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f7af6cceff805ed458278187af4f9aa4e212d008
omarchy-pkgs/pkgbuilds/linux-omarchy/0511-sound-updates-fixes.patch
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diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
--- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
+++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
@@ -237,34 +237,34 @@
/* Clock ID for RX CORE MCLK2 2X MCLK */
#define LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 70
-#define LAPSS_CLK_ID_QAIF_IF0_IBIT 71
-#define LAPSS_CLK_ID_QAIF_IF0_EBIT 72
-#define LAPSS_CLK_ID_QAIF_IF1_IBIT 73
-#define LAPSS_CLK_ID_QAIF_IF1_EBIT 74
-#define LAPSS_CLK_ID_QAIF_IF2_IBIT 75
-#define LAPSS_CLK_ID_QAIF_IF2_EBIT 76
-#define LAPSS_CLK_ID_QAIF_IF3_IBIT 77
-#define LAPSS_CLK_ID_QAIF_IF3_EBIT 78
-#define LAPSS_CLK_ID_QAIF_IF4_IBIT 79
-#define LAPSS_CLK_ID_QAIF_IF4_EBIT 80
-#define LAPSS_CLK_ID_QAIF_IF5_IBIT 81
-#define LAPSS_CLK_ID_QAIF_IF5_EBIT 82
-#define LAPSS_CLK_ID_QAIF_IF6_IBIT 83
-#define LAPSS_CLK_ID_QAIF_IF6_EBIT 84
-#define LAPSS_CLK_ID_QAIF_IF7_IBIT 85
-#define LAPSS_CLK_ID_QAIF_IF7_EBIT 86
-#define LAPSS_CLK_ID_QAIF_IF8_IBIT 87
-#define LAPSS_CLK_ID_QAIF_IF8_EBIT 88
-#define LAPSS_CLK_ID_QAIF_IF9_IBIT 89
-#define LAPSS_CLK_ID_QAIF_IF9_EBIT 90
-#define LAPSS_CLK_ID_QAIF_IF10_IBIT 91
-#define LAPSS_CLK_ID_QAIF_IF10_EBIT 92
-#define LAPSS_CLK_ID_QAIF_IF11_IBIT 93
-#define LAPSS_CLK_ID_QAIF_IF11_EBIT 94
-#define LAPSS_CLK_ID_QAIF_IF12_IBIT 95
-#define LAPSS_CLK_ID_QAIF_IF12_EBIT 96
-#define LAPSS_CLK_ID_VA_QAIF_IF0_IBIT 97
-#define LAPSS_CLK_ID_VA_QAIF_IF0_EBIT 98
+#define LPASS_CLK_ID_QAIF_IF0_IBIT 71
+#define LPASS_CLK_ID_QAIF_IF0_EBIT 72
+#define LPASS_CLK_ID_QAIF_IF1_IBIT 73
+#define LPASS_CLK_ID_QAIF_IF1_EBIT 74
+#define LPASS_CLK_ID_QAIF_IF2_IBIT 75
+#define LPASS_CLK_ID_QAIF_IF2_EBIT 76
+#define LPASS_CLK_ID_QAIF_IF3_IBIT 77
+#define LPASS_CLK_ID_QAIF_IF3_EBIT 78
+#define LPASS_CLK_ID_QAIF_IF4_IBIT 79
+#define LPASS_CLK_ID_QAIF_IF4_EBIT 80
+#define LPASS_CLK_ID_QAIF_IF5_IBIT 81
+#define LPASS_CLK_ID_QAIF_IF5_EBIT 82
+#define LPASS_CLK_ID_QAIF_IF6_IBIT 83
+#define LPASS_CLK_ID_QAIF_IF6_EBIT 84
+#define LPASS_CLK_ID_QAIF_IF7_IBIT 85
+#define LPASS_CLK_ID_QAIF_IF7_EBIT 86
+#define LPASS_CLK_ID_QAIF_IF8_IBIT 87
+#define LPASS_CLK_ID_QAIF_IF8_EBIT 88
+#define LPASS_CLK_ID_QAIF_IF9_IBIT 89
+#define LPASS_CLK_ID_QAIF_IF9_EBIT 90
+#define LPASS_CLK_ID_QAIF_IF10_IBIT 91
+#define LPASS_CLK_ID_QAIF_IF10_EBIT 92
+#define LPASS_CLK_ID_QAIF_IF11_IBIT 93
+#define LPASS_CLK_ID_QAIF_IF11_EBIT 94
+#define LPASS_CLK_ID_QAIF_IF12_IBIT 95
+#define LPASS_CLK_ID_QAIF_IF12_EBIT 96
+#define LPASS_CLK_ID_VA_QAIF_IF0_IBIT 97
+#define LPASS_CLK_ID_VA_QAIF_IF0_EBIT 98
#define LPASS_HW_AVTIMER_VOTE 101
#define LPASS_HW_MACRO_VOTE 102
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -179,7 +179,8 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info
int snd_rawmidi_kernel_open_nested(struct snd_rawmidi *rmidi, int subdevice,
int mode, struct snd_rawmidi_file *rfile,
int depth);
-int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile);
+int snd_rawmidi_kernel_release_nested(struct snd_rawmidi_file *rfile,
+ int depth);
int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
struct snd_rawmidi_params *params);
int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
@@ -201,6 +202,11 @@ static inline int snd_rawmidi_kernel_open(struct snd_rawmidi *rmidi,
return snd_rawmidi_kernel_open_nested(rmidi, subdevice, mode, rfile, 0);
}
+static inline int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile)
+{
+ return snd_rawmidi_kernel_release_nested(rfile, 0);
+}
+
/* set up the tied devices */
static inline void snd_rawmidi_tie_devices(struct snd_rawmidi *r1,
struct snd_rawmidi *r2)
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -240,6 +240,8 @@ static int i2sbus_add_dev(struct macio_dev *macio,
}
for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
int irq = irq_of_parse_and_map(np, i);
+ if (!irq)
+ goto err;
if (request_irq(irq, ints[i], 0, dev->rnames[i], dev))
goto err;
dev->interrupts[i] = irq;
diff --git a/sound/core/control_led.c b/sound/core/control_led.c
--- a/sound/core/control_led.c
+++ b/sound/core/control_led.c
@@ -255,6 +255,8 @@ static int snd_ctl_led_set_id(int card_number, struct snd_ctl_elem_id *id,
kctl = snd_ctl_find_id(card, id);
if (!kctl)
return -ENOENT;
+ if (!kctl->info || !kctl->get)
+ return -EINVAL;
ioff = snd_ctl_get_ioff(kctl, id);
vd = &kctl->vd[ioff];
access = vd->access & SNDRV_CTL_ELEM_ACCESS_LED_MASK;
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -571,7 +571,6 @@ static void rawmidi_release_priv(struct snd_rawmidi_file *rfile)
struct snd_rawmidi *rmidi;
rmidi = rfile->rmidi;
- guard(mutex)(&rmidi->open_mutex);
if (rfile->input) {
close_substream(rmidi, rfile->input, 1);
rfile->input = NULL;
@@ -585,7 +584,8 @@ static void rawmidi_release_priv(struct snd_rawmidi_file *rfile)
}
/* called from sound/core/seq/seq_midi.c */
-int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile)
+int snd_rawmidi_kernel_release_nested(struct snd_rawmidi_file *rfile,
+ int depth)
{
struct snd_rawmidi *rmidi;
@@ -593,11 +593,13 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile)
return -ENXIO;
rmidi = rfile->rmidi;
+ mutex_lock_nested(&rmidi->open_mutex, depth);
rawmidi_release_priv(rfile);
+ mutex_unlock(&rmidi->open_mutex);
module_put(rmidi->card->module);
return 0;
}
-EXPORT_SYMBOL(snd_rawmidi_kernel_release);
+EXPORT_SYMBOL(snd_rawmidi_kernel_release_nested);
static int snd_rawmidi_release(struct inode *inode, struct file *file)
{
@@ -607,7 +609,8 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file)
rfile = file->private_data;
rmidi = rfile->rmidi;
- rawmidi_release_priv(rfile);
+ scoped_guard(mutex, &rmidi->open_mutex)
+ rawmidi_release_priv(rfile);
kfree(rfile);
module = rmidi->card->module;
snd_card_file_remove(rmidi->card, file);
diff --git a/sound/core/ump.c b/sound/core/ump.c
--- a/sound/core/ump.c
+++ b/sound/core/ump.c
@@ -1184,7 +1184,8 @@ static int snd_ump_legacy_close(struct snd_rawmidi_substream *substream)
ump->legacy_substreams[dir][group] = NULL;
if (dir == SNDRV_RAWMIDI_STREAM_OUTPUT) {
if (!--ump->legacy_out_opens)
- snd_rawmidi_kernel_release(&ump->legacy_out_rfile);
+ snd_rawmidi_kernel_release_nested(&ump->legacy_out_rfile,
+ SINGLE_DEPTH_NESTING);
}
return 0;
}
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -642,9 +642,7 @@ static void snd_mtpav_free(struct snd_card *card)
{
struct mtpav *crd = card->private_data;
- guard(spinlock_irqsave)(&crd->spinlock);
- if (crd->istimer > 0)
- snd_mtpav_remove_output_timer(crd);
+ timer_shutdown_sync(&crd->timer);
}
/*
diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c
--- a/sound/hda/codecs/conexant.c
+++ b/sound/hda/codecs/conexant.c
@@ -207,7 +207,7 @@ static void cx_remove(struct hda_codec *codec)
snd_hda_gen_remove(codec);
}
-static void cx_process_headset_plugin(struct hda_codec *codec)
+static void cx_process_headset_detect_plug_type(struct hda_codec *codec)
{
unsigned int val;
unsigned int count = 0;
@@ -223,11 +223,9 @@ static void cx_process_headset_plugin(struct hda_codec *codec)
count++;
} while (count < 3);
val = snd_hda_codec_read(codec, 0x1c, 0, 0xcb0, 0x0);
- if (val & 0x800) {
- codec_dbg(codec, "headset plugin, type is CTIA\n");
- snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
- } else if (val & 0x400) {
- codec_dbg(codec, "headset plugin, type is OMTP\n");
+ if (val & 0xc00) {
+ codec_dbg(codec, "headset plugin, type is %s\n",
+ val & 0x800 ? "CTIA" : "OMTP");
snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
} else {
codec_dbg(codec, "headphone plugin\n");
@@ -243,10 +241,12 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_
* Check hp&mic tag to process headset plugin & plugout.
*/
mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0);
- if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */
+ if (!(mic_present & AC_PINSENSE_PRESENCE)) { /* mic plugout */
snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20);
- else
- cx_process_headset_plugin(codec);
+ } else {
+ cx_process_headset_detect_plug_type(codec);
+ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
+ }
}
static int cx_suspend(struct hda_codec *codec)
diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c
--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -1370,6 +1370,61 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10);
}
+/*
+ * HP Laptop 15-fd0xxx (SSID 103c:8bb6) Speaker Mute LED fix
+ *
+ * The speaker mute LED is controlled via VREF100 on NID 0x1a.
+ * This pin must remain powered (D0) even during suspend, otherwise
+ * the LED state is lost and the pin defaults to Hi-Z on resume,
+ * causing the LED to stop responding to mute toggles.
+ * Windows keeps this pin powered unconditionally, so matching that
+ * behavior ensures consistent LED operation across suspend/resume.
+ * The mic-mute LED is controlled via GPIO 0 with active-low polarity.
+ */
+static unsigned int hp_8bb6_power_filter(struct hda_codec *codec,
+ hda_nid_t nid,
+ unsigned int power_state)
+{
+ if (nid == 0x1a)
+ return AC_PWRST_D0;
+ return snd_hda_gen_path_power_filter(codec, nid, power_state);
+}
+
+static int hp_8bb6_speaker_mute_led_set(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
+{
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
+ unsigned int val = (brightness == LED_OFF) ? PIN_IN : PIN_VREF100;
+
+ snd_hda_set_pin_ctl_cache(codec, 0x1a, val);
+ return 0;
+}
+
+static void alc236_fixup_hp_15_fd0xxx(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->micmute_led_polarity = 1;
+ alc_fixup_hp_gpio_led(codec, action, 0x00, 0x01);
+
+ spec->mute_led_polarity = 0;
+ snd_hda_gen_add_mute_led_cdev(codec, hp_8bb6_speaker_mute_led_set);
+
+ codec->power_filter = hp_8bb6_power_filter;
+ spec->no_shutup_pins = 1;
+ break;
+
+ case HDA_FIXUP_ACT_INIT:
+ if (spec->gen.vmaster_mute.hook)
+ snd_hda_sync_vmaster_hook(&spec->gen.vmaster_mute);
+ break;
+ }
+}
+
static void alc285_fixup_hp_gpio_led(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -4089,6 +4144,7 @@ enum {
ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE,
ALC225_FIXUP_S3_POP_NOISE,
ALC700_FIXUP_INTEL_REFERENCE,
+ ALC700_FIXUP_INTEL_HADES_CANYON,
ALC274_FIXUP_DELL_BIND_DACS,
ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
ALC298_FIXUP_TPT470_DOCK_FIX,
@@ -4152,6 +4208,7 @@ enum {
ALC236_FIXUP_HP_GPIO_LED,
ALC236_FIXUP_HP_MUTE_LED,
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
+ ALC236_FIXUP_HP_15_FD0XXX,
ALC236_FIXUP_LENOVO_INV_DMIC,
ALC298_FIXUP_SAMSUNG_AMP,
ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS,
@@ -4297,6 +4354,7 @@ enum {
ALC287_FIXUP_LENOVO_LEGION_AW88399,
ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN_HEADSET,
ALC285_LENOVO_DAC_RENAME,
+ ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1,
};
/* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -4333,6 +4391,19 @@ static void alc287_fixup_lenovo_yoga_book_9i(struct hda_codec *codec,
__snd_hda_apply_fixup(codec, id, action, 0);
}
+static void alc700_fixup_intel_hades_canyon(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action)
+{
+ /*
+ * Windows may leave coef 0x1b bit 0x0400 cleared, causing broken
+ * analog playback after rebooting into Linux. Restore the bit
+ * during codec initialization.
+ */
+ if (action == HDA_FIXUP_ACT_INIT)
+ alc_update_coef_idx(codec, 0x1b, 0x0400, 0x0400);
+}
+
static const struct hda_fixup alc269_fixups[] = {
[ALC298_FIXUP_RAZER_BLADE16_2025_PINS] = {
.type = HDA_FIXUP_PINS,
@@ -5411,6 +5482,10 @@ static const struct hda_fixup alc269_fixups[] = {
{}
}
},
+ [ALC700_FIXUP_INTEL_HADES_CANYON] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc700_fixup_intel_hades_canyon,
+ },
[ALC274_FIXUP_DELL_BIND_DACS] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc274_fixup_bind_dacs,
@@ -5913,6 +5988,10 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc236_fixup_hp_mute_led_micmute_gpio,
},
+ [ALC236_FIXUP_HP_15_FD0XXX] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc236_fixup_hp_15_fd0xxx,
+ },
[ALC236_FIXUP_LENOVO_INV_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic,
@@ -7010,6 +7089,12 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc285_lenovo_dac_rename,
},
+ [ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC287_FIXUP_TXNW2781_I2C,
+ },
};
static const struct hda_quirk alc269_fixup_tbl[] = {
@@ -7448,6 +7533,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2),
+ SND_PCI_QUIRK(0x103c, 0x8bb6, "HP Laptop 15-fd0039nt", ALC236_FIXUP_HP_15_FD0XXX),
SND_PCI_QUIRK(0x103c, 0x8bbe, "HP Victus 16-r0xxx (MB 8BBE)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
SND_PCI_QUIRK(0x103c, 0x8bc8, "HP Victus 15-fa1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
SND_PCI_QUIRK(0x103c, 0x8bcd, "HP Omen 16-xd0xxx", ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT),
@@ -8031,6 +8117,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x3843, "Lenovo Yoga 9i / Yoga Book 9i", ALC287_FIXUP_LENOVO_YOGA_BOOK_9I),
+ SND_PCI_QUIRK(0x17aa, 0x3846, "Lenovo Yoga Pro 9 16IAH10", ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1),
/* Yoga Pro 7 14IMH9 shares PCI SSID 17aa:3847 with Legion 7 16ACHG6;
* use codec SSID to distinguish them
*/
@@ -8044,14 +8131,15 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6),
+ SND_PCI_QUIRK(0x17aa, 0x3862, "Lenovo IdeaPad Slim 3 15ABR8", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
HDA_CODEC_QUIRK(0x17aa, 0x386a, "Lenovo Yoga 7 16IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x386e, "Yoga Pro 7 14ARP8", ALC285_FIXUP_YOGA_SPEAKER2_TO_DAC1),
- HDA_CODEC_QUIRK(0x17aa, 0x38a8, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */
HDA_CODEC_QUIRK(0x17aa, 0x38a7, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */
+ HDA_CODEC_QUIRK(0x17aa, 0x38a8, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */
SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C),
SND_PCI_QUIRK(0x17aa, 0x3874, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2),
@@ -8231,6 +8319,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x2782, 0xa128, "Positivo N15RPE-S", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x2782, 0xa212, "Lunnen Ground 14", ALC269VC_FIXUP_LUNNEN_GROUND_14),
SND_PCI_QUIRK(0x7017, 0x2014, "Star Labs StarFighter", ALC233_FIXUP_STARLABS_STARFIGHTER),
+ SND_PCI_QUIRK(0x8086, 0x2073, "Intel NUC 8 Hades Canyon",
+ ALC700_FIXUP_INTEL_HADES_CANYON),
SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC),
SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10),
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2523,7 +2523,7 @@ static int snd_ice1712_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
static int dev;
- struct snd_card *card;
+ struct snd_card *card __free(snd_card_unref) = NULL;
struct snd_ice1712 *ice;
int pcm_dev = 0, err;
const struct snd_ice1712_card_info * const *tbl, *c;
@@ -2535,8 +2535,8 @@ static int snd_ice1712_probe(struct pci_dev *pci,
return -ENOENT;
}
- err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
- sizeof(*ice), &card);
+ err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+ sizeof(*ice), &card);
if (err < 0)
return err;
ice = card->private_data;
@@ -2640,6 +2640,7 @@ static int snd_ice1712_probe(struct pci_dev *pci,
if (err < 0)
return err;
pci_set_drvdata(pci, card);
+ card = NULL; /* probe succeeded, don't release as error */
dev++;
return 0;
}
diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c
--- a/sound/soc/amd/acp-config.c
+++ b/sound/soc/amd/acp-config.c
@@ -54,7 +54,7 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = {
{
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC"),
- DMI_MATCH(DMI_PRODUCT_NAME, "TUF Gaming A14 FA401EA"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "FA401EA"),
},
},
{
@@ -70,6 +70,13 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = {
DMI_MATCH(DMI_BOARD_NAME, "8EA1"),
},
},
+ {
+ /* HP OmniBook X Flip 16-cc0xxx */
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "HP"),
+ DMI_MATCH(DMI_BOARD_NAME, "8EA2"),
+ },
+ },
{}
};
@@ -209,6 +216,20 @@ static const struct config_entry config_table[] = {
{}
},
},
+ {
+ .flags = FLAG_AMD_LEGACY,
+ .device = ACP_PCI_DEV_ID,
+ .dmi_table = (const struct dmi_system_id []) {
+ {
+ .matches = {
+ DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1060"),
+ },
+ },
+ {}
+ },
+ },
};
static int snd_amd_acp_acpi_find_config(struct pci_dev *pci)
diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c
--- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c
+++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c
@@ -382,6 +382,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = {
},
.driver_data = (void *)(ES83XX_ENABLE_DMIC),
},
+ {
+ .matches = {
+ DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1060"),
+ },
+ .driver_data = (void *)(ES83XX_ENABLE_DMIC),
+ },
{}
};
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -570,6 +570,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m18 R1 AMD"),
+ }
+ },
{
.driver_data = &acp6x_card,
.matches = {
diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -360,7 +360,7 @@ static int max98926_dai_set_fmt(struct snd_soc_dai *codec_dai,
regmap_write(max98926->regmap,
MAX98926_FORMAT, MAX98926_DAI_DLY_MASK);
regmap_update_bits(max98926->regmap, MAX98926_FORMAT,
- MAX98926_DAI_BCI_MASK, invert);
+ MAX98926_DAI_BCI_MASK | MAX98926_DAI_WCI_MASK, invert);
return 0;
}
diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c
--- a/sound/soc/codecs/rt712-sdca.c
+++ b/sound/soc/codecs/rt712-sdca.c
@@ -1779,6 +1779,7 @@ static void rt712_sdca_vb_io_init(struct rt712_sdca_priv *rt712)
dev_dbg(dev, "%s jack/mic/amp func_status=0x%x, 0x%x, 0x%x\n",
__func__, jack_func_status, mic_func_status, amp_func_status);
+ rt712_sdca_index_write(rt712, RT712_VENDOR_REG, RT712_JD_CTL3, 0x7778);
/* DMIC */
if ((mic_func_status & FUNCTION_NEEDS_INITIALIZATION) || (!rt712->first_hw_init)) {
rt712_sdca_index_write(rt712, RT712_VENDOR_HDA_CTL, RT712_DMIC2_FU_IT_FLOAT_CTL, 0x1526);
diff --git a/sound/soc/codecs/rt712-sdca.h b/sound/soc/codecs/rt712-sdca.h
--- a/sound/soc/codecs/rt712-sdca.h
+++ b/sound/soc/codecs/rt712-sdca.h
@@ -74,6 +74,7 @@ struct rt712_dmic_kctrl_priv {
#define RT712_JD_PRODUCT_NUM 0x00
#define RT712_ANALOG_BIAS_CTL3 0x04
#define RT712_JD_CTL1 0x09
+#define RT712_JD_CTL3 0x0b
#define RT712_IO_CTL 0x0c
#define RT712_LDO2_3_CTL1 0x0e
#define RT712_PARA_VERB_CTL 0x1a
diff --git a/sound/soc/codecs/rt766-sdca-sdw.c b/sound/soc/codecs/rt766-sdca-sdw.c
--- a/sound/soc/codecs/rt766-sdca-sdw.c
+++ b/sound/soc/codecs/rt766-sdca-sdw.c
@@ -233,6 +233,8 @@ static void rt766_sdca_sdw_remove(struct sdw_slave *slave)
static const struct sdw_device_id rt766_sdca_id[] = {
SDW_SLAVE_ENTRY_EXT(0x025d, 0x766, 0x3, 0x1, 0),
SDW_SLAVE_ENTRY_EXT(0x025d, 0x767, 0x3, 0x1, 0),
+ SDW_SLAVE_ENTRY_EXT(0x025d, 0x766, 0x4, 0x1, 0),
+ SDW_SLAVE_ENTRY_EXT(0x025d, 0x767, 0x4, 0x1, 0),
{},
};
MODULE_DEVICE_TABLE(sdw, rt766_sdca_id);
diff --git a/sound/soc/codecs/tac5xx2-sdw.c b/sound/soc/codecs/tac5xx2-sdw.c
--- a/sound/soc/codecs/tac5xx2-sdw.c
+++ b/sound/soc/codecs/tac5xx2-sdw.c
@@ -55,6 +55,12 @@
#define TAC_XU_BYPASS_REG(func, xu) \
SDW_SDCA_CTL(TAC_FUNCTION_ID_##func, TAC_SDCA_ENT_##xu, \
TAC_SDCA_CTL_XU_BYPASS, 0)
+#define TAC_VOLUME_REG(func, fu, ch) \
+ SDW_SDCA_CTL(TAC_FUNCTION_ID_##func, TAC_SDCA_ENT_##fu, \
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_##ch)
+#define TAC_GAIN_REG(func, fu, ch) \
+ SDW_SDCA_CTL(TAC_FUNCTION_ID_##func, TAC_SDCA_ENT_##fu, \
+ TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_##ch)
/* mute registers */
#define FU21_L_MUTE_REG TAC_MUTE_REG(SA, FU21, LEFT)
@@ -123,9 +129,9 @@ static const DECLARE_TLV_DB_SCALE(tac5xx2_dvc_tlv, -7200, 50, 0);
#define TAC_DOUBLE_Q78_TLV(name, func_id, ent_id) \
SDCA_DOUBLE_Q78_TLV(name, \
SDW_SDCA_CTL(TAC_FUNCTION_ID_##func_id, TAC_SDCA_ENT_##ent_id, \
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT), \
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_LEFT), \
SDW_SDCA_CTL(TAC_FUNCTION_ID_##func_id, TAC_SDCA_ENT_##ent_id, \
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT), \
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT), \
TAC_DVC_MIN, TAC_DVC_MAX, TAC_DVC_STEP, tac5xx2_dvc_tlv)
struct tac5xx2_prv {
@@ -381,37 +387,53 @@ static const struct reg_default tac_reg_default[] = {
{TAC_REG_SDW(0, 1, 0x71), 0x5},
{TAC_REG_SDW(0, 1, 0x72), 0x6},
{TAC_REG_SDW(0, 1, 0x73), 0x7},
+ /* SA */
+ {TAC_MUTE_REG(SA, FU21, LEFT), 0x1},
+ {TAC_MUTE_REG(SA, FU21, RIGHT), 0x1},
+ {TAC_VOLUME_REG(SA, FU21, LEFT), 0x9c00},
+ {TAC_VOLUME_REG(SA, FU21, RIGHT), 0x9c00},
+ {TAC_MUTE_REG(SA, FU23, LEFT), 0x1},
+ {TAC_MUTE_REG(SA, FU23, RIGHT), 0x1},
+ {TAC_GAIN_REG(SA, FU23, LEFT), 0x0},
+ {TAC_GAIN_REG(SA, FU23, RIGHT), 0x0},
+ /* SM */
+ {TAC_USAGE_REG(SM, IT11), 0x0},
+ {TAC_USAGE_REG(SM, OT113), 0x0},
+ {TAC_MUTE_REG(SM, FU113, LEFT), 0x1},
+ {TAC_MUTE_REG(SM, FU113, RIGHT), 0x1},
+ {TAC_GAIN_REG(SM, FU113, LEFT), 0x0},
+ {TAC_GAIN_REG(SM, FU113, RIGHT), 0x0},
+ {TAC_MUTE_REG(SM, FU11, LEFT), 0x1},
+ {TAC_MUTE_REG(SM, FU11, RIGHT), 0x1},
+ {TAC_GAIN_REG(SM, FU11, LEFT), 0x0},
+ {TAC_GAIN_REG(SM, FU11, RIGHT), 0x0},
+ {TAC_XU_BYPASS_REG(SM, XU12), 0x1},
+ {SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_CS113,
+ TAC_SDCA_CTL_CS_SAMP_RATE_IDX, 0), 0x0},
+ /* UAJ */
+ {TAC_USAGE_REG(UAJ, IT33), 0x0},
+ {TAC_USAGE_REG(UAJ, IT41), 0x0},
+ {TAC_USAGE_REG(UAJ, OT36), 0x0},
+ {TAC_USAGE_REG(UAJ, OT45), 0x0},
+ {TAC_MUTE_REG(UAJ, FU41, LEFT), 0x1},
+ {TAC_MUTE_REG(UAJ, FU41, RIGHT), 0x1},
+ {TAC_VOLUME_REG(UAJ, FU41, LEFT), 0x0},
+ {TAC_VOLUME_REG(UAJ, FU41, RIGHT), 0x0},
+ {TAC_MUTE_REG(UAJ, FU36, RIGHT), 0x1},
+ {SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU36,
+ TAC_SDCA_CHANNEL_VOLUME, TAC_JACK_MONO_CS), 0x0},
+ {TAC_XU_BYPASS_REG(UAJ, XU42), 0x0},
+ {SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_CS36,
+ TAC_SDCA_CTL_CS_SAMP_RATE_IDX, 0), 0x0},
+ {SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_CS41,
+ TAC_SDCA_CTL_CS_SAMP_RATE_IDX, 0), 0x0},
};
static const struct reg_sequence tac_spk_seq[] = {
REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU21,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT), 0),
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_LEFT), 0),
REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU21,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU23,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU23,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT), 0),
-};
-
-static const struct reg_sequence tac_sm_seq[] = {
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU113,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU113,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU11,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU11,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT), 0),
-};
-
-static const struct reg_sequence tac_uaj_seq[] = {
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU41,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU41,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT), 0),
- REG_SEQ0(SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU36,
- TAC_SDCA_CHANNEL_GAIN, TAC_JACK_MONO_CS), 0),
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT), 0),
};
static bool tac_volatile_reg(struct device *dev, unsigned int reg)
@@ -468,25 +490,25 @@ static int tac_sdca_mbq_size(struct device *dev, unsigned int reg)
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU21,
TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU23,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_LEFT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU23,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SA, TAC_SDCA_ENT_FU23,
- TAC_SDCA_MASTER_GAIN, 0):
+ TAC_SDCA_CHANNEL_GAIN, 0):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU113,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_LEFT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU113,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU11,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_LEFT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_SM, TAC_SDCA_ENT_FU11,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU41,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_LEFT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_LEFT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU41,
- TAC_SDCA_CHANNEL_GAIN, TAC_CHANNEL_RIGHT):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_CHANNEL_RIGHT):
case SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU36,
- TAC_SDCA_CHANNEL_GAIN, TAC_JACK_MONO_CS):
+ TAC_SDCA_CHANNEL_VOLUME, TAC_JACK_MONO_CS):
return 2;
default:
@@ -531,7 +553,7 @@ static const struct snd_kcontrol_new tac_uaj_controls[] = {
TAC_DOUBLE_Q78_TLV("UAJ Playback Volume", UAJ, FU41),
SDCA_SINGLE_Q78_TLV("UAJ Capture Volume",
SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_FU36,
- TAC_SDCA_CHANNEL_GAIN, TAC_JACK_MONO_CS),
+ TAC_SDCA_CHANNEL_VOLUME, TAC_JACK_MONO_CS),
TAC_DVC_MIN, TAC_DVC_MAX, TAC_DVC_STEP, tac5xx2_dvc_tlv),
};
@@ -1761,16 +1783,6 @@ static int tac_io_init(struct device *dev, struct sdw_slave *slave, bool first)
goto io_init_err;
}
dev_dbg(dev, "smartmic init done\n");
-
- if (first) {
- ret = regmap_multi_reg_write(tac_dev->regmap, tac_sm_seq,
- ARRAY_SIZE(tac_sm_seq));
- if (ret) {
- dev_err(tac_dev->dev,
- "init writes failed, err=%d", ret);
- goto io_init_err;
- }
- }
}
if (tac_dev->uaj_func_data) {
@@ -1783,14 +1795,6 @@ static int tac_io_init(struct device *dev, struct sdw_slave *slave, bool first)
dev_dbg(dev, "uaj init done\n");
if (first) {
- ret = regmap_multi_reg_write(tac_dev->regmap, tac_uaj_seq,
- ARRAY_SIZE(tac_uaj_seq));
- if (ret) {
- dev_err(tac_dev->dev,
- "init writes failed, err=%d", ret);
- goto io_init_err;
- }
-
if (tac_dev->hs_jack) {
ret = tac5xx2_jack_init(tac_dev);
if (ret) {
diff --git a/sound/soc/codecs/tac5xx2.h b/sound/soc/codecs/tac5xx2.h
--- a/sound/soc/codecs/tac5xx2.h
+++ b/sound/soc/codecs/tac5xx2.h
@@ -174,14 +174,13 @@
#define TAC_SDCA_ENT_SAPU29 0x35
/* Control selector definitions */
-#define TAC_SDCA_MASTER_GAIN 0x0B
#define TAC_SDCA_MASTER_MUTE 0x01
#define TAC_SDCA_CHANNEL_MUTE 0x01
-#define TAC_SDCA_CHANNEL_GAIN 0x02
+#define TAC_SDCA_CHANNEL_VOLUME 0x02
#define TAC_SDCA_POSTURENUMBER 0x10
#define TAC_SDCA_REQUESTED_PS 0x01
#define TAC_SDCA_ACTUAL_PS 0x10
-#define TAC_SDCA_CHANNEL_VOLUME 0x02
+#define TAC_SDCA_CHANNEL_GAIN 0x0B
/* 2. smart mic function */
#define TAC_FUNCTION_ID_SM 0x2
diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c
--- a/sound/soc/codecs/tas2783-sdw.c
+++ b/sound/soc/codecs/tas2783-sdw.c
@@ -1310,6 +1310,7 @@ static int tas_port_prep(struct sdw_slave *slave, struct sdw_prepare_ch *prep_ch
enum sdw_port_prep_ops pre_ops)
{
struct device *dev = &slave->dev;
+ struct tas2783_prv *tas_dev = dev_get_drvdata(dev);
struct sdw_dpn_prop *dpn_prop;
u32 addr;
int ret;
@@ -1321,6 +1322,25 @@ static int tas_port_prep(struct sdw_slave *slave, struct sdw_prepare_ch *prep_ch
addr = SDW_DPN_PREPARECTRL(prep_ch->num);
switch (pre_ops) {
case SDW_OPS_PORT_PRE_PREP:
+ /*
+ * The Function has to be powered before the port can complete
+ * channel preparation. hw_params() does that when a stream is
+ * set up, but a stream that is only re-prepared - as it is
+ * after the peripheral lost power in S0i3 - does not go
+ * through hw_params() again, and the peripheral is back at its
+ * PS3 reset default. Power it up here, where it is needed.
+ */
+ scoped_guard(mutex, &tas_dev->pde_lock)
+ ret = regmap_write(tas_dev->regmap,
+ SDW_SDCA_CTL(1, TAS2783_SDCA_ENT_PDE23,
+ TAS2783_SDCA_CTL_REQ_POW_STATE, 0),
+ TAS2783_SDCA_POW_STATE_ON);
+ if (ret) {
+ dev_err(dev, "power up failed for port %d, err=%d\n",
+ prep_ch->num, ret);
+ return ret;
+ }
+
ret = sdw_write_no_pm(slave, addr, prep_ch->ch_mask);
if (ret)
dev_err(dev, "prep failed for port %d, err=%d\n",
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -244,7 +244,7 @@ config SND_SOC_MPC5200_AC97
config SND_MPC52xx_SOC_PCM030
tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
depends on PPC_MPC5200_SIMPLE
- select SND_SOC_MPC5200_AC97
+ depends on SND_SOC_MPC5200_AC97
select SND_SOC_WM9712
help
Say Y if you want to add support for sound on the Phytec pcm030
@@ -253,7 +253,7 @@ config SND_MPC52xx_SOC_PCM030
config SND_MPC52xx_SOC_EFIKA
tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
depends on PPC_EFIKA
- select SND_SOC_MPC5200_AC97
+ depends on SND_SOC_MPC5200_AC97
select SND_SOC_STAC9766
help
Say Y if you want to add support for sound on the Efika.
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -22,8 +22,6 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include "mpc5200_dma.h"
-
#define DRV_NAME "efika-audio-fabric"
SND_SOC_DAILINK_DEFS(analog,
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -13,8 +13,6 @@
#include <sound/soc.h>
-#include "mpc5200_dma.h"
-
#define DRV_NAME "pcm030-audio-fabric"
struct pcm030_audio_data {
diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c
--- a/sound/soc/loongson/loongson_card.c
+++ b/sound/soc/loongson/loongson_card.c
@@ -55,8 +55,8 @@ static const struct loongson_card_config ls2k0300_dl2k0300b_card_config = {
.add_dapm_routes = true,
};
-static int tegra_machine_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
+static int loongson_asoc_machine_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
{
struct snd_soc_card *card = snd_soc_dapm_to_card(w->dapm);
struct loongson_card_data *priv = snd_soc_card_get_drvdata(card);
@@ -72,9 +72,9 @@ static int tegra_machine_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const struct snd_soc_dapm_widget loongson_aosc_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", tegra_machine_event),
- SND_SOC_DAPM_SPK("Speaker", tegra_machine_event),
+static const struct snd_soc_dapm_widget loongson_asoc_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", loongson_asoc_machine_event),
+ SND_SOC_DAPM_SPK("Speaker", loongson_asoc_machine_event),
};
/* Headphones Jack */
@@ -314,8 +314,8 @@ static int loongson_asoc_card_probe(struct platform_device *pdev)
card->num_links = ARRAY_SIZE(loongson_dai_links);
if (ls_priv->cfg->add_dapm_widgets) {
- card->dapm_widgets = loongson_aosc_dapm_widgets;
- card->num_dapm_widgets = ARRAY_SIZE(loongson_aosc_dapm_widgets);
+ card->dapm_widgets = loongson_asoc_dapm_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(loongson_asoc_dapm_widgets);
}
snd_soc_card_set_drvdata(card, ls_priv);
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -103,14 +103,14 @@ static int qcom_snd_parse_dai_tdm_slots(struct snd_soc_pcm_runtime *rtd,
int ret;
if (!link_np)
- return -EINVAL;
+ return -ENOENT;
struct device_node *cpu_np __free(device_node) =
of_get_child_by_name(link_np, "cpu");
struct device_node *codec_np __free(device_node) =
of_get_child_by_name(link_np, "codec");
if (!cpu_np || !codec_np)
- return -EINVAL;
+ return -ENOENT;
ret = qcom_snd_parse_tdm_slot(cpu_np, cpu_cfg);
if (ret)
@@ -172,7 +172,7 @@ int qcom_snd_apply_dai_tdm_slots(struct snd_soc_pcm_runtime *rtd)
ret = qcom_snd_get_dai_tdm_slots(rtd, &cpu_cfg, &codec_cfg);
if (ret)
- return ret == -EINVAL ? 0 : ret;
+ return ret == -ENOENT ? 0 : ret;
return qcom_snd_apply_dai_tdm_slots_cfg(rtd, &cpu_cfg, &codec_cfg);
}
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -216,7 +216,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
- struct audioreach_module_config cfg;
+ struct audioreach_module_config cfg = {};
struct device *dev = component->dev;
struct q6apm_dai_data *pdata;
int ret;
@@ -814,7 +814,7 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
struct q6apm_dai_data *pdata;
- struct audioreach_module_config cfg;
+ struct audioreach_module_config cfg = {};
struct snd_codec *codec = &params->codec;
int dir = stream->direction;
int ret;
diff --git a/sound/soc/qcom/qdsp6/q6prm-clocks.c b/sound/soc/qcom/qdsp6/q6prm-clocks.c
--- a/sound/soc/qcom/qdsp6/q6prm-clocks.c
+++ b/sound/soc/qcom/qdsp6/q6prm-clocks.c
@@ -64,34 +64,34 @@ static const struct q6dsp_clk_init q6prm_clks[] = {
Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_MCLK),
Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK),
Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF0_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF0_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF1_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF1_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF2_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF2_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF3_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF3_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF4_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF4_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF5_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF5_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF6_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF6_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF7_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF7_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF8_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF8_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF9_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF9_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF10_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF10_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF11_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF11_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF12_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_QAIF_IF12_EBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_VA_QAIF_IF0_IBIT),
- Q6PRM_CLK(LAPSS_CLK_ID_VA_QAIF_IF0_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF0_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF0_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF1_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF1_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF2_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF2_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF3_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF3_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF4_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF4_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF5_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF5_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF6_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF6_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF7_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF7_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF8_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF8_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF9_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF9_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF10_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF10_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF11_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF11_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF12_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_QAIF_IF12_EBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_VA_QAIF_IF0_IBIT),
+ Q6PRM_CLK(LPASS_CLK_ID_VA_QAIF_IF0_EBIT),
Q6DSP_VOTE_CLK(LPASS_HW_MACRO_VOTE, Q6PRM_HW_CORE_ID_LPASS,
"LPASS_HW_MACRO"),
Q6DSP_VOTE_CLK(LPASS_HW_DCODEC_VOTE, Q6PRM_HW_CORE_ID_DCODEC,
diff --git a/sound/soc/qcom/qdsp6/q6prm.h b/sound/soc/qcom/qdsp6/q6prm.h
--- a/sound/soc/qcom/qdsp6/q6prm.h
+++ b/sound/soc/qcom/qdsp6/q6prm.h
@@ -97,34 +97,34 @@
/* Clock ID for RX CORE MCLK2 2X MCLK */
#define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 0x318
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF0_IBIT 0x500
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF0_EBIT 0x501
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF1_IBIT 0x502
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF1_EBIT 0x503
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF2_IBIT 0x504
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF2_EBIT 0x505
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF3_IBIT 0x506
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF3_EBIT 0x507
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF4_IBIT 0x508
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF4_EBIT 0x509
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF5_IBIT 0x50A
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF5_EBIT 0x50B
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF6_IBIT 0x50C
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF6_EBIT 0x50D
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF7_IBIT 0x50E
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF7_EBIT 0x50F
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF8_IBIT 0x510
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF8_EBIT 0x511
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF9_IBIT 0x512
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF9_EBIT 0x513
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF10_IBIT 0x514
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF10_EBIT 0x515
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF11_IBIT 0x516
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF11_EBIT 0x517
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF12_IBIT 0x518
-#define Q6PRM_LAPSS_CLK_ID_QAIF_IF12_EBIT 0x519
-#define Q6PRM_LAPSS_CLK_ID_VA_QAIF_IF0_IBIT 0x550
-#define Q6PRM_LAPSS_CLK_ID_VA_QAIF_IF0_EBIT 0x551
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF0_IBIT 0x500
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF0_EBIT 0x501
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF1_IBIT 0x502
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF1_EBIT 0x503
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF2_IBIT 0x504
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF2_EBIT 0x505
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF3_IBIT 0x506
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF3_EBIT 0x507
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF4_IBIT 0x508
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF4_EBIT 0x509
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF5_IBIT 0x50A
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF5_EBIT 0x50B
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF6_IBIT 0x50C
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF6_EBIT 0x50D
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF7_IBIT 0x50E
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF7_EBIT 0x50F
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF8_IBIT 0x510
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF8_EBIT 0x511
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF9_IBIT 0x512
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF9_EBIT 0x513
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF10_IBIT 0x514
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF10_EBIT 0x515
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF11_IBIT 0x516
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF11_EBIT 0x517
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF12_IBIT 0x518
+#define Q6PRM_LPASS_CLK_ID_QAIF_IF12_EBIT 0x519
+#define Q6PRM_LPASS_CLK_ID_VA_QAIF_IF0_IBIT 0x550
+#define Q6PRM_LPASS_CLK_ID_VA_QAIF_IF0_EBIT 0x551
#define Q6PRM_LPASS_CLK_SRC_INTERNAL 1
#define Q6PRM_LPASS_CLK_ROOT_DEFAULT 0
diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c
--- a/sound/soc/qcom/sc8280xp.c
+++ b/sound/soc/qcom/sc8280xp.c
@@ -115,33 +115,42 @@ static int sc8280xp_tdm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai;
struct qcom_snd_tdm_slot_cfg cpu_cfg;
struct qcom_snd_tdm_slot_cfg codec_cfg;
- unsigned int bclk_freq;
+ int bclk_freq;
int ret;
int i;
ret = qcom_snd_get_dai_tdm_slots(rtd, &cpu_cfg, &codec_cfg);
if (ret)
- return ret == -EINVAL ? 0 : ret;
+ return ret == -ENOENT ? 0 : ret;
if (!cpu_cfg.slots)
return 0;
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_BP_FP);
- if (ret)
+ if (ret && ret != -ENOTSUPP)
return ret;
+ if (data->priv->codec_dai_fmt) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ data->priv->codec_dai_fmt);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+ }
+
ret = qcom_snd_apply_dai_tdm_slots_cfg(rtd, &cpu_cfg, &codec_cfg);
if (ret)
return ret;
bclk_freq = snd_soc_tdm_params_to_bclk(params, cpu_cfg.slot_width, cpu_cfg.slots, 1);
- if (!bclk_freq)
+ if (bclk_freq <= 0)
return -EINVAL;
if (data->priv->mi2s_bclk_enable) {
ret = snd_soc_dai_set_sysclk(cpu_dai, LPAIF_MI2S_BCLK, bclk_freq,
SND_SOC_CLOCK_IN);
- if (ret) {
+ if (ret && ret != -ENOTSUPP) {
dev_err(rtd->dev, "%s: failed to set cpu sysclk: %d\n",
__func__, ret);
return ret;
@@ -152,7 +161,7 @@ static int sc8280xp_tdm_hw_params(struct snd_pcm_substream *substream,
for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_sysclk(codec_dai, 0, bclk_freq,
SND_SOC_CLOCK_IN);
- if (ret) {
+ if (ret && ret != -ENOTSUPP) {
dev_err(rtd->dev, "%s: failed to set codec sysclk on %s: %d\n",
__func__, codec_dai->name, ret);
return ret;
diff --git a/sound/soc/sdw_utils/soc_sdw_rt_amp.c b/sound/soc/sdw_utils/soc_sdw_rt_amp.c
--- a/sound/soc/sdw_utils/soc_sdw_rt_amp.c
+++ b/sound/soc/sdw_utils/soc_sdw_rt_amp.c
@@ -252,11 +252,13 @@ int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai
if (ctx->amp_dev1) {
device_remove_software_node(ctx->amp_dev1);
put_device(ctx->amp_dev1);
+ ctx->amp_dev1 = NULL;
}
if (ctx->amp_dev2) {
device_remove_software_node(ctx->amp_dev2);
put_device(ctx->amp_dev2);
+ ctx->amp_dev2 = NULL;
}
return 0;
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -81,6 +81,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* speaker */
.direction = {true, false},
.dai_name = "tac5xx2-aif1",
+ .component_name = "tac5572",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
.init = asoc_sdw_ti_amp_init,
@@ -94,6 +95,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* mic */
.direction = {false, true},
.dai_name = "tac5xx2-aif2",
+ .component_name = "tac5572",
.dai_type = SOC_SDW_DAI_TYPE_MIC,
.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID},
.rtd_init = asoc_sdw_ti_dmic_rtd_init,
@@ -102,6 +104,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* UAJ */
.direction = {true, true},
.dai_name = "tac5xx2-aif3",
+ .component_name = "tac5572",
.dai_type = SOC_SDW_DAI_TYPE_JACK,
.dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID},
.controls = generic_jack_controls,
@@ -122,6 +125,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* speaker with IV sense feedback */
.direction = {true, true},
.dai_name = "tac5xx2-aif1",
+ .component_name = "tac5672",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
.init = asoc_sdw_ti_amp_init,
@@ -135,6 +139,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* mic */
.direction = {false, true},
.dai_name = "tac5xx2-aif2",
+ .component_name = "tac5672",
.dai_type = SOC_SDW_DAI_TYPE_MIC,
.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID},
.rtd_init = asoc_sdw_ti_dmic_rtd_init,
@@ -143,6 +148,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* UAJ */
.direction = {true, true},
.dai_name = "tac5xx2-aif3",
+ .component_name = "tac5672",
.dai_type = SOC_SDW_DAI_TYPE_JACK,
.dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID},
.controls = generic_jack_controls,
@@ -163,6 +169,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* speaker with echo reference feedback */
.direction = {true, true},
.dai_name = "tac5xx2-aif1",
+ .component_name = "tac5682",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
.init = asoc_sdw_ti_amp_init,
@@ -176,6 +183,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* mic */
.direction = {false, true},
.dai_name = "tac5xx2-aif2",
+ .component_name = "tac5682",
.dai_type = SOC_SDW_DAI_TYPE_MIC,
.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID},
.rtd_init = asoc_sdw_ti_dmic_rtd_init,
@@ -184,6 +192,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* UAJ */
.direction = {true, true},
.dai_name = "tac5xx2-aif3",
+ .component_name = "tac5682",
.dai_type = SOC_SDW_DAI_TYPE_JACK,
.dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID},
.controls = generic_jack_controls,
@@ -203,6 +212,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
{
.direction = {true, false},
.dai_name = "tac5xx2-aif1",
+ .component_name = "tas2883",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
.init = asoc_sdw_ti_amp_init,
@@ -216,6 +226,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
/* mic */
.direction = {false, true},
.dai_name = "tac5xx2-aif2",
+ .component_name = "tas2883",
.dai_type = SOC_SDW_DAI_TYPE_MIC,
.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID},
.rtd_init = asoc_sdw_ti_dmic_rtd_init,
@@ -1510,9 +1521,24 @@ int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd)
}
switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /*
+ * The peripherals lose their port configuration when the
+ * controller is power-gated during system suspend, and an
+ * application that restarts the stream with
+ * SNDRV_PCM_IOCTL_RESUME - which platforms advertising
+ * SNDRV_PCM_INFO_RESUME allow - never goes through
+ * .prepare() again. Prepare the stream here so that the
+ * ports are reprogrammed before they are enabled;
+ * sdw_prepare_stream() reapplies the parameters without
+ * recomputing them when the stream is disabled.
+ */
+ ret = sdw_prepare_stream(sdw_stream);
+ if (ret)
+ break;
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- case SNDRV_PCM_TRIGGER_RESUME:
ret = sdw_enable_stream(sdw_stream);
break;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3614,7 +3614,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
mask = e->mask << e->shift_l;
if (e->shift_l != e->shift_r) {
- if (item[1] > e->items)
+ if (item[1] >= e->items)
return -EINVAL;
val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r;
mask |= e->mask << e->shift_r;
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -397,24 +397,8 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
if (PTR_ERR(chan) == -EPROBE_DEFER)
return -EPROBE_DEFER;
- bool has_fw_node = dev->of_node || is_acpi_device_node(dev->fwnode);
- bool name_exists_in_fw = false;
-
- if (has_fw_node)
- name_exists_in_fw = device_property_match_string(dev,
- "dma-names",
- name) >= 0;
-
- if (has_fw_node && name_exists_in_fw)
- dev_warn(dev, "DTS/ACPI DMA channel '%s' request failed (%ld)\n",
- name, PTR_ERR(chan));
-
- if (has_fw_node && !name_exists_in_fw)
- dev_warn(dev, "DTS/ACPI name '%s' not found, legacy failed (%ld)\n",
- name, PTR_ERR(chan));
-
- if (!has_fw_node)
- dev_warn(dev, "Legacy DMA channel '%s' request failed (%ld)\n",
+ if (device_property_match_string(dev, "dma-names", name) >= 0)
+ dev_warn(dev, "dma-names has '%s' but request failed (%ld)\n",
name, PTR_ERR(chan));
pcm->chan[i] = NULL;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1237,21 +1237,27 @@ static void init_cur_mix_raw(struct usb_mixer_elem_info *cval, int ch, int idx)
}
/*
- * Additional checks for sticky mixers
+ * Additional checks for sticky GET_CUR
*
- * Some devices' volume control mixers are sticky, which accept SET_CUR but
- * do absolutely nothing.
+ * Some devices' volume control mixers have sticky GET_CUR, which implies either
+ * stubbed SET_CUR or broken GET_CUR. For the former case, the mixer accepts
+ * SET_CUR but do absolutely nothing, so falling back to soft mixer is the only
+ * way to control the volume. For the latter case, the mixer has effective
+ * SET_CUR despite GET_CUR being constant, and the mixer is usable as long as we
+ * always provide mixer value from the ceche.
*
* Check the return values of GET_CUR with different SET_CUR values. Consider
- * the mixer as sticky if GET_CUR always returns a constant value.
+ * GET_CUR as sticky if GET_CUR always returns a constant value.
*
- * Some devices have effective SET_CUR despite GET_CUR being constant. Do not
- * consider the mixer as sticky if a quirk flag indicates that.
+ * Unfortunately, we can't distinguish between stubbed SET_CUR and broken
+ * GET_CUR with simple read-back tests. Disabling the mixer regardless and
+ * forcing userspace to use soft mixer instead can lead to audible distortion at
+ * low volume on some wireless headphones, probably due to their poorly-
+ * performed lossy codec.
*
- * Gate the registration of sticky mixers to prevent confusing userspace, so
- * that they won't cause ineffective volume control. However, for mixers with
- * effective SET_CUR but broken GET_CUR, the registration can continue normally
- * but further GET_CUR requests will be gated.
+ * Instead, mark GET_CUR as broken regardless and only provide mixer value from
+ * the cache. Users may opt into soft mixer in userspace audio stack if they
+ * need it.
*/
static int check_sticky_volume_control(struct usb_mixer_elem_info *cval,
int channel, int saved)
@@ -1271,24 +1277,13 @@ static int check_sticky_volume_control(struct usb_mixer_elem_info *cval,
return 0;
}
- if (cval->head.mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_GET_CUR_BROKEN) {
- usb_audio_info(cval->head.mixer->chip,
- "%d:%d: broken mixer GET_CUR (%d/%d/%d => %d)\n",
- cval->head.id, mixer_ctrl_intf(cval->head.mixer),
- cval->min, cval->max, cval->res, saved);
-
- cval->get_cur_broken = 1;
- return -ENXIO;
- }
-
- usb_audio_err(cval->head.mixer->chip,
- "%d:%d: sticky mixer values (%d/%d/%d => %d), disabling\n",
- cval->head.id, mixer_ctrl_intf(cval->head.mixer),
- cval->min, cval->max, cval->res, saved);
usb_audio_info(cval->head.mixer->chip,
- "check MIXER_GET_CUR_BROKEN if you believe the mixer is non-sticky");
+ "%d:%d: broken mixer GET_CUR (%d/%d/%d => %d)\n",
+ cval->head.id, mixer_ctrl_intf(cval->head.mixer),
+ cval->min, cval->max, cval->res, saved);
- return -ENODEV;
+ cval->get_cur_broken = 1;
+ return -ENXIO;
}
/*
@@ -1385,8 +1380,6 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
goto no_checks;
ret = check_sticky_volume_control(cval, minchn, saved);
- if (ret == -ENODEV)
- goto sticky;
if (ret)
goto no_checks;
@@ -1454,34 +1447,15 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
}
}
- return 0;
-
-sticky:
/*
- * It makes no sense to restore the saved value for a sticky mixer,
- * since setting any value is a no-op.
- *
- * However, in some rare cases, SET_CUR is effective despite GET_CUR
- * always returns a constant value. These mixers are not sticky, but
- * there's no way to distinguish them. Without any additional
- * information, the best thing we can do is to set the mixer value to
- * the maximum before bailing out, so that a soft mixer can still reach
- * the maximum hardware volume if the mixer turns out to be non-sticky.
- * Meanwhile, all channels must be synchronized to prevent imbalance
- * volume.
+ * When GET_CUR is sticky, the saved value is bogus, so mixer values set
+ * by the sanity checks must be discarded through init_cur_mix_raw().
+ * After that, we can clear the flag as per QUIRK_FLAG_MIXER_GET_CUR_OK.
*/
- if (!cval->cmask) {
- snd_usb_set_cur_mix_value(cval, 0, 0, cval->max);
- } else {
- idx = 0;
- for (i = 0; i < MAX_CHANNELS; i++) {
- if (cval->cmask & BIT(i)) {
- snd_usb_set_cur_mix_value(cval, i + 1, idx, cval->max);
- idx++;
- }
- }
- }
- return ret;
+ if (cval->head.mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_GET_CUR_OK)
+ cval->get_cur_broken = 0;
+
+ return 0;
}
#define get_min_max(cval, def) get_min_max_with_quirks(cval, def, NULL)
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -505,13 +505,13 @@ static const struct usbmix_connector_map gigabyte_b450_connector_map[] = {
{}
};
-/* Audient iD14: FU 12 advertises Volume on only 4 of its 6 logical channels
- * and sits on the monitor mixer branch, but it is traced through to the
- * Speaker output terminal and gets named "Speaker Playback Volume". Userspace
- * then adopts it as the stream's hardware volume, and any setting below 0 dB
- * attenuates some channels but not others (20 dB imbalance at 80%). Give it a
- * non-standard name so that it is no longer taken for the stream's master
- * volume, while remaining reachable for anyone who wants the monitor gain.
+/* Audient iD14 MkI and MkII: FU 12 sits on the monitor mixer branch but is
+ * traced through to the Speaker output terminal, so it is named "Speaker
+ * Playback Volume". On MkII it controls only 4 of 6 playback channels. MkI
+ * testing found asymmetric attenuation within the main stereo pair. Userspace
+ * adopts this control as the stream's hardware volume, causing imbalance below
+ * 0 dB. Give it a non-standard name so that userspace no longer treats it as
+ * the stream master, while keeping the monitor gain reachable.
*/
static const struct usbmix_name_map audient_id14_map[] = {
{ 12, "Monitor Mix Playback" }, /* FU, partial coverage */
@@ -602,7 +602,12 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
.map = maya44_map,
},
{
- /* Audient iD14 */
+ /* Audient iD14 MkI */
+ .id = USB_ID(0x2708, 0x0002),
+ .map = audient_id14_map,
+ },
+ {
+ /* Audient iD14 MkII */
.id = USB_ID(0x2708, 0x0008),
.map = audient_id14_map,
},
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1814,6 +1814,28 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
+{
+ /*
+ * M-Audio Venom
+ *
+ * The AudioControl interface times out on every GET_CUR request,
+ * which adds around 47 seconds to the card registration and
+ * freezes the device, blocking streaming.
+ * Using an explicit composite quirk to skip the mixer entirely.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2084),
+ QUIRK_DRIVER_INFO {
+ .vendor_name = "M-Audio",
+ .product_name = "Venom",
+ QUIRK_DATA_COMPOSITE {
+ { QUIRK_DATA_IGNORE(0) },
+ { QUIRK_DATA_STANDARD_AUDIO(1) },
+ { QUIRK_DATA_STANDARD_AUDIO(2) },
+ { QUIRK_DATA_STANDARD_MIDI(3) },
+ QUIRK_COMPOSITE_END
+ }
+ }
+},
/* Casio devices */
{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -2215,10 +2215,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY),
DEVICE_FLG(0x03f0, 0x654a, /* HP 320 FHD Webcam */
QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16),
- DEVICE_FLG(0x040b, 0x0897, /* Weltrend Semiconductor, sold as Redragon H510-PRO Wireless headset */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN),
DEVICE_FLG(0x041e, 0x3000, /* Creative SB Extigy */
QUIRK_FLAG_IGNORE_CTL_ERROR),
+ DEVICE_FLG(0x041e, 0x324d, /* Creative Sound Blaster Play! 3 */
+ QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE),
DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */
QUIRK_FLAG_GET_SAMPLE_RATE),
DEVICE_FLG(0x045e, 0x083c, /* MS USB Link headset */
@@ -2256,8 +2256,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
DEVICE_FLG(0x046d, 0x0a8f, /* Logitech H390 headset */
QUIRK_FLAG_CTL_MSG_DELAY_1M |
QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE),
- DEVICE_FLG(0x046d, 0x0af7, /* Logitech PRO X 2 LIGHTSPEED */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN),
+ DEVICE_FLG(0x046d, 0x0aba, /* Logitech PRO X Wireless */
+ QUIRK_FLAG_MIXER_GET_CUR_OK |
+ QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE),
DEVICE_FLG(0x0499, 0x1506, /* Yamaha THR5 */
QUIRK_FLAG_GENERIC_IMPLICIT_FB),
DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */
@@ -2329,9 +2330,11 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */
QUIRK_FLAG_GENERIC_IMPLICIT_FB),
DEVICE_FLG(0x0763, 0x2080, /* M-Audio Fast Track Ultra */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN | QUIRK_FLAG_GENERIC_IMPLICIT_FB),
+ QUIRK_FLAG_GENERIC_IMPLICIT_FB),
DEVICE_FLG(0x0763, 0x2081, /* M-Audio Fast Track Ultra */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN | QUIRK_FLAG_GENERIC_IMPLICIT_FB),
+ QUIRK_FLAG_GENERIC_IMPLICIT_FB),
+ DEVICE_FLG(0x0763, 0x2084, /* M-Audio Venom */
+ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_DISABLE_AUTOSUSPEND),
DEVICE_FLG(0x07fd, 0x000b, /* MOTU M Series 2nd hardware revision */
QUIRK_FLAG_CTL_MSG_DELAY_1M),
DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */
@@ -2370,8 +2373,6 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
DEVICE_FLG(0x1038, 0x1294, /* SteelSeries Arctis Pro Wireless */
QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE),
- DEVICE_FLG(0x1038, 0x2232, /* SteelSeries Arctis Nova 5 */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN),
DEVICE_FLG(0x1101, 0x0003, /* Audioengine D1 */
QUIRK_FLAG_GET_SAMPLE_RATE),
DEVICE_FLG(0x12d1, 0x3a07, /* HUAWEI USB-C HEADSET */
@@ -2379,8 +2380,6 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY),
DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */
QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16),
- DEVICE_FLG(0x1377, 0x6004, /* Sennheiser MOMENTUM 3 */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN),
DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */
QUIRK_FLAG_GET_SAMPLE_RATE),
DEVICE_FLG(0x1397, 0x0507, /* Behringer UMC202HD */
@@ -2393,6 +2392,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
QUIRK_FLAG_IFB_SILENCE_ON_EMPTY),
DEVICE_FLG(0x13e5, 0x0001, /* Serato Phono */
QUIRK_FLAG_IGNORE_CTL_ERROR),
+ DEVICE_FLG(0x152a, 0x85dd, /* SMSL USB DAC */
+ QUIRK_FLAG_DSD_RAW | QUIRK_FLAG_DISABLE_AUTOSUSPEND |
+ QUIRK_FLAG_SKIP_IFACE_SETUP),
DEVICE_FLG(0x152a, 0x880a, /* NeuralDSP Quad Cortex */
0), /* Doesn't have the vendor quirk which would otherwise apply */
DEVICE_FLG(0x1532, 0x055e, /* Razer Nommo V2 X */
@@ -2505,8 +2507,6 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
QUIRK_FLAG_CTL_MSG_DELAY_1M),
DEVICE_FLG(0x2d99, 0x0026, /* HECATE G2 GAMING HEADSET */
QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE),
- DEVICE_FLG(0x2d99, 0xa024, /* Edifier MF200 */
- QUIRK_FLAG_MIXER_GET_CUR_BROKEN),
DEVICE_FLG(0x2fc6, 0xf06b, /* MOONDROP Moonriver2 Ti */
QUIRK_FLAG_CTL_MSG_DELAY),
DEVICE_FLG(0x2fc6, 0xf0b5, /* iBasso DC-Elite */
@@ -2647,7 +2647,7 @@ static const char *const snd_usb_audio_quirk_flag_names[] = {
QUIRK_STRING_ENTRY(MIXER_PLAYBACK_LINEAR_VOL),
QUIRK_STRING_ENTRY(MIXER_CAPTURE_LINEAR_VOL),
QUIRK_STRING_ENTRY(IFB_SILENCE_ON_EMPTY),
- QUIRK_STRING_ENTRY(MIXER_GET_CUR_BROKEN),
+ QUIRK_STRING_ENTRY(MIXER_GET_CUR_OK),
QUIRK_STRING_ENTRY(PLAYBACK_URB_FIXUP),
QUIRK_STRING_ENTRY(ALWAYS_SET_RATE),
NULL
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -243,24 +243,18 @@ extern bool snd_usb_skip_validation;
* from snd_usb_handle_sync_urb. Instead fall through and enqueue a
* packet_info containing only size-0 packets, so the OUT ring keeps
* moving (emits silence). Needed by Behringer Flow 8 (1397:050c).
- * QUIRK_FLAG_MIXER_GET_CUR_BROKEN
- * Some mixers are sticky, which means that setting their current volume is a
- * no-op, and reading the current volume returns a constant value. The sticky
- * check disables these mixers to prevent confusing userspace. However, some
- * devices do have a tunable volume despite the reported current volume being
- * constant. As the sticky check can't distinguish between the two categories,
- * setting this flag tells that the device should fall into the second
- * category when GET_CUR returns a constant value, resulting in the sticky
- * check being non-fatal and only disabling GET_CUR instead of the whole mixer.
- * The current volume will then be provided by the internal cache that stores
- * the last set volume
+ * QUIRK_FLAG_MIXER_GET_CUR_OK
+ * On some devices, whether their GET_CUR being sticky depends on whether
+ * hotpluggable components are present. When the hotpluggable components are
+ * missing on probe, their GET_CUR behavior is classified as broken. Set the
+ * flag to prevent the heuristics from gating GET_CUR.
* QUIRK_FLAG_PLAYBACK_URB_FIXUP
* Set URB_ISO_ASAP flag for isochronous URBs and force nurbs to MAX_URBS.
* This is needed for devices that exhibit boot-time audio stuttering due
* to insufficient buffer depth combined with xHCI scheduling variability.
* The larger buffer (MAX_URBS = 12, ~64ms) absorbs system scheduling
* jitter during boot, while URB_ISO_ASAP ensures consistent xHCI scheduling.
- * QUIRK_FLAG_ALWAYS_SET_RATE:
+ * QUIRK_FLAG_ALWAYS_SET_RATE
* Issue SET_CUR for the sample rate even when the clock already reports the
* requested rate. A device advertising a single rate is otherwise never sent
* the request at all, and some require it before streaming will start.
@@ -297,10 +291,10 @@ enum {
QUIRK_TYPE_MIXER_PLAYBACK_LINEAR_VOL = 27,
QUIRK_TYPE_MIXER_CAPTURE_LINEAR_VOL = 28,
QUIRK_TYPE_IFB_SILENCE_ON_EMPTY = 29,
- QUIRK_TYPE_MIXER_GET_CUR_BROKEN = 30,
+ QUIRK_TYPE_MIXER_GET_CUR_OK = 30,
QUIRK_TYPE_PLAYBACK_URB_FIXUP = 31,
QUIRK_TYPE_ALWAYS_SET_RATE = 32,
-/* Please also edit snd_usb_audio_quirk_flag_names */
+/* Please also edit snd_usb_audio_quirk_flag_names and alsa-configuration.rst */
};
#define QUIRK_FLAG(x) BIT_U64(QUIRK_TYPE_ ## x)
@@ -335,7 +329,7 @@ enum {
#define QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL QUIRK_FLAG(MIXER_PLAYBACK_LINEAR_VOL)
#define QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL QUIRK_FLAG(MIXER_CAPTURE_LINEAR_VOL)
#define QUIRK_FLAG_IFB_SILENCE_ON_EMPTY QUIRK_FLAG(IFB_SILENCE_ON_EMPTY)
-#define QUIRK_FLAG_MIXER_GET_CUR_BROKEN QUIRK_FLAG(MIXER_GET_CUR_BROKEN)
+#define QUIRK_FLAG_MIXER_GET_CUR_OK QUIRK_FLAG(MIXER_GET_CUR_OK)
#define QUIRK_FLAG_PLAYBACK_URB_FIXUP QUIRK_FLAG(PLAYBACK_URB_FIXUP)
#define QUIRK_FLAG_ALWAYS_SET_RATE QUIRK_FLAG(ALWAYS_SET_RATE)
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