56_3_3amp_adr,
449
-
},
450
-
{}
451
-
};
452
-
453
-
static const struct snd_soc_acpi_link_adr ptl_cs42l43_l3[] = {
454
-
{
455
-
.mask = BIT(3),
456
-
.num_adr = ARRAY_SIZE(cs42l43_3_adr),
457
-
.adr_d = cs42l43_3_adr,
458
485
},
459
486
{}
460
487
};
···
665
710
.links = ptl_rt722_l1,
666
711
.drv_name = "sof_sdw",
667
712
.sof_tplg_filename = "sof-ptl-rt722.tplg",
668
-
.get_function_tplg_files = sof_sdw_get_tplg_files,
669
-
},
670
-
{
671
-
.link_mask = BIT(3),
672
-
.links = ptl_cs42l43_l3,
673
-
.drv_name = "sof_sdw",
674
-
.sof_tplg_filename = "sof-ptl-cs42l43-l3.tplg",
675
713
.get_function_tplg_files = sof_sdw_get_tplg_files,
676
714
},
677
715
{
-1
sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
-1
sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
-1
sound/soc/mediatek/mt8365/mt8365-afe-pcm.c
-1
sound/soc/mediatek/mt8365/mt8365-afe-pcm.c
+1
-1
sound/soc/qcom/qdsp6/q6asm.c
+1
-1
sound/soc/qcom/qdsp6/q6asm.c
+12
-13
sound/soc/renesas/rz-ssi.c
+12
-13
sound/soc/renesas/rz-ssi.c
···
85
85
struct snd_pcm_substream *substream;
86
86
int fifo_sample_size; /* sample capacity of SSI FIFO */
87
87
int dma_buffer_pos; /* The address for the next DMA descriptor */
88
+
int completed_dma_buf_pos; /* The address of the last completed DMA descriptor. */
88
89
int period_counter; /* for keeping track of periods transferred */
89
90
int sample_width;
90
91
int buffer_pos; /* current frame position in the buffer */
···
216
215
rz_ssi_set_substream(strm, substream);
217
216
strm->sample_width = samples_to_bytes(runtime, 1);
218
217
strm->dma_buffer_pos = 0;
218
+
strm->completed_dma_buf_pos = 0;
219
219
strm->period_counter = 0;
220
220
strm->buffer_pos = 0;
221
221
···
439
437
snd_pcm_period_elapsed(strm->substream);
440
438
strm->period_counter = current_period;
441
439
}
440
+
441
+
strm->completed_dma_buf_pos += runtime->period_size;
442
+
if (strm->completed_dma_buf_pos >= runtime->buffer_size)
443
+
strm->completed_dma_buf_pos = 0;
442
444
}
443
445
444
446
static int rz_ssi_pio_recv(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm)
···
784
778
return -ENODEV;
785
779
}
786
780
787
-
static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi)
781
+
static int rz_ssi_trigger_resume(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm)
788
782
{
783
+
struct snd_pcm_substream *substream = strm->substream;
784
+
struct snd_pcm_runtime *runtime = substream->runtime;
789
785
int ret;
786
+
787
+
strm->dma_buffer_pos = strm->completed_dma_buf_pos + runtime->period_size;
790
788
791
789
if (rz_ssi_is_stream_running(&ssi->playback) ||
792
790
rz_ssi_is_stream_running(&ssi->capture))
···
804
794
ssi->hw_params_cache.channels);
805
795
}
806
796
807
-
static void rz_ssi_streams_suspend(struct rz_ssi_priv *ssi)
808
-
{
809
-
if (rz_ssi_is_stream_running(&ssi->playback) ||
810
-
rz_ssi_is_stream_running(&ssi->capture))
811
-
return;
812
-
813
-
ssi->playback.dma_buffer_pos = 0;
814
-
ssi->capture.dma_buffer_pos = 0;
815
-
}
816
-
817
797
static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
818
798
struct snd_soc_dai *dai)
819
799
{
···
813
813
814
814
switch (cmd) {
815
815
case SNDRV_PCM_TRIGGER_RESUME:
816
-
ret = rz_ssi_trigger_resume(ssi);
816
+
ret = rz_ssi_trigger_resume(ssi, strm);
817
817
if (ret)
818
818
return ret;
819
819
···
852
852
853
853
case SNDRV_PCM_TRIGGER_SUSPEND:
854
854
rz_ssi_stop(ssi, strm);
855
-
rz_ssi_streams_suspend(ssi);
856
855
break;
857
856
858
857
case SNDRV_PCM_TRIGGER_STOP:
-1
sound/soc/sdw_utils/soc_sdw_utils.c
-1
sound/soc/sdw_utils/soc_sdw_utils.c
+20
-17
sound/usb/mixer_s1810c.c
+20
-17
sound/usb/mixer_s1810c.c
···
178
178
179
179
pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1;
180
180
pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2;
181
-
ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
181
+
ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
182
182
SC1810C_SET_STATE_REQ,
183
183
SC1810C_SET_STATE_REQTYPE,
184
184
(*seqnum), 0, &pkt_out, sizeof(pkt_out));
···
597
597
if (!list_empty(&chip->mixer_list))
598
598
return 0;
599
599
600
-
dev_info(&dev->dev,
601
-
"Presonus Studio 1810c, device_setup: %u\n", chip->setup);
602
-
if (chip->setup == 1)
603
-
dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
604
-
else if (chip->setup == 2)
605
-
dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
606
-
else
607
-
dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
608
-
609
600
ret = snd_s1810c_init_mixer_maps(chip);
610
601
if (ret < 0)
611
602
return ret;
···
625
634
if (ret < 0)
626
635
return ret;
627
636
628
-
// The 1824c has a Mono Main switch instead of a
629
-
// A/B select switch.
630
-
if (mixer->chip->usb_id == USB_ID(0x194f, 0x010d)) {
631
-
ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw);
632
-
if (ret < 0)
633
-
return ret;
634
-
} else if (mixer->chip->usb_id == USB_ID(0x194f, 0x010c)) {
637
+
switch (chip->usb_id) {
638
+
case USB_ID(0x194f, 0x010c): /* Presonus Studio 1810c */
639
+
dev_info(&dev->dev,
640
+
"Presonus Studio 1810c, device_setup: %u\n", chip->setup);
641
+
if (chip->setup == 1)
642
+
dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
643
+
else if (chip->setup == 2)
644
+
dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
645
+
else
646
+
dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
647
+
635
648
ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
636
649
if (ret < 0)
637
650
return ret;
651
+
652
+
break;
653
+
case USB_ID(0x194f, 0x010d): /* Presonus Studio 1824c */
654
+
ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw);
655
+
if (ret < 0)
656
+
return ret;
657
+
658
+
break;
638
659
}
639
660
640
661
return ret;