Linux kernel mirror (for testing)
git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
kernel
os
linux
Merge tag 'sound-7.0-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A bunch of small device-specific fixes. Mostly quirks and fix-ups for
USB- and HD-audio at this time, in addition to a couple of ASoC AMD
and Cirrus fixes"
* tag 'sound-7.0-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
ASoC: SDCA: Fix comments for sdca_irq_request()
ALSA: us144mkii: Drop kernel-doc markers
ALSA: usb: qcom: Correct parameter comment for uaudio_transfer_buffer_setup()
ALSA: usb-audio: Drop superfluous kernel-doc markers
ALSA: hda: cs35l56: Remove unnecessary struct cs_dsp_client_ops
ALSA: hda: cs35l56: Fix signedness error in cs35l56_hda_posture_put()
ALSA: usb-audio: Use correct version for UAC3 header validation
ALSA: hda/realtek: add quirk for Acer Nitro ANV15-51
ALSA: hda/intel: increase default bdl_pos_adj for Nvidia controllers
ALSA: usb-audio: Use inclusive terms
ALSA: usb-audio: Avoid implicit feedback mode on DIYINHK USB Audio 2.0
ALSA: usb-audio: Check max frame size for implicit feedback mode, too
ALSA: usb-audio: Cap the packet size pre-calculations
ASoC: amd: yc: Add ASUS EXPERTBOOK BM1503CDA to quirk table
ASoC: cs42l43: Report insert for exotic peripherals
ALSA: usb-audio: Skip clock selector for Focusrite devices
ALSA: usb-audio: Add QUIRK_FLAG_SKIP_IFACE_SETUP
ALSA: usb-audio: Remove VALIDATE_RATES quirk for Focusrite devices
ALSA: usb-audio: Improve Focusrite sample rate filtering
ALSA: hda/realtek: add quirk for Samsung Galaxy Book Flex (NT950QCT-A38A)
...
+196
-83
+36
-1
sound/hda/codecs/realtek/alc269.c
+36
-1
sound/hda/codecs/realtek/alc269.c
···
1017
1017
return 0;
1018
1018
}
1019
1019
1020
+
#define STARLABS_STARFIGHTER_SHUTUP_DELAY_MS 30
1021
+
1022
+
static void starlabs_starfighter_shutup(struct hda_codec *codec)
1023
+
{
1024
+
if (snd_hda_gen_shutup_speakers(codec))
1025
+
msleep(STARLABS_STARFIGHTER_SHUTUP_DELAY_MS);
1026
+
}
1027
+
1028
+
static void alc233_fixup_starlabs_starfighter(struct hda_codec *codec,
1029
+
const struct hda_fixup *fix,
1030
+
int action)
1031
+
{
1032
+
struct alc_spec *spec = codec->spec;
1033
+
1034
+
if (action == HDA_FIXUP_ACT_PRE_PROBE)
1035
+
spec->shutup = starlabs_starfighter_shutup;
1036
+
}
1037
+
1020
1038
static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec,
1021
1039
const struct hda_fixup *fix, int action)
1022
1040
{
···
4058
4040
ALC245_FIXUP_CLEVO_NOISY_MIC,
4059
4041
ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE,
4060
4042
ALC233_FIXUP_MEDION_MTL_SPK,
4043
+
ALC233_FIXUP_STARLABS_STARFIGHTER,
4061
4044
ALC294_FIXUP_BASS_SPEAKER_15,
4062
4045
ALC283_FIXUP_DELL_HP_RESUME,
4063
4046
ALC294_FIXUP_ASUS_CS35L41_SPI_2,
···
4075
4056
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO,
4076
4057
ALC233_FIXUP_LENOVO_GPIO2_MIC_HOTKEY,
4077
4058
ALC245_FIXUP_BASS_HP_DAC,
4059
+
ALC245_FIXUP_ACER_MICMUTE_LED,
4078
4060
};
4079
4061
4080
4062
/* A special fixup for Lenovo C940 and Yoga Duet 7;
···
6519
6499
{ }
6520
6500
},
6521
6501
},
6502
+
[ALC233_FIXUP_STARLABS_STARFIGHTER] = {
6503
+
.type = HDA_FIXUP_FUNC,
6504
+
.v.func = alc233_fixup_starlabs_starfighter,
6505
+
},
6522
6506
[ALC294_FIXUP_BASS_SPEAKER_15] = {
6523
6507
.type = HDA_FIXUP_FUNC,
6524
6508
.v.func = alc294_fixup_bass_speaker_15,
···
6600
6576
/* Borrow the DAC routing selected for those Thinkpads */
6601
6577
.v.func = alc285_fixup_thinkpad_x1_gen7,
6602
6578
},
6579
+
[ALC245_FIXUP_ACER_MICMUTE_LED] = {
6580
+
.type = HDA_FIXUP_FUNC,
6581
+
.v.func = alc285_fixup_hp_coef_micmute_led,
6582
+
.chained = true,
6583
+
.chain_id = ALC2XX_FIXUP_HEADSET_MIC,
6584
+
}
6603
6585
};
6604
6586
6605
6587
static const struct hda_quirk alc269_fixup_tbl[] = {
···
6621
6591
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
6622
6592
SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
6623
6593
SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF),
6594
+
SND_PCI_QUIRK(0x1025, 0x0943, "Acer Aspire V3-572G", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
6624
6595
SND_PCI_QUIRK(0x1025, 0x100c, "Acer Aspire E5-574G", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST),
6625
6596
SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK),
6626
6597
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
···
6658
6627
SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC2XX_FIXUP_HEADSET_MIC),
6659
6628
SND_PCI_QUIRK(0x1025, 0x1597, "Acer Nitro 5 AN517-55", ALC2XX_FIXUP_HEADSET_MIC),
6660
6629
SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED),
6630
+
SND_PCI_QUIRK(0x1025, 0x171e, "Acer Nitro ANV15-51", ALC245_FIXUP_ACER_MICMUTE_LED),
6661
6631
SND_PCI_QUIRK(0x1025, 0x1826, "Acer Helios ZPC", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2),
6662
6632
SND_PCI_QUIRK(0x1025, 0x182c, "Acer Helios ZPD", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2),
6663
6633
SND_PCI_QUIRK(0x1025, 0x1844, "Acer Helios ZPS", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2),
···
7343
7311
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
7344
7312
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP),
7345
7313
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP),
7346
-
SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP),
7314
+
SND_PCI_QUIRK(0x144d, 0xc188, "Samsung Galaxy Book Flex (NT950QCT-A38A)", ALC298_FIXUP_SAMSUNG_AMP),
7315
+
SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Book Flex (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP),
7347
7316
SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP),
7348
7317
SND_PCI_QUIRK(0x144d, 0xc1a3, "Samsung Galaxy Book Pro (NP935XDB-KC1SE)", ALC298_FIXUP_SAMSUNG_AMP),
7349
7318
SND_PCI_QUIRK(0x144d, 0xc1a4, "Samsung Galaxy Book Pro 360 (NT935QBD)", ALC298_FIXUP_SAMSUNG_AMP),
···
7684
7651
SND_PCI_QUIRK(0x2782, 0x1705, "MEDION E15433", ALC269VC_FIXUP_INFINIX_Y4_MAX),
7685
7652
SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME),
7686
7653
SND_PCI_QUIRK(0x2782, 0x4900, "MEDION E15443", ALC233_FIXUP_MEDION_MTL_SPK),
7654
+
SND_PCI_QUIRK(0x7017, 0x2014, "Star Labs StarFighter", ALC233_FIXUP_STARLABS_STARFIGHTER),
7687
7655
SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC),
7688
7656
SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
7689
7657
SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10),
···
7781
7747
{.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"},
7782
7748
{.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"},
7783
7749
{.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"},
7750
+
{.id = ALC233_FIXUP_STARLABS_STARFIGHTER, .name = "starlabs-starfighter"},
7784
7751
{.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"},
7785
7752
{.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"},
7786
7753
{.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"},
+1
-6
sound/hda/codecs/side-codecs/cs35l56_hda.c
+1
-6
sound/hda/codecs/side-codecs/cs35l56_hda.c
···
249
249
struct snd_ctl_elem_value *ucontrol)
250
250
{
251
251
struct cs35l56_hda *cs35l56 = snd_kcontrol_chip(kcontrol);
252
-
unsigned long pos = ucontrol->value.integer.value[0];
252
+
long pos = ucontrol->value.integer.value[0];
253
253
bool changed;
254
254
int ret;
255
255
···
402
402
snd_ctl_remove(cs35l56->codec->card, cs35l56->posture_ctl);
403
403
snd_ctl_remove(cs35l56->codec->card, cs35l56->volume_ctl);
404
404
}
405
-
406
-
static const struct cs_dsp_client_ops cs35l56_hda_client_ops = {
407
-
/* cs_dsp requires the client to provide this even if it is empty */
408
-
};
409
405
410
406
static int cs35l56_hda_request_firmware_file(struct cs35l56_hda *cs35l56,
411
407
const struct firmware **firmware, char **filename,
···
1145
1149
cs35l56->base.cal_index = cs35l56->index;
1146
1150
1147
1151
cs35l56_init_cs_dsp(&cs35l56->base, &cs35l56->cs_dsp);
1148
-
cs35l56->cs_dsp.client_ops = &cs35l56_hda_client_ops;
1149
1152
1150
1153
if (cs35l56->base.reset_gpio) {
1151
1154
dev_dbg(cs35l56->base.dev, "Hard reset\n");
+2
sound/hda/controllers/intel.c
+2
sound/hda/controllers/intel.c
+7
sound/soc/amd/yc/acp6x-mach.c
+7
sound/soc/amd/yc/acp6x-mach.c
···
703
703
DMI_MATCH(DMI_PRODUCT_NAME, "Vivobook_ASUSLaptop M6501RR_M6501RR"),
704
704
}
705
705
},
706
+
{
707
+
.driver_data = &acp6x_card,
708
+
.matches = {
709
+
DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
710
+
DMI_MATCH(DMI_PRODUCT_NAME, "ASUS EXPERTBOOK BM1503CDA"),
711
+
}
712
+
},
706
713
{}
707
714
};
708
715
+1
sound/soc/codecs/cs42l43-jack.c
+1
sound/soc/codecs/cs42l43-jack.c
+2
-2
sound/soc/sdca/sdca_interrupts.c
+2
-2
sound/soc/sdca/sdca_interrupts.c
···
265
265
}
266
266
267
267
/**
268
-
* sdca_request_irq - request an individual SDCA interrupt
268
+
* sdca_irq_request - request an individual SDCA interrupt
269
269
* @dev: Pointer to the struct device against which things should be allocated.
270
-
* @interrupt_info: Pointer to the interrupt information structure.
270
+
* @info: Pointer to the interrupt information structure.
271
271
* @sdca_irq: SDCA interrupt position.
272
272
* @name: Name to be given to the IRQ.
273
273
* @handler: A callback thread function to be called for the IRQ.
+7
-3
sound/usb/endpoint.c
+7
-3
sound/usb/endpoint.c
···
160
160
* This won't be used for implicit feedback which takes the packet size
161
161
* returned from the sync source
162
162
*/
163
-
static int slave_next_packet_size(struct snd_usb_endpoint *ep,
164
-
unsigned int avail)
163
+
static int synced_next_packet_size(struct snd_usb_endpoint *ep,
164
+
unsigned int avail)
165
165
{
166
166
unsigned int phase;
167
167
int ret;
···
221
221
222
222
packet = ctx->packet_size[idx];
223
223
if (packet) {
224
+
packet = min(packet, ep->maxframesize);
224
225
if (avail && packet >= avail)
225
226
return -EAGAIN;
226
227
return packet;
227
228
}
228
229
229
230
if (ep->sync_source)
230
-
return slave_next_packet_size(ep, avail);
231
+
return synced_next_packet_size(ep, avail);
231
232
else
232
233
return next_packet_size(ep, avail);
233
234
}
···
1378
1377
ep->maxpacksize, ep->cur_rate, ep->pps);
1379
1378
return -EINVAL;
1380
1379
}
1380
+
1381
+
ep->packsize[0] = min(ep->packsize[0], ep->maxframesize);
1382
+
ep->packsize[1] = min(ep->packsize[1], ep->maxframesize);
1381
1383
1382
1384
/* calculate the frequency in 16.16 format */
1383
1385
ep->freqm = ep->freqn;
+65
-5
sound/usb/format.c
+65
-5
sound/usb/format.c
···
305
305
}
306
306
307
307
/*
308
-
* Many Focusrite devices supports a limited set of sampling rates per
309
-
* altsetting. Maximum rate is exposed in the last 4 bytes of Format Type
310
-
* descriptor which has a non-standard bLength = 10.
308
+
* Focusrite devices use rate pairs: 44100/48000, 88200/96000, and
309
+
* 176400/192000. Return true if rate is in the pair for max_rate.
310
+
*/
311
+
static bool focusrite_rate_pair(unsigned int rate,
312
+
unsigned int max_rate)
313
+
{
314
+
switch (max_rate) {
315
+
case 48000: return rate == 44100 || rate == 48000;
316
+
case 96000: return rate == 88200 || rate == 96000;
317
+
case 192000: return rate == 176400 || rate == 192000;
318
+
default: return true;
319
+
}
320
+
}
321
+
322
+
/*
323
+
* Focusrite devices report all supported rates in a single clock
324
+
* source but only a subset is valid per altsetting.
325
+
*
326
+
* Detection uses two descriptor features:
327
+
*
328
+
* 1. Format Type descriptor bLength == 10: non-standard extension
329
+
* with max sample rate in bytes 6..9.
330
+
*
331
+
* 2. bmControls VAL_ALT_SETTINGS readable bit: when set, the device
332
+
* only supports the highest rate pair for that altsetting, and when
333
+
* clear, all rates up to max_rate are valid.
334
+
*
335
+
* For devices without the bLength == 10 extension but with
336
+
* VAL_ALT_SETTINGS readable and multiple altsettings (only seen in
337
+
* Scarlett 18i8 3rd Gen playback), fall back to the Focusrite
338
+
* convention: alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz.
311
339
*/
312
340
static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip,
313
341
struct audioformat *fp,
314
342
unsigned int rate)
315
343
{
344
+
struct usb_interface *iface;
316
345
struct usb_host_interface *alts;
346
+
struct uac2_as_header_descriptor *as;
317
347
unsigned char *fmt;
318
348
unsigned int max_rate;
349
+
bool val_alt;
319
350
320
351
alts = snd_usb_get_host_interface(chip, fp->iface, fp->altsetting);
321
352
if (!alts)
···
357
326
if (!fmt)
358
327
return true;
359
328
329
+
as = snd_usb_find_csint_desc(alts->extra, alts->extralen,
330
+
NULL, UAC_AS_GENERAL);
331
+
if (!as)
332
+
return true;
333
+
334
+
val_alt = uac_v2v3_control_is_readable(as->bmControls,
335
+
UAC2_AS_VAL_ALT_SETTINGS);
336
+
360
337
if (fmt[0] == 10) { /* bLength */
361
338
max_rate = combine_quad(&fmt[6]);
362
339
340
+
if (val_alt)
341
+
return focusrite_rate_pair(rate, max_rate);
342
+
343
+
/* No val_alt: rates fall through from higher */
363
344
switch (max_rate) {
364
345
case 192000:
365
346
if (rate == 176400 || rate == 192000)
···
387
344
usb_audio_info(chip,
388
345
"%u:%d : unexpected max rate: %u\n",
389
346
fp->iface, fp->altsetting, max_rate);
390
-
391
347
return true;
392
348
}
393
349
}
394
350
395
-
return true;
351
+
if (!val_alt)
352
+
return true;
353
+
354
+
/* Multi-altsetting device with val_alt but no max_rate
355
+
* in the format descriptor. Use Focusrite convention:
356
+
* alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz.
357
+
*/
358
+
iface = usb_ifnum_to_if(chip->dev, fp->iface);
359
+
if (!iface || iface->num_altsetting <= 2)
360
+
return true;
361
+
362
+
switch (fp->altsetting) {
363
+
case 1: max_rate = 48000; break;
364
+
case 2: max_rate = 96000; break;
365
+
case 3: max_rate = 192000; break;
366
+
default: return true;
367
+
}
368
+
369
+
return focusrite_rate_pair(rate, max_rate);
396
370
}
397
371
398
372
/*
+6
-6
sound/usb/mixer_s1810c.c
+6
-6
sound/usb/mixer_s1810c.c
···
71
71
* * e I guess the same as with mixer
72
72
*
73
73
*/
74
-
/** struct s1810c_ctl_packet - basic vendor request
74
+
/* struct s1810c_ctl_packet - basic vendor request
75
75
* @selector: device/mixer/output
76
76
* @b: request-dependant field b
77
77
* @tag: fixed value identifying type of request
···
94
94
__le32 e;
95
95
};
96
96
97
-
/** selectors for CMD request
97
+
/* selectors for CMD request
98
98
*/
99
99
#define SC1810C_SEL_DEVICE 0
100
100
#define SC1810C_SEL_MIXER 0x64
101
101
#define SC1810C_SEL_OUTPUT 0x65
102
102
103
103
104
-
/** control ids */
104
+
/* control ids */
105
105
#define SC1810C_CTL_LINE_SW 0
106
106
#define SC1810C_CTL_MUTE_SW 1
107
107
#define SC1824C_CTL_MONO_SW 2
···
127
127
#define SC1810C_GET_STATE_TAG SC1810C_SET_STATE_TAG
128
128
#define SC1810C_GET_STATE_LEN SC1810C_SET_STATE_LEN
129
129
130
-
/** Mixer levels normally range from 0 (off) to 0x0100 0000 (0 dB).
130
+
/* Mixer levels normally range from 0 (off) to 0x0100 0000 (0 dB).
131
131
* raw_level = 2^24 * 10^(db_level / 20), thus
132
132
* -3dB = 0xb53bf0 (technically, half-power -3.01...dB would be 0xb504f3)
133
133
* -96dB = 0x109
···
145
145
#define MIXER_LEVEL_N3DB 0xb53bf0
146
146
#define MIXER_LEVEL_0DB 0x1000000
147
147
148
-
/**
148
+
/*
149
149
* This packet includes mixer volumes and
150
150
* various other fields, it's an extended
151
151
* version of ctl_packet, with a and b
···
155
155
__le32 fields[63];
156
156
};
157
157
158
-
/** indices into s1810c_state_packet.fields[]
158
+
/* indices into s1810c_state_packet.fields[]
159
159
*/
160
160
#define SC1810C_STATE_TAG_IDX 2
161
161
#define SC1810C_STATE_LEN_IDX 3
+3
-7
sound/usb/mixer_scarlett2.c
+3
-7
sound/usb/mixer_scarlett2.c
···
1328
1328
struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX];
1329
1329
struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX];
1330
1330
struct snd_kcontrol *compressor_ctls[SCARLETT2_COMPRESSOR_CTLS_MAX];
1331
-
struct snd_kcontrol *precomp_flt_ctls[SCARLETT2_PRECOMP_FLT_CTLS_MAX];
1332
-
struct snd_kcontrol *peq_flt_ctls[SCARLETT2_PEQ_FLT_CTLS_MAX];
1333
1331
struct snd_kcontrol *precomp_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX];
1334
1332
struct snd_kcontrol *peq_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX];
1335
1333
struct snd_kcontrol *direct_monitor_ctl;
···
3445
3447
private->autogain_status[i] =
3446
3448
private->num_autogain_status_texts - 1;
3447
3449
3448
-
3449
3450
for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
3450
3451
if (scarlett2_has_config_item(private,
3451
3452
scarlett2_ag_target_configs[i])) {
···
5369
5372
5370
5373
err = scarlett2_usb_get_config(
5371
5374
mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH,
5372
-
info->dsp_input_count * info->peq_flt_count,
5373
-
private->peq_flt_switch);
5375
+
info->dsp_input_count, private->peq_flt_switch);
5374
5376
if (err < 0)
5375
5377
return err;
5376
5378
···
6542
6546
err = scarlett2_add_new_ctl(
6543
6547
mixer, &scarlett2_precomp_flt_ctl,
6544
6548
i * info->precomp_flt_count + j,
6545
-
1, s, &private->precomp_flt_switch_ctls[j]);
6549
+
1, s, NULL);
6546
6550
if (err < 0)
6547
6551
return err;
6548
6552
}
···
6552
6556
err = scarlett2_add_new_ctl(
6553
6557
mixer, &scarlett2_peq_flt_ctl,
6554
6558
i * info->peq_flt_count + j,
6555
-
1, s, &private->peq_flt_switch_ctls[j]);
6559
+
1, s, NULL);
6556
6560
if (err < 0)
6557
6561
return err;
6558
6562
}
+1
-1
sound/usb/qcom/qc_audio_offload.c
+1
-1
sound/usb/qcom/qc_audio_offload.c
···
1007
1007
/**
1008
1008
* uaudio_transfer_buffer_setup() - fetch and populate xfer buffer params
1009
1009
* @subs: usb substream
1010
-
* @xfer_buf: xfer buf to be allocated
1010
+
* @xfer_buf_cpu: xfer buf to be allocated
1011
1011
* @xfer_buf_len: size of allocation
1012
1012
* @mem_info: QMI response info
1013
1013
*
+5
-1
sound/usb/quirks.c
+5
-1
sound/usb/quirks.c
···
2365
2365
QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
2366
2366
DEVICE_FLG(0x2040, 0x7281, /* Hauppauge HVR-950Q-MXL */
2367
2367
QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
2368
+
DEVICE_FLG(0x20b1, 0x2009, /* XMOS Ltd DIYINHK USB Audio 2.0 */
2369
+
QUIRK_FLAG_SKIP_IMPLICIT_FB | QUIRK_FLAG_DSD_RAW),
2368
2370
DEVICE_FLG(0x2040, 0x8200, /* Hauppauge Woodbury */
2369
2371
QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
2370
2372
DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */
···
2426
2424
VENDOR_FLG(0x07fd, /* MOTU */
2427
2425
QUIRK_FLAG_VALIDATE_RATES),
2428
2426
VENDOR_FLG(0x1235, /* Focusrite Novation */
2429
-
QUIRK_FLAG_VALIDATE_RATES),
2427
+
QUIRK_FLAG_SKIP_CLOCK_SELECTOR |
2428
+
QUIRK_FLAG_SKIP_IFACE_SETUP),
2430
2429
VENDOR_FLG(0x1511, /* AURALiC */
2431
2430
QUIRK_FLAG_DSD_RAW),
2432
2431
VENDOR_FLG(0x152a, /* Thesycon devices */
···
2509
2506
QUIRK_STRING_ENTRY(MIC_RES_384),
2510
2507
QUIRK_STRING_ENTRY(MIXER_PLAYBACK_MIN_MUTE),
2511
2508
QUIRK_STRING_ENTRY(MIXER_CAPTURE_MIN_MUTE),
2509
+
QUIRK_STRING_ENTRY(SKIP_IFACE_SETUP),
2512
2510
NULL
2513
2511
};
2514
2512
+3
sound/usb/stream.c
+3
sound/usb/stream.c
···
1259
1259
set_iface_first = true;
1260
1260
1261
1261
/* try to set the interface... */
1262
+
if (chip->quirk_flags & QUIRK_FLAG_SKIP_IFACE_SETUP)
1263
+
continue;
1264
+
1262
1265
usb_set_interface(chip->dev, iface_no, 0);
1263
1266
if (set_iface_first)
1264
1267
usb_set_interface(chip->dev, iface_no, altno);
+6
sound/usb/usbaudio.h
+6
sound/usb/usbaudio.h
···
224
224
* playback value represents muted state instead of minimum audible volume
225
225
* QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE
226
226
* Similar to QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE, but for capture streams
227
+
* QUIRK_FLAG_SKIP_IFACE_SETUP
228
+
* Skip the probe-time interface setup (usb_set_interface,
229
+
* init_pitch, init_sample_rate); redundant with
230
+
* snd_usb_endpoint_prepare() at stream-open time
227
231
*/
228
232
229
233
enum {
···
257
253
QUIRK_TYPE_MIC_RES_384 = 23,
258
254
QUIRK_TYPE_MIXER_PLAYBACK_MIN_MUTE = 24,
259
255
QUIRK_TYPE_MIXER_CAPTURE_MIN_MUTE = 25,
256
+
QUIRK_TYPE_SKIP_IFACE_SETUP = 26,
260
257
/* Please also edit snd_usb_audio_quirk_flag_names */
261
258
};
262
259
···
289
284
#define QUIRK_FLAG_MIC_RES_384 QUIRK_FLAG(MIC_RES_384)
290
285
#define QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE QUIRK_FLAG(MIXER_PLAYBACK_MIN_MUTE)
291
286
#define QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE QUIRK_FLAG(MIXER_CAPTURE_MIN_MUTE)
287
+
#define QUIRK_FLAG_SKIP_IFACE_SETUP QUIRK_FLAG(SKIP_IFACE_SETUP)
292
288
293
289
#endif /* __USBAUDIO_H */
+7
-7
sound/usb/usx2y/us144mkii.c
+7
-7
sound/usb/usx2y/us144mkii.c
···
10
10
MODULE_DESCRIPTION("ALSA Driver for TASCAM US-144MKII");
11
11
MODULE_LICENSE("GPL");
12
12
13
-
/**
14
-
* @brief Module parameters for ALSA card instantiation.
13
+
/*
14
+
* Module parameters for ALSA card instantiation.
15
15
*
16
16
* These parameters allow users to configure how the ALSA sound card
17
17
* for the TASCAM US-144MKII is instantiated.
···
269
269
atomic_set(&tascam->active_urbs, 0);
270
270
}
271
271
272
-
/**
272
+
/*
273
273
* tascam_card_private_free() - Frees private data associated with the sound
274
274
* card.
275
275
* @card: Pointer to the ALSA sound card instance.
···
291
291
}
292
292
}
293
293
294
-
/**
294
+
/*
295
295
* tascam_suspend() - Handles device suspension.
296
296
* @intf: The USB interface being suspended.
297
297
* @message: Power management message.
···
332
332
return 0;
333
333
}
334
334
335
-
/**
335
+
/*
336
336
* tascam_resume() - Handles device resumption from suspend.
337
337
* @intf: The USB interface being resumed.
338
338
*
···
390
390
schedule_work(&tascam->midi_out_work);
391
391
}
392
392
393
-
/**
393
+
/*
394
394
* tascam_probe() - Probes for the TASCAM US-144MKII device.
395
395
* @intf: The USB interface being probed.
396
396
* @usb_id: The USB device ID.
···
565
565
return err;
566
566
}
567
567
568
-
/**
568
+
/*
569
569
* tascam_disconnect() - Disconnects the TASCAM US-144MKII device.
570
570
* @intf: The USB interface being disconnected.
571
571
*
+6
-6
sound/usb/usx2y/us144mkii_capture.c
+6
-6
sound/usb/usx2y/us144mkii_capture.c
···
3
3
4
4
#include "us144mkii.h"
5
5
6
-
/**
6
+
/*
7
7
* tascam_capture_open() - Opens the PCM capture substream.
8
8
* @substream: The ALSA PCM substream to open.
9
9
*
···
23
23
return 0;
24
24
}
25
25
26
-
/**
26
+
/*
27
27
* tascam_capture_close() - Closes the PCM capture substream.
28
28
* @substream: The ALSA PCM substream to close.
29
29
*
···
41
41
return 0;
42
42
}
43
43
44
-
/**
44
+
/*
45
45
* tascam_capture_prepare() - Prepares the PCM capture substream for use.
46
46
* @substream: The ALSA PCM substream to prepare.
47
47
*
···
62
62
return 0;
63
63
}
64
64
65
-
/**
65
+
/*
66
66
* tascam_capture_pointer() - Returns the current capture pointer position.
67
67
* @substream: The ALSA PCM substream.
68
68
*
···
91
91
return do_div(pos, runtime->buffer_size);
92
92
}
93
93
94
-
/**
94
+
/*
95
95
* tascam_capture_ops - ALSA PCM operations for capture.
96
96
*
97
97
* This structure defines the callback functions for capture stream operations,
···
109
109
.pointer = tascam_capture_pointer,
110
110
};
111
111
112
-
/**
112
+
/*
113
113
* decode_tascam_capture_block() - Decodes a raw 512-byte block from the device.
114
114
* @src_block: Pointer to the 512-byte raw source block.
115
115
* @dst_block: Pointer to the destination buffer for decoded audio frames.
+21
-21
sound/usb/usx2y/us144mkii_controls.c
+21
-21
sound/usb/usx2y/us144mkii_controls.c
···
3
3
4
4
#include "us144mkii.h"
5
5
6
-
/**
7
-
* @brief Text descriptions for playback output source options.
6
+
/*
7
+
* Text descriptions for playback output source options.
8
8
*
9
9
* Used by ALSA kcontrol elements to provide user-friendly names for
10
10
* the playback routing options (e.g., "Playback 1-2", "Playback 3-4").
···
12
12
static const char *const playback_source_texts[] = { "Playback 1-2",
13
13
"Playback 3-4" };
14
14
15
-
/**
16
-
* @brief Text descriptions for capture input source options.
15
+
/*
16
+
* Text descriptions for capture input source options.
17
17
*
18
18
* Used by ALSA kcontrol elements to provide user-friendly names for
19
19
* the capture routing options (e.g., "Analog In", "Digital In").
20
20
*/
21
21
static const char *const capture_source_texts[] = { "Analog In", "Digital In" };
22
22
23
-
/**
23
+
/*
24
24
* tascam_playback_source_info() - ALSA control info callback for playback
25
25
* source.
26
26
* @kcontrol: The ALSA kcontrol instance.
···
38
38
return snd_ctl_enum_info(uinfo, 1, 2, playback_source_texts);
39
39
}
40
40
41
-
/**
41
+
/*
42
42
* tascam_line_out_get() - ALSA control get callback for Line Outputs Source.
43
43
* @kcontrol: The ALSA kcontrol instance.
44
44
* @ucontrol: The ALSA control element value structure to fill.
···
60
60
return 0;
61
61
}
62
62
63
-
/**
63
+
/*
64
64
* tascam_line_out_put() - ALSA control put callback for Line Outputs Source.
65
65
* @kcontrol: The ALSA kcontrol instance.
66
66
* @ucontrol: The ALSA control element value structure containing the new value.
···
89
89
return changed;
90
90
}
91
91
92
-
/**
92
+
/*
93
93
* tascam_line_out_control - ALSA kcontrol definition for Line Outputs Source.
94
94
*
95
95
* This defines a new ALSA mixer control named "Line OUTPUTS Source" that allows
···
106
106
.put = tascam_line_out_put,
107
107
};
108
108
109
-
/**
109
+
/*
110
110
* tascam_digital_out_get() - ALSA control get callback for Digital Outputs
111
111
* Source.
112
112
* @kcontrol: The ALSA kcontrol instance.
···
129
129
return 0;
130
130
}
131
131
132
-
/**
132
+
/*
133
133
* tascam_digital_out_put() - ALSA control put callback for Digital Outputs
134
134
* Source.
135
135
* @kcontrol: The ALSA kcontrol instance.
···
159
159
return changed;
160
160
}
161
161
162
-
/**
162
+
/*
163
163
* tascam_digital_out_control - ALSA kcontrol definition for Digital Outputs
164
164
* Source.
165
165
*
···
177
177
.put = tascam_digital_out_put,
178
178
};
179
179
180
-
/**
180
+
/*
181
181
* tascam_capture_source_info() - ALSA control info callback for capture source.
182
182
* @kcontrol: The ALSA kcontrol instance.
183
183
* @uinfo: The ALSA control element info structure to fill.
···
194
194
return snd_ctl_enum_info(uinfo, 1, 2, capture_source_texts);
195
195
}
196
196
197
-
/**
197
+
/*
198
198
* tascam_capture_12_get() - ALSA control get callback for Capture channels 1
199
199
* and 2 Source.
200
200
* @kcontrol: The ALSA kcontrol instance.
···
217
217
return 0;
218
218
}
219
219
220
-
/**
220
+
/*
221
221
* tascam_capture_12_put() - ALSA control put callback for Capture channels 1
222
222
* and 2 Source.
223
223
* @kcontrol: The ALSA kcontrol instance.
···
247
247
return changed;
248
248
}
249
249
250
-
/**
250
+
/*
251
251
* tascam_capture_12_control - ALSA kcontrol definition for Capture channels 1
252
252
* and 2 Source.
253
253
*
···
265
265
.put = tascam_capture_12_put,
266
266
};
267
267
268
-
/**
268
+
/*
269
269
* tascam_capture_34_get() - ALSA control get callback for Capture channels 3
270
270
* and 4 Source.
271
271
* @kcontrol: The ALSA kcontrol instance.
···
288
288
return 0;
289
289
}
290
290
291
-
/**
291
+
/*
292
292
* tascam_capture_34_put() - ALSA control put callback for Capture channels 3
293
293
* and 4 Source.
294
294
* @kcontrol: The ALSA kcontrol instance.
···
318
318
return changed;
319
319
}
320
320
321
-
/**
321
+
/*
322
322
* tascam_capture_34_control - ALSA kcontrol definition for Capture channels 3
323
323
* and 4 Source.
324
324
*
···
336
336
.put = tascam_capture_34_put,
337
337
};
338
338
339
-
/**
339
+
/*
340
340
* tascam_samplerate_info() - ALSA control info callback for Sample Rate.
341
341
* @kcontrol: The ALSA kcontrol instance.
342
342
* @uinfo: The ALSA control element info structure to fill.
···
356
356
return 0;
357
357
}
358
358
359
-
/**
359
+
/*
360
360
* tascam_samplerate_get() - ALSA control get callback for Sample Rate.
361
361
* @kcontrol: The ALSA kcontrol instance.
362
362
* @ucontrol: The ALSA control element value structure to fill.
···
400
400
return 0;
401
401
}
402
402
403
-
/**
403
+
/*
404
404
* tascam_samplerate_control - ALSA kcontrol definition for Sample Rate.
405
405
*
406
406
* This defines a new ALSA mixer control named "Sample Rate" that displays
+11
-11
sound/usb/usx2y/us144mkii_midi.c
+11
-11
sound/usb/usx2y/us144mkii_midi.c
···
3
3
4
4
#include "us144mkii.h"
5
5
6
-
/**
6
+
/*
7
7
* tascam_midi_in_work_handler() - Deferred work for processing MIDI input.
8
8
* @work: The work_struct instance.
9
9
*
···
75
75
usb_put_urb(urb);
76
76
}
77
77
78
-
/**
78
+
/*
79
79
* tascam_midi_in_open() - Opens the MIDI input substream.
80
80
* @substream: The ALSA rawmidi substream to open.
81
81
*
···
92
92
return 0;
93
93
}
94
94
95
-
/**
95
+
/*
96
96
* tascam_midi_in_close() - Closes the MIDI input substream.
97
97
* @substream: The ALSA rawmidi substream to close.
98
98
*
···
103
103
return 0;
104
104
}
105
105
106
-
/**
106
+
/*
107
107
* tascam_midi_in_trigger() - Triggers MIDI input stream activity.
108
108
* @substream: The ALSA rawmidi substream.
109
109
* @up: Boolean indicating whether to start (1) or stop (0) the stream.
···
150
150
}
151
151
}
152
152
153
-
/**
153
+
/*
154
154
* tascam_midi_in_ops - ALSA rawmidi operations for MIDI input.
155
155
*
156
156
* This structure defines the callback functions for MIDI input stream
···
205
205
usb_put_urb(urb);
206
206
}
207
207
208
-
/**
208
+
/*
209
209
* tascam_midi_out_work_handler() - Deferred work for sending MIDI data
210
210
* @work: The work_struct instance.
211
211
*
···
282
282
}
283
283
}
284
284
285
-
/**
285
+
/*
286
286
* tascam_midi_out_open() - Opens the MIDI output substream.
287
287
* @substream: The ALSA rawmidi substream to open.
288
288
*
···
301
301
return 0;
302
302
}
303
303
304
-
/**
304
+
/*
305
305
* tascam_midi_out_close() - Closes the MIDI output substream.
306
306
* @substream: The ALSA rawmidi substream to close.
307
307
*
···
312
312
return 0;
313
313
}
314
314
315
-
/**
315
+
/*
316
316
* tascam_midi_out_drain() - Drains the MIDI output stream.
317
317
* @substream: The ALSA rawmidi substream.
318
318
*
···
340
340
usb_kill_anchored_urbs(&tascam->midi_out_anchor);
341
341
}
342
342
343
-
/**
343
+
/*
344
344
* tascam_midi_out_trigger() - Triggers MIDI output stream activity.
345
345
* @substream: The ALSA rawmidi substream.
346
346
* @up: Boolean indicating whether to start (1) or stop (0) the stream.
···
361
361
}
362
362
}
363
363
364
-
/**
364
+
/*
365
365
* tascam_midi_out_ops - ALSA rawmidi operations for MIDI output.
366
366
*
367
367
* This structure defines the callback functions for MIDI output stream
+5
-5
sound/usb/usx2y/us144mkii_playback.c
+5
-5
sound/usb/usx2y/us144mkii_playback.c
···
3
3
4
4
#include "us144mkii.h"
5
5
6
-
/**
6
+
/*
7
7
* tascam_playback_open() - Opens the PCM playback substream.
8
8
* @substream: The ALSA PCM substream to open.
9
9
*
···
23
23
return 0;
24
24
}
25
25
26
-
/**
26
+
/*
27
27
* tascam_playback_close() - Closes the PCM playback substream.
28
28
* @substream: The ALSA PCM substream to close.
29
29
*
···
41
41
return 0;
42
42
}
43
43
44
-
/**
44
+
/*
45
45
* tascam_playback_prepare() - Prepares the PCM playback substream for use.
46
46
* @substream: The ALSA PCM substream to prepare.
47
47
*
···
108
108
return 0;
109
109
}
110
110
111
-
/**
111
+
/*
112
112
* tascam_playback_pointer() - Returns the current playback pointer position.
113
113
* @substream: The ALSA PCM substream.
114
114
*
···
137
137
return do_div(pos, runtime->buffer_size);
138
138
}
139
139
140
-
/**
140
+
/*
141
141
* tascam_playback_ops - ALSA PCM operations for playback.
142
142
*
143
143
* This structure defines the callback functions for playback stream operations,
+1
-1
sound/usb/validate.c
+1
-1
sound/usb/validate.c
···
281
281
/* UAC_VERSION_2, UAC2_SAMPLE_RATE_CONVERTER: not implemented yet */
282
282
283
283
/* UAC3 */
284
-
FIXED(UAC_VERSION_2, UAC_HEADER, struct uac3_ac_header_descriptor),
284
+
FIXED(UAC_VERSION_3, UAC_HEADER, struct uac3_ac_header_descriptor),
285
285
FIXED(UAC_VERSION_3, UAC_INPUT_TERMINAL,
286
286
struct uac3_input_terminal_descriptor),
287
287
FIXED(UAC_VERSION_3, UAC_OUTPUT_TERMINAL,