Linux kernel mirror (for testing)
git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
kernel
os
linux
Merge tag 'sound-6.14-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of fixes. The only slightly large change is for ASoC
Cirrus codec, but that's still in a normal range. All the rest are
small device-specific fixes and should be fairly safe to take"
* tag 'sound-6.14-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: Fix microphone regression on ASUS N705UD
ALSA: hda/realtek: Fix wrong mic setup for ASUS VivoBook 15
ASoC: cs35l56: Prevent races when soft-resetting using SPI control
firmware: cs_dsp: Remove async regmap writes
ASoC: Intel: sof_sdw: warn both sdw and pch dmic are used
ASoC: SOF: Intel: don't check number of sdw links when set dmic_fixup
ASoC: dapm-graph: set fill colour of turned on nodes
ASoC: fsl: Rename stream name of SAI DAI driver
ASoC: es8328: fix route from DAC to output
ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-900NXS2
ASoC: tas2764: Set the SDOUT polarity correctly
ASoC: tas2764: Fix power control mask
ALSA: usb-audio: Avoid dropping MIDI events at closing multiple ports
ASoC: tas2770: Fix volume scale
+162
-56
+6
-18
drivers/firmware/cirrus/cs_dsp.c
+6
-18
drivers/firmware/cirrus/cs_dsp.c
···
1609
1609
goto out_fw;
1610
1610
}
1611
1611
1612
-
ret = regmap_raw_write_async(regmap, reg, buf->buf,
1613
-
le32_to_cpu(region->len));
1612
+
ret = regmap_raw_write(regmap, reg, buf->buf,
1613
+
le32_to_cpu(region->len));
1614
1614
if (ret != 0) {
1615
1615
cs_dsp_err(dsp,
1616
1616
"%s.%d: Failed to write %d bytes at %d in %s: %d\n",
···
1625
1625
regions++;
1626
1626
}
1627
1627
1628
-
ret = regmap_async_complete(regmap);
1629
-
if (ret != 0) {
1630
-
cs_dsp_err(dsp, "Failed to complete async write: %d\n", ret);
1631
-
goto out_fw;
1632
-
}
1633
-
1634
1628
if (pos > firmware->size)
1635
1629
cs_dsp_warn(dsp, "%s.%d: %zu bytes at end of file\n",
1636
1630
file, regions, pos - firmware->size);
···
1632
1638
cs_dsp_debugfs_save_wmfwname(dsp, file);
1633
1639
1634
1640
out_fw:
1635
-
regmap_async_complete(regmap);
1636
1641
cs_dsp_buf_free(&buf_list);
1637
1642
1638
1643
if (ret == -EOVERFLOW)
···
2319
2326
cs_dsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
2320
2327
file, blocks, le32_to_cpu(blk->len),
2321
2328
reg);
2322
-
ret = regmap_raw_write_async(regmap, reg, buf->buf,
2323
-
le32_to_cpu(blk->len));
2329
+
ret = regmap_raw_write(regmap, reg, buf->buf,
2330
+
le32_to_cpu(blk->len));
2324
2331
if (ret != 0) {
2325
2332
cs_dsp_err(dsp,
2326
2333
"%s.%d: Failed to write to %x in %s: %d\n",
···
2332
2339
blocks++;
2333
2340
}
2334
2341
2335
-
ret = regmap_async_complete(regmap);
2336
-
if (ret != 0)
2337
-
cs_dsp_err(dsp, "Failed to complete async write: %d\n", ret);
2338
-
2339
2342
if (pos > firmware->size)
2340
2343
cs_dsp_warn(dsp, "%s.%d: %zu bytes at end of file\n",
2341
2344
file, blocks, pos - firmware->size);
···
2339
2350
cs_dsp_debugfs_save_binname(dsp, file);
2340
2351
2341
2352
out_fw:
2342
-
regmap_async_complete(regmap);
2343
2353
cs_dsp_buf_free(&buf_list);
2344
2354
2345
2355
if (ret == -EOVERFLOW)
···
2549
2561
{
2550
2562
int ret;
2551
2563
2552
-
ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL,
2553
-
ADSP2_SYS_ENA, ADSP2_SYS_ENA);
2564
+
ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
2565
+
ADSP2_SYS_ENA, ADSP2_SYS_ENA);
2554
2566
if (ret != 0)
2555
2567
return ret;
2556
2568
+31
include/sound/cs35l56.h
+31
include/sound/cs35l56.h
···
12
12
#include <linux/firmware/cirrus/cs_dsp.h>
13
13
#include <linux/regulator/consumer.h>
14
14
#include <linux/regmap.h>
15
+
#include <linux/spi/spi.h>
15
16
#include <sound/cs-amp-lib.h>
16
17
17
18
#define CS35L56_DEVID 0x0000000
···
62
61
#define CS35L56_IRQ1_MASK_8 0x000E0AC
63
62
#define CS35L56_IRQ1_MASK_18 0x000E0D4
64
63
#define CS35L56_IRQ1_MASK_20 0x000E0DC
64
+
#define CS35L56_DSP_MBOX_1_RAW 0x0011000
65
65
#define CS35L56_DSP_VIRTUAL1_MBOX_1 0x0011020
66
66
#define CS35L56_DSP_VIRTUAL1_MBOX_2 0x0011024
67
67
#define CS35L56_DSP_VIRTUAL1_MBOX_3 0x0011028
···
226
224
#define CS35L56_HALO_STATE_SHUTDOWN 1
227
225
#define CS35L56_HALO_STATE_BOOT_DONE 2
228
226
227
+
#define CS35L56_MBOX_CMD_PING 0x0A000000
229
228
#define CS35L56_MBOX_CMD_AUDIO_PLAY 0x0B000001
230
229
#define CS35L56_MBOX_CMD_AUDIO_PAUSE 0x0B000002
231
230
#define CS35L56_MBOX_CMD_AUDIO_REINIT 0x0B000003
···
257
254
#define CS35L56_NUM_BULK_SUPPLIES 3
258
255
#define CS35L56_NUM_DSP_REGIONS 5
259
256
257
+
/* Additional margin for SYSTEM_RESET to control port ready on SPI */
258
+
#define CS35L56_SPI_RESET_TO_PORT_READY_US (CS35L56_CONTROL_PORT_READY_US + 2500)
259
+
260
+
struct cs35l56_spi_payload {
261
+
__be32 addr;
262
+
__be16 pad;
263
+
__be32 value;
264
+
} __packed;
265
+
static_assert(sizeof(struct cs35l56_spi_payload) == 10);
266
+
260
267
struct cs35l56_base {
261
268
struct device *dev;
262
269
struct regmap *regmap;
···
282
269
s8 cal_index;
283
270
struct cirrus_amp_cal_data cal_data;
284
271
struct gpio_desc *reset_gpio;
272
+
struct cs35l56_spi_payload *spi_payload_buf;
285
273
};
286
274
287
275
static inline bool cs35l56_is_otp_register(unsigned int reg)
288
276
{
289
277
return (reg >> 16) == 3;
278
+
}
279
+
280
+
static inline int cs35l56_init_config_for_spi(struct cs35l56_base *cs35l56,
281
+
struct spi_device *spi)
282
+
{
283
+
cs35l56->spi_payload_buf = devm_kzalloc(&spi->dev,
284
+
sizeof(*cs35l56->spi_payload_buf),
285
+
GFP_KERNEL | GFP_DMA);
286
+
if (!cs35l56->spi_payload_buf)
287
+
return -ENOMEM;
288
+
289
+
return 0;
290
+
}
291
+
292
+
static inline bool cs35l56_is_spi(struct cs35l56_base *cs35l56)
293
+
{
294
+
return IS_ENABLED(CONFIG_SPI_MASTER) && !!cs35l56->spi_payload_buf;
290
295
}
291
296
292
297
extern const struct regmap_config cs35l56_regmap_i2c;
+3
sound/pci/hda/cs35l56_hda_spi.c
+3
sound/pci/hda/cs35l56_hda_spi.c
+1
-1
sound/pci/hda/patch_realtek.c
+1
-1
sound/pci/hda/patch_realtek.c
···
10623
10623
SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC),
10624
10624
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
10625
10625
SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC),
10626
+
SND_PCI_QUIRK(0x1043, 0x1460, "Asus VivoBook 15", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
10626
10627
SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC),
10627
10628
SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC),
10628
10629
SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC),
···
10657
10656
SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
10658
10657
SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE),
10659
10658
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
10660
-
SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
10661
10659
SND_PCI_QUIRK(0x1043, 0x1a63, "ASUS UX3405MA", ALC245_FIXUP_CS35L41_SPI_2),
10662
10660
SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2),
10663
10661
SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
+3
sound/soc/codecs/cs35l56-spi.c
+3
sound/soc/codecs/cs35l56-spi.c
+4
-11
sound/soc/codecs/es8328.c
+4
-11
sound/soc/codecs/es8328.c
···
233
233
234
234
/* Left Mixer */
235
235
static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
236
-
SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0),
237
236
SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0),
238
237
SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0),
239
238
SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0),
···
242
243
static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
243
244
SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0),
244
245
SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0),
245
-
SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0),
246
246
SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0),
247
247
};
248
248
···
334
336
SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER,
335
337
ES8328_DACPOWER_LDAC_OFF, 1),
336
338
337
-
SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
339
+
SND_SOC_DAPM_MIXER("Left Mixer", ES8328_DACCONTROL17, 7, 0,
338
340
&es8328_left_mixer_controls[0],
339
341
ARRAY_SIZE(es8328_left_mixer_controls)),
340
-
SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
342
+
SND_SOC_DAPM_MIXER("Right Mixer", ES8328_DACCONTROL20, 7, 0,
341
343
&es8328_right_mixer_controls[0],
342
344
ARRAY_SIZE(es8328_right_mixer_controls)),
343
345
···
416
418
{ "Right Line Mux", "PGA", "Right PGA Mux" },
417
419
{ "Right Line Mux", "Differential", "Differential Mux" },
418
420
419
-
{ "Left Out 1", NULL, "Left DAC" },
420
-
{ "Right Out 1", NULL, "Right DAC" },
421
-
{ "Left Out 2", NULL, "Left DAC" },
422
-
{ "Right Out 2", NULL, "Right DAC" },
423
-
424
-
{ "Left Mixer", "Playback Switch", "Left DAC" },
421
+
{ "Left Mixer", NULL, "Left DAC" },
425
422
{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
426
423
{ "Left Mixer", "Right Playback Switch", "Right DAC" },
427
424
{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
428
425
429
426
{ "Right Mixer", "Left Playback Switch", "Left DAC" },
430
427
{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
431
-
{ "Right Mixer", "Playback Switch", "Right DAC" },
428
+
{ "Right Mixer", NULL, "Right DAC" },
432
429
{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
433
430
434
431
{ "DAC DIG", NULL, "DAC STM" },
+9
-1
sound/soc/codecs/tas2764.c
+9
-1
sound/soc/codecs/tas2764.c
···
365
365
{
366
366
struct snd_soc_component *component = dai->component;
367
367
struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
368
-
u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
368
+
u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0, asi_cfg_4 = 0;
369
369
int ret;
370
370
371
371
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
···
374
374
fallthrough;
375
375
case SND_SOC_DAIFMT_NB_NF:
376
376
asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
377
+
asi_cfg_4 = TAS2764_TDM_CFG4_TX_FALLING;
377
378
break;
378
379
case SND_SOC_DAIFMT_IB_IF:
379
380
asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
380
381
fallthrough;
381
382
case SND_SOC_DAIFMT_IB_NF:
382
383
asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
384
+
asi_cfg_4 = TAS2764_TDM_CFG4_TX_RISING;
383
385
break;
384
386
}
385
387
386
388
ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
387
389
TAS2764_TDM_CFG1_RX_MASK,
388
390
asi_cfg_1);
391
+
if (ret < 0)
392
+
return ret;
393
+
394
+
ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG4,
395
+
TAS2764_TDM_CFG4_TX_MASK,
396
+
asi_cfg_4);
389
397
if (ret < 0)
390
398
return ret;
391
399
+7
-1
sound/soc/codecs/tas2764.h
+7
-1
sound/soc/codecs/tas2764.h
···
25
25
26
26
/* Power Control */
27
27
#define TAS2764_PWR_CTRL TAS2764_REG(0X0, 0x02)
28
-
#define TAS2764_PWR_CTRL_MASK GENMASK(1, 0)
28
+
#define TAS2764_PWR_CTRL_MASK GENMASK(2, 0)
29
29
#define TAS2764_PWR_CTRL_ACTIVE 0x0
30
30
#define TAS2764_PWR_CTRL_MUTE BIT(0)
31
31
#define TAS2764_PWR_CTRL_SHUTDOWN BIT(1)
···
78
78
#define TAS2764_TDM_CFG3_RXS_MASK GENMASK(7, 4)
79
79
#define TAS2764_TDM_CFG3_RXS_SHIFT 0x4
80
80
#define TAS2764_TDM_CFG3_MASK GENMASK(3, 0)
81
+
82
+
/* TDM Configuration Reg4 */
83
+
#define TAS2764_TDM_CFG4 TAS2764_REG(0X0, 0x0d)
84
+
#define TAS2764_TDM_CFG4_TX_MASK BIT(0)
85
+
#define TAS2764_TDM_CFG4_TX_RISING 0x0
86
+
#define TAS2764_TDM_CFG4_TX_FALLING BIT(0)
81
87
82
88
/* TDM Configuration Reg5 */
83
89
#define TAS2764_TDM_CFG5 TAS2764_REG(0X0, 0x0e)
+1
-1
sound/soc/codecs/tas2770.c
+1
-1
sound/soc/codecs/tas2770.c
···
506
506
}
507
507
508
508
static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0);
509
-
static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0);
509
+
static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -10050, 50, 0);
510
510
511
511
static const struct snd_kcontrol_new tas2770_snd_controls[] = {
512
512
SOC_SINGLE_TLV("Speaker Playback Volume", TAS2770_PLAY_CFG_REG2,
+3
-3
sound/soc/fsl/fsl_sai.c
+3
-3
sound/soc/fsl/fsl_sai.c
···
994
994
{
995
995
.name = "sai-tx",
996
996
.playback = {
997
-
.stream_name = "CPU-Playback",
997
+
.stream_name = "SAI-Playback",
998
998
.channels_min = 1,
999
999
.channels_max = 32,
1000
-
.rate_min = 8000,
1000
+
.rate_min = 8000,
1001
1001
.rate_max = 2822400,
1002
1002
.rates = SNDRV_PCM_RATE_KNOT,
1003
1003
.formats = FSL_SAI_FORMATS,
···
1007
1007
{
1008
1008
.name = "sai-rx",
1009
1009
.capture = {
1010
-
.stream_name = "CPU-Capture",
1010
+
.stream_name = "SAI-Capture",
1011
1011
.channels_min = 1,
1012
1012
.channels_max = 32,
1013
1013
.rate_min = 8000,
+2
-2
sound/soc/fsl/imx-audmix.c
+2
-2
sound/soc/fsl/imx-audmix.c
···
119
119
static const char *name[][3] = {
120
120
{"HiFi-AUDMIX-FE-0", "HiFi-AUDMIX-FE-1", "HiFi-AUDMIX-FE-2"},
121
121
{"sai-tx", "sai-tx", "sai-rx"},
122
-
{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "CPU-Capture"},
123
-
{"CPU-Playback", "CPU-Playback", "AUDMIX-Capture-0"},
122
+
{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "SAI-Capture"},
123
+
{"SAI-Playback", "SAI-Playback", "AUDMIX-Capture-0"},
124
124
};
125
125
126
126
static int imx_audmix_probe(struct platform_device *pdev)
+7
sound/soc/intel/boards/sof_sdw.c
+7
sound/soc/intel/boards/sof_sdw.c
···
803
803
int *be_id, struct snd_soc_codec_conf **codec_conf)
804
804
{
805
805
struct device *dev = card->dev;
806
+
struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev);
806
807
struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
808
+
struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
807
809
struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private;
808
810
struct asoc_sdw_endpoint *sof_end;
809
811
int stream;
···
902
900
903
901
codecs[j].name = sof_end->codec_name;
904
902
codecs[j].dai_name = sof_end->dai_info->dai_name;
903
+
if (sof_end->dai_info->dai_type == SOC_SDW_DAI_TYPE_MIC &&
904
+
mach_params->dmic_num > 0) {
905
+
dev_warn(dev,
906
+
"Both SDW DMIC and PCH DMIC are present, if incorrect, please set kernel params snd_sof_intel_hda_generic dmic_num=0 to disable PCH DMIC\n");
907
+
}
905
908
j++;
906
909
}
907
910
+2
-16
sound/soc/sof/intel/hda.c
+2
-16
sound/soc/sof/intel/hda.c
···
1312
1312
/* report to machine driver if any DMICs are found */
1313
1313
mach->mach_params.dmic_num = check_dmic_num(sdev);
1314
1314
1315
-
if (sdw_mach_found) {
1316
-
/*
1317
-
* DMICs use up to 4 pins and are typically pin-muxed with SoundWire
1318
-
* link 2 and 3, or link 1 and 2, thus we only try to enable dmics
1319
-
* if all conditions are true:
1320
-
* a) 2 or fewer links are used by SoundWire
1321
-
* b) the NHLT table reports the presence of microphones
1322
-
*/
1323
-
if (hweight_long(mach->link_mask) <= 2)
1324
-
dmic_fixup = true;
1325
-
else
1326
-
mach->mach_params.dmic_num = 0;
1327
-
} else {
1328
-
if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER)
1329
-
dmic_fixup = true;
1330
-
}
1315
+
if (sdw_mach_found || mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER)
1316
+
dmic_fixup = true;
1331
1317
1332
1318
if (tplg_fixup &&
1333
1319
dmic_fixup &&
+1
-1
sound/usb/midi.c
+1
-1
sound/usb/midi.c
+1
sound/usb/quirks.c
+1
sound/usb/quirks.c
···
1868
1868
case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
1869
1869
subs->stream_offset_adj = 2;
1870
1870
break;
1871
+
case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */
1871
1872
case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */
1872
1873
pioneer_djm_set_format_quirk(subs, 0x0082);
1873
1874
break;
+1
-1
tools/sound/dapm-graph
+1
-1
tools/sound/dapm-graph
···
10
10
11
11
STYLE_COMPONENT_ON="color=dodgerblue;style=bold"
12
12
STYLE_COMPONENT_OFF="color=gray40;style=filled;fillcolor=gray90"
13
-
STYLE_NODE_ON="shape=box,style=bold,color=green4"
13
+
STYLE_NODE_ON="shape=box,style=bold,color=green4,fillcolor=white"
14
14
STYLE_NODE_OFF="shape=box,style=filled,color=gray30,fillcolor=gray95"
15
15
16
16
# Print usage and exit