diff options
Diffstat (limited to 'sound')
45 files changed, 871 insertions, 139 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 67cf6a0e17ba..5a64453da728 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2144,6 +2144,10 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, for (;;) { long tout; struct snd_pcm_runtime *to_check; + unsigned int drain_rate; + snd_pcm_uframes_t drain_bufsz; + bool drain_no_period_wakeup; + if (signal_pending(current)) { result = -ERESTARTSYS; break; @@ -2163,16 +2167,25 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_group_unref(group, substream); if (!to_check) break; /* all drained */ + /* + * Cache the runtime fields needed after unlock. + * A concurrent close() on the linked stream may free + * its runtime via snd_pcm_detach_substream() once we + * release the stream lock below. + */ + drain_no_period_wakeup = to_check->no_period_wakeup; + drain_rate = to_check->rate; + drain_bufsz = to_check->buffer_size; init_waitqueue_entry(&wait, current); set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); - if (runtime->no_period_wakeup) + if (drain_no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { tout = 100; - if (runtime->rate) { - long t = runtime->buffer_size * 1100 / runtime->rate; + if (drain_rate) { + long t = drain_bufsz * 1100 / drain_rate; tout = max(t, tout); } tout = msecs_to_jiffies(tout); diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 85d265c7d544..f7a50bae4b55 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -122,7 +122,7 @@ static void dice_card_strings(struct snd_dice *dice) fw_csr_string(dev->config_rom + 5, CSR_VENDOR, vendor, sizeof(vendor)); strscpy(model, "?"); fw_csr_string(dice->unit->directory, CSR_MODEL, model, sizeof(model)); - snprintf(card->longname, sizeof(card->longname), + scnprintf(card->longname, sizeof(card->longname), "%s %s (serial %u) at %s, S%d", vendor, model, dev->config_rom[4] & 0x3fffff, dev_name(&dice->unit->device), 100 << dev->max_speed); diff --git a/sound/hda/codecs/ca0132.c b/sound/hda/codecs/ca0132.c index bf342a76807c..a0677d7da8e2 100644 --- a/sound/hda/codecs/ca0132.c +++ b/sound/hda/codecs/ca0132.c @@ -9816,6 +9816,15 @@ static void ca0132_config(struct hda_codec *codec) spec->dig_in = 0x09; break; } + + /* Default HP/Speaker auto-detect from headphone pin verb: enable if the + * pin config indicates presence detect (not AC_DEFCFG_MISC_NO_PRESENCE). + */ + if (spec->unsol_tag_hp && + (snd_hda_query_pin_caps(codec, spec->unsol_tag_hp) & AC_PINCAP_PRES_DETECT) && + !(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, spec->unsol_tag_hp)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID] = 1; } static int ca0132_prepare_verbs(struct hda_codec *codec) diff --git a/sound/hda/codecs/hdmi/tegrahdmi.c b/sound/hda/codecs/hdmi/tegrahdmi.c index 5f6fe31aa202..ebb6410a4831 100644 --- a/sound/hda/codecs/hdmi/tegrahdmi.c +++ b/sound/hda/codecs/hdmi/tegrahdmi.c @@ -299,6 +299,7 @@ static const struct hda_device_id snd_hda_id_tegrahdmi[] = { HDA_CODEC_ID_MODEL(0x10de002f, "Tegra194 HDMI/DP2", MODEL_TEGRA), HDA_CODEC_ID_MODEL(0x10de0030, "Tegra194 HDMI/DP3", MODEL_TEGRA), HDA_CODEC_ID_MODEL(0x10de0031, "Tegra234 HDMI/DP", MODEL_TEGRA234), + HDA_CODEC_ID_MODEL(0x10de0032, "Tegra238 HDMI/DP", MODEL_TEGRA234), HDA_CODEC_ID_MODEL(0x10de0033, "SoC 33 HDMI/DP", MODEL_TEGRA234), HDA_CODEC_ID_MODEL(0x10de0034, "Tegra264 HDMI/DP", MODEL_TEGRA234), HDA_CODEC_ID_MODEL(0x10de0035, "SoC 35 HDMI/DP", MODEL_TEGRA234), diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 36053042ca77..ab4b22fcb72e 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -1017,6 +1017,24 @@ static int alc269_resume(struct hda_codec *codec) return 0; } +#define STARLABS_STARFIGHTER_SHUTUP_DELAY_MS 30 + +static void starlabs_starfighter_shutup(struct hda_codec *codec) +{ + if (snd_hda_gen_shutup_speakers(codec)) + msleep(STARLABS_STARFIGHTER_SHUTUP_DELAY_MS); +} + +static void alc233_fixup_starlabs_starfighter(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->shutup = starlabs_starfighter_shutup; +} + static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4040,6 +4058,7 @@ enum { ALC245_FIXUP_CLEVO_NOISY_MIC, ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE, ALC233_FIXUP_MEDION_MTL_SPK, + ALC233_FIXUP_STARLABS_STARFIGHTER, ALC294_FIXUP_BASS_SPEAKER_15, ALC283_FIXUP_DELL_HP_RESUME, ALC294_FIXUP_ASUS_CS35L41_SPI_2, @@ -4056,6 +4075,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO, ALC233_FIXUP_LENOVO_GPIO2_MIC_HOTKEY, ALC245_FIXUP_BASS_HP_DAC, + ALC245_FIXUP_ACER_MICMUTE_LED, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -6499,6 +6519,10 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC233_FIXUP_STARLABS_STARFIGHTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_starlabs_starfighter, + }, [ALC294_FIXUP_BASS_SPEAKER_15] = { .type = HDA_FIXUP_FUNC, .v.func = alc294_fixup_bass_speaker_15, @@ -6576,6 +6600,12 @@ static const struct hda_fixup alc269_fixups[] = { /* Borrow the DAC routing selected for those Thinkpads */ .v.func = alc285_fixup_thinkpad_x1_gen7, }, + [ALC245_FIXUP_ACER_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_coef_micmute_led, + .chained = true, + .chain_id = ALC2XX_FIXUP_HEADSET_MIC, + } }; static const struct hda_quirk alc269_fixup_tbl[] = { @@ -6591,6 +6621,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF), + SND_PCI_QUIRK(0x1025, 0x0943, "Acer Aspire V3-572G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x100c, "Acer Aspire E5-574G", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), @@ -6627,6 +6658,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1597, "Acer Nitro 5 AN517-55", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), + SND_PCI_QUIRK(0x1025, 0x171e, "Acer Nitro ANV15-51", ALC245_FIXUP_ACER_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1826, "Acer Helios ZPC", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), SND_PCI_QUIRK(0x1025, 0x182c, "Acer Helios ZPD", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), SND_PCI_QUIRK(0x1025, 0x1844, "Acer Helios ZPS", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), @@ -6872,6 +6904,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88b3, "HP ENVY x360 Convertible 15-es0xxx", ALC245_FIXUP_HP_ENVY_X360_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x88d1, "HP Pavilion 15-eh1xxx (mainboard 88D1)", ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT), SND_PCI_QUIRK(0x103c, 0x88dd, "HP Pavilion 15z-ec200", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x88eb, "HP Victus 16-e0xxx", ALC245_FIXUP_HP_MUTE_LED_V2_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED), @@ -6907,6 +6940,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89da, "HP Spectre x360 14t-ea100", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x89e7, "HP Elite x2 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8a1f, "HP Laptop 14s-dr5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a26, "HP Victus 16-d1xxx (MB 8A26)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), @@ -7240,6 +7274,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e93, "ASUS ExpertBook B9403CVAR", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RCLA72", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), + HDA_CODEC_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), @@ -7311,7 +7346,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), - SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc188, "Samsung Galaxy Book Flex (NT950QCT-A38A)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Book Flex (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc1a3, "Samsung Galaxy Book Pro (NP935XDB-KC1SE)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc1a4, "Samsung Galaxy Book Pro 360 (NT935QBD)", ALC298_FIXUP_SAMSUNG_AMP), @@ -7459,6 +7495,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2288, "Thinkpad X390", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), @@ -7651,6 +7688,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x2782, 0x1705, "MEDION E15433", ALC269VC_FIXUP_INFINIX_Y4_MAX), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x2782, 0x4900, "MEDION E15443", ALC233_FIXUP_MEDION_MTL_SPK), + SND_PCI_QUIRK(0x7017, 0x2014, "Star Labs StarFighter", ALC233_FIXUP_STARLABS_STARFIGHTER), 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), @@ -7747,6 +7785,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"}, {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, + {.id = ALC233_FIXUP_STARLABS_STARFIGHTER, .name = "starlabs-starfighter"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, {.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"}, {.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"}, diff --git a/sound/hda/codecs/realtek/alc662.c b/sound/hda/codecs/realtek/alc662.c index 5073165d1f3c..3a943adf9087 100644 --- a/sound/hda/codecs/realtek/alc662.c +++ b/sound/hda/codecs/realtek/alc662.c @@ -313,6 +313,7 @@ enum { ALC897_FIXUP_HEADSET_MIC_PIN2, ALC897_FIXUP_UNIS_H3C_X500S, ALC897_FIXUP_HEADSET_MIC_PIN3, + ALC897_FIXUP_H610M_HP_PIN, }; static const struct hda_fixup alc662_fixups[] = { @@ -766,6 +767,13 @@ static const struct hda_fixup alc662_fixups[] = { { } }, }, + [ALC897_FIXUP_H610M_HP_PIN] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x0321403f }, /* HP out */ + { } + }, + }, }; static const struct hda_quirk alc662_fixup_tbl[] = { @@ -815,6 +823,7 @@ static const struct hda_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), + SND_PCI_QUIRK(0x1458, 0xa194, "H610M H V2 DDR4", ALC897_FIXUP_H610M_HP_PIN), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN), diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 3a50d4b3a064..6239a25bb8f3 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -19,15 +19,13 @@ #include "hda_jack.h" #include "generic.h" -/* GPIO node ID */ -#define SENARY_GPIO_NODE 0x01 - struct senary_spec { struct hda_gen_spec gen; /* extra EAPD pins */ unsigned int num_eapds; hda_nid_t eapds[4]; + bool dynamic_eapd; hda_nid_t mute_led_eapd; unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ @@ -123,19 +121,23 @@ static void senary_init_gpio_led(struct hda_codec *codec) unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask; if (mask) { - snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_MASK, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_MASK, mask); - snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DIRECTION, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DIRECTION, mask); - snd_hda_codec_write(codec, SENARY_GPIO_NODE, 0, AC_VERB_SET_GPIO_DATA, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); } } static int senary_init(struct hda_codec *codec) { + struct senary_spec *spec = codec->spec; + snd_hda_gen_init(codec); senary_init_gpio_led(codec); + if (!spec->dynamic_eapd) + senary_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); return 0; diff --git a/sound/hda/codecs/side-codecs/cs35l56_hda.c b/sound/hda/codecs/side-codecs/cs35l56_hda.c index cfc8de2ae499..1ace4beef508 100644 --- a/sound/hda/codecs/side-codecs/cs35l56_hda.c +++ b/sound/hda/codecs/side-codecs/cs35l56_hda.c @@ -249,7 +249,7 @@ static int cs35l56_hda_posture_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct cs35l56_hda *cs35l56 = snd_kcontrol_chip(kcontrol); - unsigned long pos = ucontrol->value.integer.value[0]; + long pos = ucontrol->value.integer.value[0]; bool changed; int ret; @@ -403,10 +403,6 @@ static void cs35l56_hda_remove_controls(struct cs35l56_hda *cs35l56) snd_ctl_remove(cs35l56->codec->card, cs35l56->volume_ctl); } -static const struct cs_dsp_client_ops cs35l56_hda_client_ops = { - /* cs_dsp requires the client to provide this even if it is empty */ -}; - static int cs35l56_hda_request_firmware_file(struct cs35l56_hda *cs35l56, const struct firmware **firmware, char **filename, const char *base_name, const char *system_name, @@ -1149,7 +1145,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) cs35l56->base.cal_index = cs35l56->index; cs35l56_init_cs_dsp(&cs35l56->base, &cs35l56->cs_dsp); - cs35l56->cs_dsp.client_ops = &cs35l56_hda_client_ops; if (cs35l56->base.reset_gpio) { dev_dbg(cs35l56->base.dev, "Hard reset\n"); diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 74c3cf1e45e1..67240ce184e1 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -60,7 +60,6 @@ struct tas2781_hda_i2c_priv { int (*save_calibration)(struct tas2781_hda *h); int hda_chip_id; - bool skip_calibration; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -479,8 +478,7 @@ static void tasdevice_dspfw_init(void *context) /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - if (!hda_priv->skip_calibration) - hda_priv->save_calibration(tas_hda); + hda_priv->save_calibration(tas_hda); } static void tasdev_fw_ready(const struct firmware *fmw, void *context) @@ -535,7 +533,6 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, void *master_data) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; struct hda_component_parent *parent = master_data; struct hda_component *comp; struct hda_codec *codec; @@ -564,14 +561,6 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, break; } - /* - * Using ASUS ROG Xbox Ally X (RC73XA) UEFI calibration data - * causes audio dropouts during playback, use fallback data - * from DSP firmware as a workaround. - */ - if (codec->core.subsystem_id == 0x10431384) - hda_priv->skip_calibration = true; - guard(pm_runtime_active_auto)(dev); comp->dev = dev; @@ -643,6 +632,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) */ device_name = "TIAS2781"; hda_priv->hda_chip_id = HDA_TAS2781; + tas_hda->priv->chip_id = TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2770")) { @@ -656,6 +646,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; hda_priv->hda_chip_id = HDA_TAS2781; + tas_hda->priv->chip_id = TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { diff --git a/sound/hda/controllers/intel.c b/sound/hda/controllers/intel.c index 6fddf400c4a3..3f434994c18d 100644 --- a/sound/hda/controllers/intel.c +++ b/sound/hda/controllers/intel.c @@ -1751,6 +1751,8 @@ static int default_bdl_pos_adj(struct azx *chip) return 1; case AZX_DRIVER_ZHAOXINHDMI: return 128; + case AZX_DRIVER_NVIDIA: + return 64; default: return 32; } diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 4d99472c75ba..09f6c9a2c041 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -127,8 +127,13 @@ static int acp_card_rt5682_init(struct snd_soc_pcm_runtime *rtd) if (drvdata->hs_codec_id != RT5682) return -EINVAL; - drvdata->wclk = clk_get(component->dev, "rt5682-dai-wclk"); - drvdata->bclk = clk_get(component->dev, "rt5682-dai-bclk"); + drvdata->wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(drvdata->wclk)) + return PTR_ERR(drvdata->wclk); + + drvdata->bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(drvdata->bclk)) + return PTR_ERR(drvdata->bclk); ret = snd_soc_dapm_new_controls(dapm, rt5682_widgets, ARRAY_SIZE(rt5682_widgets)); @@ -370,8 +375,13 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) return -EINVAL; if (!drvdata->soc_mclk) { - drvdata->wclk = clk_get(component->dev, "rt5682-dai-wclk"); - drvdata->bclk = clk_get(component->dev, "rt5682-dai-bclk"); + drvdata->wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(drvdata->wclk)) + return PTR_ERR(drvdata->wclk); + + drvdata->bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(drvdata->bclk)) + return PTR_ERR(drvdata->bclk); } ret = snd_soc_dapm_new_controls(dapm, rt5682s_widgets, diff --git a/sound/soc/amd/acp/amd-acp63-acpi-match.c b/sound/soc/amd/acp/amd-acp63-acpi-match.c index 9b6a49c051cd..1dbbaba3c75b 100644 --- a/sound/soc/amd/acp/amd-acp63-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp63-acpi-match.c @@ -30,6 +30,20 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1 }; +static const struct snd_soc_acpi_endpoint spk_2_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 2, + .group_id = 1 +}; + +static const struct snd_soc_acpi_endpoint spk_3_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 3, + .group_id = 1 +}; + static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = { { .adr = 0x000030025D071101ull, @@ -103,6 +117,345 @@ static const struct snd_soc_acpi_adr_device rt722_0_single_adr[] = { } }; +static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { + { /* Jack Playback Endpoint */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Jack Capture Endpoint */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Speaker Playback Endpoint */ + .num = 3, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l56x4_l1u3210_adr[] = { + { + .adr = 0x00013301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00013201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, + { + .adr = 0x00013101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00013001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l63x2_l0u01_adr[] = { + { + .adr = 0x00003001FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00003101FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l63x2_l1u01_adr[] = { + { + .adr = 0x00013001FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00013101FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l63x2_l1u13_adr[] = { + { + .adr = 0x00013101FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00013301FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l63x4_l0u0246_adr[] = { + { + .adr = 0x00003001FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00003201FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, + { + .adr = 0x00003401FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00003601FA356301ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + }, +}; + +static const struct snd_soc_acpi_adr_device cs42l43_l0u0_adr[] = { + { + .adr = 0x00003001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + +static const struct snd_soc_acpi_adr_device cs42l43_l0u1_adr[] = { + { + .adr = 0x00003101FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + +static const struct snd_soc_acpi_adr_device cs42l43b_l0u1_adr[] = { + { + .adr = 0x00003101FA2A3B01ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + +static const struct snd_soc_acpi_adr_device cs42l43_l1u0_cs35l56x4_l1u0123_adr[] = { + { + .adr = 0x00013001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + }, + { + .adr = 0x00013001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00013101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, + { + .adr = 0x00013201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00013301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + }, +}; + +static const struct snd_soc_acpi_adr_device cs42l45_l0u0_adr[] = { + { + .adr = 0x00003001FA424501ull, + /* Re-use endpoints, but cs42l45 has no speaker */ + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints) - 1, + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l45" + } +}; + +static const struct snd_soc_acpi_adr_device cs42l45_l1u0_adr[] = { + { + .adr = 0x00013001FA424501ull, + /* Re-use endpoints, but cs42l45 has no speaker */ + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints) - 1, + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l45" + } +}; + +static const struct snd_soc_acpi_link_adr acp63_cs35l56x4_l1u3210[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l56x4_l1u3210_adr), + .adr_d = cs35l56x4_l1u3210_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs35l63x4_l0u0246[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs35l63x4_l0u0246_adr), + .adr_d = cs35l63x4_l0u0246_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l43_l0u1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_l0u1_adr), + .adr_d = cs42l43_l0u1_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l43b_l0u1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43b_l0u1_adr), + .adr_d = cs42l43b_l0u1_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l43_l0u0_cs35l56x4_l1u3210[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_l0u0_adr), + .adr_d = cs42l43_l0u0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l56x4_l1u3210_adr), + .adr_d = cs35l56x4_l1u3210_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l43_l1u0_cs35l56x4_l1u0123[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs42l43_l1u0_cs35l56x4_l1u0123_adr), + .adr_d = cs42l43_l1u0_cs35l56x4_l1u0123_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l45_l0u0[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l45_l0u0_adr), + .adr_d = cs42l45_l0u0_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l45_l0u0_cs35l63x2_l1u01[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l45_l0u0_adr), + .adr_d = cs42l45_l0u0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l63x2_l1u01_adr), + .adr_d = cs35l63x2_l1u01_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l45_l0u0_cs35l63x2_l1u13[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l45_l0u0_adr), + .adr_d = cs42l45_l0u0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l63x2_l1u13_adr), + .adr_d = cs35l63x2_l1u13_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l45_l1u0[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs42l45_l1u0_adr), + .adr_d = cs42l45_l1u0_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l45_l1u0_cs35l63x2_l0u01[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs42l45_l1u0_adr), + .adr_d = cs42l45_l1u0_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs35l63x2_l0u01_adr), + .adr_d = cs35l63x2_l0u01_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr acp63_cs42l45_l1u0_cs35l63x4_l0u0246[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs42l45_l1u0_adr), + .adr_d = cs42l45_l1u0_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs35l63x4_l0u0246_adr), + .adr_d = cs35l63x4_l0u0246_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr acp63_rt722_only[] = { { .mask = BIT(0), @@ -135,6 +488,66 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sdw_machines[] = { .links = acp63_4_in_1_sdca, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_cs42l43_l0u0_cs35l56x4_l1u3210, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_cs42l45_l1u0_cs35l63x4_l0u0246, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_cs42l45_l0u0_cs35l63x2_l1u01, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_cs42l45_l0u0_cs35l63x2_l1u13, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_cs42l45_l1u0_cs35l63x2_l0u01, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(1), + .links = acp63_cs42l43_l1u0_cs35l56x4_l1u0123, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(1), + .links = acp63_cs35l56x4_l1u3210, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0), + .links = acp63_cs35l63x4_l0u0246, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0), + .links = acp63_cs42l43_l0u1, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0), + .links = acp63_cs42l43b_l0u1, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(0), + .links = acp63_cs42l45_l0u0, + .drv_name = "amd_sdw", + }, + { + .link_mask = BIT(1), + .links = acp63_cs42l45_l1u0, + .drv_name = "amd_sdw", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp63_sdw_machines); diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 4ca1978020a9..d1eb6f12a183 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -94,8 +94,13 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) return ret; } - rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk"); - rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); + rt5682_dai_wclk = devm_clk_get(component->dev, "rt5682-dai-wclk"); + if (IS_ERR(rt5682_dai_wclk)) + return PTR_ERR(rt5682_dai_wclk); + + rt5682_dai_bclk = devm_clk_get(component->dev, "rt5682-dai-bclk"); + if (IS_ERR(rt5682_dai_bclk)) + return PTR_ERR(rt5682_dai_bclk); ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index f1a63475100d..1324543b42d7 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -703,6 +703,20 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Vivobook_ASUSLaptop M6501RR_M6501RR"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "ASUS EXPERTBOOK BM1503CDA"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_BOARD_NAME, "PM1503CDA"), + } + }, {} }; diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 4707f28bfca2..af87ebae98cb 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -26,7 +26,7 @@ #include "cs35l56.h" -static const struct reg_sequence cs35l56_patch[] = { +static const struct reg_sequence cs35l56_asp_patch[] = { /* * Firmware can change these to non-defaults to satisfy SDCA. * Ensure that they are at known defaults. @@ -43,6 +43,20 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_ASP1TX2_INPUT, 0x00000000 }, { CS35L56_ASP1TX3_INPUT, 0x00000000 }, { CS35L56_ASP1TX4_INPUT, 0x00000000 }, +}; + +int cs35l56_set_asp_patch(struct cs35l56_base *cs35l56_base) +{ + return regmap_register_patch(cs35l56_base->regmap, cs35l56_asp_patch, + ARRAY_SIZE(cs35l56_asp_patch)); +} +EXPORT_SYMBOL_NS_GPL(cs35l56_set_asp_patch, "SND_SOC_CS35L56_SHARED"); + +static const struct reg_sequence cs35l56_patch[] = { + /* + * Firmware can change these to non-defaults to satisfy SDCA. + * Ensure that they are at known defaults. + */ { CS35L56_SWIRE_DP3_CH1_INPUT, 0x00000018 }, { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2ff8b172b76e..37909a319f88 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -348,6 +348,13 @@ static int cs35l56_dsp_event(struct snd_soc_dapm_widget *w, return wm_adsp_event(w, kcontrol, event); } +static int cs35l56_asp_dai_probe(struct snd_soc_dai *codec_dai) +{ + struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(codec_dai->component); + + return cs35l56_set_asp_patch(&cs35l56->base); +} + static int cs35l56_asp_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(codec_dai->component); @@ -552,6 +559,7 @@ static int cs35l56_asp_dai_set_sysclk(struct snd_soc_dai *dai, } static const struct snd_soc_dai_ops cs35l56_ops = { + .probe = cs35l56_asp_dai_probe, .set_fmt = cs35l56_asp_dai_set_fmt, .set_tdm_slot = cs35l56_asp_dai_set_tdm_slot, .hw_params = cs35l56_asp_dai_hw_params, @@ -1617,9 +1625,9 @@ static int cs35l56_process_xu_onchip_speaker_id(struct cs35l56_private *cs35l56, if (num_pulls < 0) return num_pulls; - if (num_pulls != num_gpios) { + if (num_pulls && (num_pulls != num_gpios)) { dev_warn(cs35l56->base.dev, "%s count(%d) != %s count(%d)\n", - pull_name, num_pulls, gpio_name, num_gpios); + pull_name, num_pulls, gpio_name, num_gpios); } ret = cs35l56_check_and_save_onchip_spkid_gpios(&cs35l56->base, diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index b83bc4de1301..3e04e6897b14 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -699,6 +699,7 @@ static int cs42l43_run_type_detect(struct cs42l43_codec *priv) switch (type & CS42L43_HSDET_TYPE_STS_MASK) { case 0x0: // CTIA case 0x1: // OMTP + case 0x4: return cs42l43_run_load_detect(priv, true); case 0x2: // 3-pole return cs42l43_run_load_detect(priv, false); diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 9f34a6a35487..03f31d9d916e 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1047,7 +1047,7 @@ static int rt1011_recv_spk_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_to_dapm(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_component_to_dapm(component); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c index 50f65662e143..8bb7e8497c72 100644 --- a/sound/soc/codecs/rt1320-sdw.c +++ b/sound/soc/codecs/rt1320-sdw.c @@ -2629,7 +2629,7 @@ static int rt1320_sdw_hw_params(struct snd_pcm_substream *substream, struct sdw_port_config port_config; struct sdw_port_config dmic_port_config[2]; struct sdw_stream_runtime *sdw_stream; - int retval; + int retval, num_channels; unsigned int sampling_rate; dev_dbg(dai->dev, "%s %s", __func__, dai->name); @@ -2661,7 +2661,8 @@ static int rt1320_sdw_hw_params(struct snd_pcm_substream *substream, dmic_port_config[1].num = 10; break; case RT1321_DEV_ID: - dmic_port_config[0].ch_mask = BIT(0) | BIT(1); + num_channels = params_channels(params); + dmic_port_config[0].ch_mask = GENMASK(num_channels - 1, 0); dmic_port_config[0].num = 8; break; default: diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c969eb38704e..5798d518d94c 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -32,6 +32,10 @@ #define TAS2781_YRAM1_PAGE 42 #define TAS2781_YRAM1_START_REG 88 +#define TAS2781_PG_REG TASDEVICE_REG(0x00, 0x00, 0x7c) +#define TAS2781_PG_1_0 0xA0 +#define TAS2781_PG_2_0 0xA8 + #define TAS2781_YRAM2_START_PAGE 43 #define TAS2781_YRAM2_END_PAGE 49 #define TAS2781_YRAM2_START_REG 8 @@ -98,6 +102,12 @@ struct blktyp_devidx_map { unsigned char dev_idx; }; +struct tas2781_cali_specific { + unsigned char sin_gni[4]; + int sin_gni_reg; + bool is_sin_gn_flush; +}; + static const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4, 1, 2 }; @@ -2454,6 +2464,84 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, return ret; } +static int tas2781_cali_preproc(struct tasdevice_priv *priv, int i) +{ + struct tas2781_cali_specific *spec = priv->tasdevice[i].cali_specific; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; + int rc; + + /* + * On TAS2781, if the Speaker calibrated impedance is lower than + * default value hard-coded inside the TAS2781, it will cuase vol + * lower than normal. In order to fix this issue, the parameter of + * SineGainI need updating. + */ + if (spec == NULL) { + int k = i * (cali_data->cali_dat_sz_per_dev + 1); + int re_org, re_cal, corrected_sin_gn, pg_id; + unsigned char r0_deflt[4]; + + spec = devm_kzalloc(priv->dev, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + priv->tasdevice[i].cali_specific = spec; + rc = tasdevice_dev_bulk_read(priv, i, p->r0_reg, r0_deflt, 4); + if (rc < 0) { + dev_err(priv->dev, "invalid RE from %d = %d\n", i, rc); + return rc; + } + /* + * SineGainI need to be re-calculated, calculate the high 16 + * bits. + */ + re_org = r0_deflt[0] << 8 | r0_deflt[1]; + re_cal = data[k + 1] << 8 | data[k + 2]; + if (re_org > re_cal) { + rc = tasdevice_dev_read(priv, i, TAS2781_PG_REG, + &pg_id); + if (rc < 0) { + dev_err(priv->dev, "invalid PG id %d = %d\n", + i, rc); + return rc; + } + + spec->sin_gni_reg = (pg_id == TAS2781_PG_1_0) ? + TASDEVICE_REG(0, 0x1b, 0x34) : + TASDEVICE_REG(0, 0x18, 0x1c); + + rc = tasdevice_dev_bulk_read(priv, i, + spec->sin_gni_reg, + spec->sin_gni, 4); + if (rc < 0) { + dev_err(priv->dev, "wrong sinegaini %d = %d\n", + i, rc); + return rc; + } + corrected_sin_gn = re_org * ((spec->sin_gni[0] << 8) + + spec->sin_gni[1]); + corrected_sin_gn /= re_cal; + spec->sin_gni[0] = corrected_sin_gn >> 8; + spec->sin_gni[1] = corrected_sin_gn & 0xff; + + spec->is_sin_gn_flush = true; + } + } + + if (spec->is_sin_gn_flush) { + rc = tasdevice_dev_bulk_write(priv, i, spec->sin_gni_reg, + spec->sin_gni, 4); + if (rc < 0) { + dev_err(priv->dev, "update failed %d = %d\n", + i, rc); + return rc; + } + } + + return 0; +} + static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { struct calidata *cali_data = &priv->cali_data; @@ -2469,6 +2557,12 @@ static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) } k++; + if (priv->chip_id == TAS2781) { + rc = tas2781_cali_preproc(priv, i); + if (rc < 0) + return; + } + rc = tasdevice_dev_bulk_write(priv, i, p->r0_reg, &(data[k]), 4); if (rc < 0) { dev_err(priv->dev, "chn %d r0_reg bulk_wr err = %d\n", i, rc); diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index e64a0d97afd0..6c56134c60cc 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -52,10 +52,13 @@ static int fsl_easrc_iec958_put_bits(struct snd_kcontrol *kcontrol, struct soc_mreg_control *mc = (struct soc_mreg_control *)kcontrol->private_value; unsigned int regval = ucontrol->value.integer.value[0]; + int ret; + + ret = (easrc_priv->bps_iec958[mc->regbase] != regval); easrc_priv->bps_iec958[mc->regbase] = regval; - return 0; + return ret; } static int fsl_easrc_iec958_get_bits(struct snd_kcontrol *kcontrol, @@ -93,14 +96,17 @@ static int fsl_easrc_set_reg(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct soc_mreg_control *mc = (struct soc_mreg_control *)kcontrol->private_value; + struct fsl_asrc *easrc = snd_soc_component_get_drvdata(component); unsigned int regval = ucontrol->value.integer.value[0]; + bool changed; int ret; - ret = snd_soc_component_write(component, mc->regbase, regval); - if (ret < 0) + ret = regmap_update_bits_check(easrc->regmap, mc->regbase, + GENMASK(31, 0), regval, &changed); + if (ret != 0) return ret; - return 0; + return changed; } #define SOC_SINGLE_REG_RW(xname, xreg) \ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index bdc02e85b089..9e5be0eaa77f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1038,11 +1038,15 @@ int graph_util_is_ports0(struct device_node *np) else port = np; - struct device_node *ports __free(device_node) = of_get_parent(port); - struct device_node *top __free(device_node) = of_get_parent(ports); - struct device_node *ports0 __free(device_node) = of_get_child_by_name(top, "ports"); + struct device_node *ports __free(device_node) = of_get_parent(port); + const char *at = strchr(kbasename(ports->full_name), '@'); - return ports0 == ports; + /* + * Since child iteration order may differ + * between a base DT and DT overlays, + * string match "ports" or "ports@0" in the node name instead. + */ + return !at || !strcmp(at, "@0"); } EXPORT_SYMBOL_GPL(graph_util_is_ports0); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f230991f5f8e..c18ec607e029 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -763,6 +763,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Alienware"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CCD") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, /* Pantherlake devices*/ { .callback = sof_sdw_quirk_cb, diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index de3bdac3e791..168c166c960d 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -838,6 +838,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = { .ack = q6apm_dai_ack, .compress_ops = &q6apm_dai_compress_ops, .use_dai_pcm_id = true, + .remove_order = SND_SOC_COMP_ORDER_EARLY, }; static int q6apm_dai_probe(struct platform_device *pdev) diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index 528756f1332b..5be37eeea329 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -278,6 +278,7 @@ static const struct snd_soc_component_driver q6apm_lpass_dai_component = { .of_xlate_dai_name = q6dsp_audio_ports_of_xlate_dai_name, .be_pcm_base = AUDIOREACH_BE_PCM_BASE, .use_dai_pcm_id = true, + .remove_order = SND_SOC_COMP_ORDER_FIRST, }; static int q6apm_lpass_dai_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 44841fde3856..970b08c89bb3 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -715,6 +715,7 @@ static const struct snd_soc_component_driver q6apm_audio_component = { .name = APM_AUDIO_DRV_NAME, .probe = q6apm_audio_probe, .remove = q6apm_audio_remove, + .remove_order = SND_SOC_COMP_ORDER_LAST, }; static int apm_probe(gpr_device_t *gdev) diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index 95b67bb904c3..e0ed593697ba 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -1156,9 +1156,12 @@ static int find_sdca_entity_iot(struct device *dev, if (!terminal->is_dataport) { const char *type_name = sdca_find_terminal_name(terminal->type); - if (type_name) + if (type_name) { entity->label = devm_kasprintf(dev, GFP_KERNEL, "%s %s", entity->label, type_name); + if (!entity->label) + return -ENOMEM; + } } ret = fwnode_property_read_u32(entity_node, diff --git a/sound/soc/sdca/sdca_interrupts.c b/sound/soc/sdca/sdca_interrupts.c index d9e22cf40f77..95b1ab4ba1b0 100644 --- a/sound/soc/sdca/sdca_interrupts.c +++ b/sound/soc/sdca/sdca_interrupts.c @@ -265,9 +265,9 @@ static int sdca_irq_request_locked(struct device *dev, } /** - * sdca_request_irq - request an individual SDCA interrupt + * sdca_irq_request - request an individual SDCA interrupt * @dev: Pointer to the struct device against which things should be allocated. - * @interrupt_info: Pointer to the interrupt information structure. + * @info: Pointer to the interrupt information structure. * @sdca_irq: SDCA interrupt position. * @name: Name to be given to the IRQ. * @handler: A callback thread function to be called for the IRQ. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d0fffef65daf..573693e21780 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -462,8 +462,7 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) list_del(&rtd->list); - if (delayed_work_pending(&rtd->delayed_work)) - flush_delayed_work(&rtd->delayed_work); + flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); /* @@ -1864,12 +1863,15 @@ static void cleanup_dmi_name(char *name) /* * Check if a DMI field is valid, i.e. not containing any string - * in the black list. + * in the black list and not the empty string. */ static int is_dmi_valid(const char *field) { int i = 0; + if (!field[0]) + return 0; + while (dmi_blacklist[i]) { if (strstr(field, dmi_blacklist[i])) return 0; @@ -2122,6 +2124,9 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) for_each_card_rtds(card, rtd) if (rtd->initialized) snd_soc_link_exit(rtd); + /* flush delayed work before removing DAIs and DAPM widgets */ + snd_soc_flush_all_delayed_work(card); + /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index 94b5ab77649b..ea10e6e8a9fe 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -231,6 +231,15 @@ static const struct tegra_audio_cdata tegra186_data = { .plla_out0_rates[x11_RATE] = 45158400, }; +static const struct tegra_audio_cdata tegra238_data = { + /* PLLA */ + .plla_rates[x8_RATE] = 1277952000, + .plla_rates[x11_RATE] = 1264435200, + /* PLLA_OUT0 */ + .plla_out0_rates[x8_RATE] = 49152000, + .plla_out0_rates[x11_RATE] = 45158400, +}; + static const struct tegra_audio_cdata tegra264_data = { /* PLLA1 */ .plla_rates[x8_RATE] = 983040000, @@ -245,6 +254,8 @@ static const struct of_device_id graph_of_tegra_match[] = { .data = &tegra210_data }, { .compatible = "nvidia,tegra186-audio-graph-card", .data = &tegra186_data }, + { .compatible = "nvidia,tegra238-audio-graph-card", + .data = &tegra238_data }, { .compatible = "nvidia,tegra264-audio-graph-card", .data = &tegra264_data }, {}, diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 73bce9712dbd..bf4401aba76c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -160,8 +160,8 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) * This won't be used for implicit feedback which takes the packet size * returned from the sync source */ -static int slave_next_packet_size(struct snd_usb_endpoint *ep, - unsigned int avail) +static int synced_next_packet_size(struct snd_usb_endpoint *ep, + unsigned int avail) { unsigned int phase; int ret; @@ -221,13 +221,14 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep, packet = ctx->packet_size[idx]; if (packet) { + packet = min(packet, ep->maxframesize); if (avail && packet >= avail) return -EAGAIN; return packet; } if (ep->sync_source) - return slave_next_packet_size(ep, avail); + return synced_next_packet_size(ep, avail); else return next_packet_size(ep, avail); } @@ -1378,6 +1379,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, return -EINVAL; } + ep->packsize[0] = min(ep->packsize[0], ep->maxframesize); + ep->packsize[1] = min(ep->packsize[1], ep->maxframesize); + /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; ep->freqshift = INT_MIN; diff --git a/sound/usb/format.c b/sound/usb/format.c index 64cfe4a9d8cd..1207c507882a 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -305,17 +305,48 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, } /* - * Many Focusrite devices supports a limited set of sampling rates per - * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type - * descriptor which has a non-standard bLength = 10. + * Focusrite devices use rate pairs: 44100/48000, 88200/96000, and + * 176400/192000. Return true if rate is in the pair for max_rate. + */ +static bool focusrite_rate_pair(unsigned int rate, + unsigned int max_rate) +{ + switch (max_rate) { + case 48000: return rate == 44100 || rate == 48000; + case 96000: return rate == 88200 || rate == 96000; + case 192000: return rate == 176400 || rate == 192000; + default: return true; + } +} + +/* + * Focusrite devices report all supported rates in a single clock + * source but only a subset is valid per altsetting. + * + * Detection uses two descriptor features: + * + * 1. Format Type descriptor bLength == 10: non-standard extension + * with max sample rate in bytes 6..9. + * + * 2. bmControls VAL_ALT_SETTINGS readable bit: when set, the device + * only supports the highest rate pair for that altsetting, and when + * clear, all rates up to max_rate are valid. + * + * For devices without the bLength == 10 extension but with + * VAL_ALT_SETTINGS readable and multiple altsettings (only seen in + * Scarlett 18i8 3rd Gen playback), fall back to the Focusrite + * convention: alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz. */ static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int rate) { + struct usb_interface *iface; struct usb_host_interface *alts; + struct uac2_as_header_descriptor *as; unsigned char *fmt; unsigned int max_rate; + bool val_alt; alts = snd_usb_get_host_interface(chip, fp->iface, fp->altsetting); if (!alts) @@ -326,9 +357,21 @@ static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, if (!fmt) return true; + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_AS_GENERAL); + if (!as) + return true; + + val_alt = uac_v2v3_control_is_readable(as->bmControls, + UAC2_AS_VAL_ALT_SETTINGS); + if (fmt[0] == 10) { /* bLength */ max_rate = combine_quad(&fmt[6]); + if (val_alt) + return focusrite_rate_pair(rate, max_rate); + + /* No val_alt: rates fall through from higher */ switch (max_rate) { case 192000: if (rate == 176400 || rate == 192000) @@ -344,12 +387,29 @@ static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, usb_audio_info(chip, "%u:%d : unexpected max rate: %u\n", fp->iface, fp->altsetting, max_rate); - return true; } } - return true; + if (!val_alt) + return true; + + /* Multi-altsetting device with val_alt but no max_rate + * in the format descriptor. Use Focusrite convention: + * alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz. + */ + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface || iface->num_altsetting <= 2) + return true; + + switch (fp->altsetting) { + case 1: max_rate = 48000; break; + case 2: max_rate = 96000; break; + case 3: max_rate = 192000; break; + default: return true; + } + + return focusrite_rate_pair(rate, max_rate); } /* diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 473cb29efa7f..7eac7d1bce64 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -71,7 +71,7 @@ * * e I guess the same as with mixer * */ -/** struct s1810c_ctl_packet - basic vendor request +/* struct s1810c_ctl_packet - basic vendor request * @selector: device/mixer/output * @b: request-dependant field b * @tag: fixed value identifying type of request @@ -94,14 +94,14 @@ struct s1810c_ctl_packet { __le32 e; }; -/** selectors for CMD request +/* selectors for CMD request */ #define SC1810C_SEL_DEVICE 0 #define SC1810C_SEL_MIXER 0x64 #define SC1810C_SEL_OUTPUT 0x65 -/** control ids */ +/* control ids */ #define SC1810C_CTL_LINE_SW 0 #define SC1810C_CTL_MUTE_SW 1 #define SC1824C_CTL_MONO_SW 2 @@ -127,7 +127,7 @@ struct s1810c_ctl_packet { #define SC1810C_GET_STATE_TAG SC1810C_SET_STATE_TAG #define SC1810C_GET_STATE_LEN SC1810C_SET_STATE_LEN -/** Mixer levels normally range from 0 (off) to 0x0100 0000 (0 dB). +/* Mixer levels normally range from 0 (off) to 0x0100 0000 (0 dB). * raw_level = 2^24 * 10^(db_level / 20), thus * -3dB = 0xb53bf0 (technically, half-power -3.01...dB would be 0xb504f3) * -96dB = 0x109 @@ -145,7 +145,7 @@ struct s1810c_ctl_packet { #define MIXER_LEVEL_N3DB 0xb53bf0 #define MIXER_LEVEL_0DB 0x1000000 -/** +/* * This packet includes mixer volumes and * various other fields, it's an extended * version of ctl_packet, with a and b @@ -155,7 +155,7 @@ struct s1810c_state_packet { __le32 fields[63]; }; -/** indices into s1810c_state_packet.fields[] +/* indices into s1810c_state_packet.fields[] */ #define SC1810C_STATE_TAG_IDX 2 #define SC1810C_STATE_LEN_IDX 3 diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 85a0316889d4..fd1fb668929a 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1328,8 +1328,6 @@ struct scarlett2_data { struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX]; struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX]; struct snd_kcontrol *compressor_ctls[SCARLETT2_COMPRESSOR_CTLS_MAX]; - struct snd_kcontrol *precomp_flt_ctls[SCARLETT2_PRECOMP_FLT_CTLS_MAX]; - struct snd_kcontrol *peq_flt_ctls[SCARLETT2_PEQ_FLT_CTLS_MAX]; struct snd_kcontrol *precomp_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX]; struct snd_kcontrol *peq_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX]; struct snd_kcontrol *direct_monitor_ctl; @@ -3447,7 +3445,6 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) private->autogain_status[i] = private->num_autogain_status_texts - 1; - for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) if (scarlett2_has_config_item(private, scarlett2_ag_target_configs[i])) { @@ -5372,8 +5369,7 @@ static int scarlett2_update_filter_values(struct usb_mixer_interface *mixer) err = scarlett2_usb_get_config( mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH, - info->dsp_input_count * info->peq_flt_count, - private->peq_flt_switch); + info->dsp_input_count, private->peq_flt_switch); if (err < 0) return err; @@ -6546,7 +6542,7 @@ static int scarlett2_add_dsp_ctls(struct usb_mixer_interface *mixer, int i) err = scarlett2_add_new_ctl( mixer, &scarlett2_precomp_flt_ctl, i * info->precomp_flt_count + j, - 1, s, &private->precomp_flt_switch_ctls[j]); + 1, s, NULL); if (err < 0) return err; } @@ -6556,7 +6552,7 @@ static int scarlett2_add_dsp_ctls(struct usb_mixer_interface *mixer, int i) err = scarlett2_add_new_ctl( mixer, &scarlett2_peq_flt_ctl, i * info->peq_flt_count + j, - 1, s, &private->peq_flt_switch_ctls[j]); + 1, s, NULL); if (err < 0) return err; } @@ -8255,6 +8251,8 @@ static int scarlett2_find_fc_interface(struct usb_device *dev, if (desc->bInterfaceClass != 255) continue; + if (desc->bNumEndpoints < 1) + continue; epd = get_endpoint(intf->altsetting, 0); private->bInterfaceNumber = desc->bInterfaceNumber; diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index 01e6063c2207..510b68cced33 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -1007,7 +1007,7 @@ put_suspend: /** * uaudio_transfer_buffer_setup() - fetch and populate xfer buffer params * @subs: usb substream - * @xfer_buf: xfer buf to be allocated + * @xfer_buf_cpu: xfer buf to be allocated * @xfer_buf_len: size of allocation * @mem_info: QMI response info * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4cac0dfb0094..049a94079f9e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2219,6 +2219,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x05e1, 0x0480, /* Hauppauge Woodbury */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x0624, 0x3d3f, /* AB13X USB Audio */ + QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x0644, 0x8043, /* TEAC UD-501/UD-501V2/UD-503/NT-503 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY), @@ -2241,6 +2243,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IFACE_DELAY | QUIRK_FLAG_FORCE_IFACE_RESET), DEVICE_FLG(0x0661, 0x0883, /* iBasso DC04 Ultra */ QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x0666, 0x0880, /* SPACETOUCH USB Audio */ + QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ @@ -2365,6 +2369,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x2040, 0x7281, /* Hauppauge HVR-950Q-MXL */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x20b1, 0x2009, /* XMOS Ltd DIYINHK USB Audio 2.0 */ + QUIRK_FLAG_SKIP_IMPLICIT_FB | QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2040, 0x8200, /* Hauppauge Woodbury */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ @@ -2424,7 +2430,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { VENDOR_FLG(0x07fd, /* MOTU */ QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x1235, /* Focusrite Novation */ - QUIRK_FLAG_VALIDATE_RATES), + QUIRK_FLAG_SKIP_CLOCK_SELECTOR | + QUIRK_FLAG_SKIP_IFACE_SETUP), VENDOR_FLG(0x1511, /* AURALiC */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x152a, /* Thesycon devices */ @@ -2506,6 +2513,7 @@ static const char *const snd_usb_audio_quirk_flag_names[] = { QUIRK_STRING_ENTRY(MIC_RES_384), QUIRK_STRING_ENTRY(MIXER_PLAYBACK_MIN_MUTE), QUIRK_STRING_ENTRY(MIXER_CAPTURE_MIN_MUTE), + QUIRK_STRING_ENTRY(SKIP_IFACE_SETUP), NULL }; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ac4d92065dd9..d38c39e28f38 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1259,6 +1259,9 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, set_iface_first = true; /* try to set the interface... */ + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IFACE_SETUP) + continue; + usb_set_interface(chip->dev, iface_no, 0); if (set_iface_first) usb_set_interface(chip->dev, iface_no, altno); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 79978cae9799..085530cf62d9 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -224,6 +224,10 @@ extern bool snd_usb_skip_validation; * playback value represents muted state instead of minimum audible volume * QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE * Similar to QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE, but for capture streams + * QUIRK_FLAG_SKIP_IFACE_SETUP + * Skip the probe-time interface setup (usb_set_interface, + * init_pitch, init_sample_rate); redundant with + * snd_usb_endpoint_prepare() at stream-open time */ enum { @@ -253,6 +257,7 @@ enum { QUIRK_TYPE_MIC_RES_384 = 23, QUIRK_TYPE_MIXER_PLAYBACK_MIN_MUTE = 24, QUIRK_TYPE_MIXER_CAPTURE_MIN_MUTE = 25, + QUIRK_TYPE_SKIP_IFACE_SETUP = 26, /* Please also edit snd_usb_audio_quirk_flag_names */ }; @@ -284,5 +289,6 @@ enum { #define QUIRK_FLAG_MIC_RES_384 QUIRK_FLAG(MIC_RES_384) #define QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE QUIRK_FLAG(MIXER_PLAYBACK_MIN_MUTE) #define QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE QUIRK_FLAG(MIXER_CAPTURE_MIN_MUTE) +#define QUIRK_FLAG_SKIP_IFACE_SETUP QUIRK_FLAG(SKIP_IFACE_SETUP) #endif /* __USBAUDIO_H */ diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 011ea96e9779..f00b53346abd 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -520,8 +520,6 @@ static int us122l_usb_probe(struct usb_interface *intf, return err; } - usb_get_intf(usb_ifnum_to_if(device, 0)); - usb_get_dev(device); *cardp = card; return 0; } @@ -542,11 +540,9 @@ static int snd_us122l_probe(struct usb_interface *intf, if (intf->cur_altsetting->desc.bInterfaceNumber != 1) return 0; - err = us122l_usb_probe(usb_get_intf(intf), id, &card); - if (err < 0) { - usb_put_intf(intf); + err = us122l_usb_probe(intf, id, &card); + if (err < 0) return err; - } usb_set_intfdata(intf, card); return 0; @@ -574,10 +570,6 @@ static void snd_us122l_disconnect(struct usb_interface *intf) snd_usbmidi_disconnect(p); } - usb_put_intf(usb_ifnum_to_if(us122l->dev, 0)); - usb_put_intf(usb_ifnum_to_if(us122l->dev, 1)); - usb_put_dev(us122l->dev); - snd_card_free_when_closed(card); } diff --git a/sound/usb/usx2y/us144mkii.c b/sound/usb/usx2y/us144mkii.c index bc71968df8e2..0cf4fa74e210 100644 --- a/sound/usb/usx2y/us144mkii.c +++ b/sound/usb/usx2y/us144mkii.c @@ -10,8 +10,8 @@ MODULE_AUTHOR("Šerif Rami <ramiserifpersia@gmail.com>"); MODULE_DESCRIPTION("ALSA Driver for TASCAM US-144MKII"); MODULE_LICENSE("GPL"); -/** - * @brief Module parameters for ALSA card instantiation. +/* + * Module parameters for ALSA card instantiation. * * These parameters allow users to configure how the ALSA sound card * for the TASCAM US-144MKII is instantiated. @@ -269,7 +269,7 @@ void tascam_stop_work_handler(struct work_struct *work) atomic_set(&tascam->active_urbs, 0); } -/** +/* * tascam_card_private_free() - Frees private data associated with the sound * card. * @card: Pointer to the ALSA sound card instance. @@ -291,7 +291,7 @@ static void tascam_card_private_free(struct snd_card *card) } } -/** +/* * tascam_suspend() - Handles device suspension. * @intf: The USB interface being suspended. * @message: Power management message. @@ -332,7 +332,7 @@ static int tascam_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -/** +/* * tascam_resume() - Handles device resumption from suspend. * @intf: The USB interface being resumed. * @@ -390,7 +390,7 @@ static void tascam_error_timer(struct timer_list *t) schedule_work(&tascam->midi_out_work); } -/** +/* * tascam_probe() - Probes for the TASCAM US-144MKII device. * @intf: The USB interface being probed. * @usb_id: The USB device ID. @@ -565,7 +565,7 @@ free_card: return err; } -/** +/* * tascam_disconnect() - Disconnects the TASCAM US-144MKII device. * @intf: The USB interface being disconnected. * diff --git a/sound/usb/usx2y/us144mkii_capture.c b/sound/usb/usx2y/us144mkii_capture.c index 00188ff6cd51..af120bf62173 100644 --- a/sound/usb/usx2y/us144mkii_capture.c +++ b/sound/usb/usx2y/us144mkii_capture.c @@ -3,7 +3,7 @@ #include "us144mkii.h" -/** +/* * tascam_capture_open() - Opens the PCM capture substream. * @substream: The ALSA PCM substream to open. * @@ -23,7 +23,7 @@ static int tascam_capture_open(struct snd_pcm_substream *substream) return 0; } -/** +/* * tascam_capture_close() - Closes the PCM capture substream. * @substream: The ALSA PCM substream to close. * @@ -41,7 +41,7 @@ static int tascam_capture_close(struct snd_pcm_substream *substream) return 0; } -/** +/* * tascam_capture_prepare() - Prepares the PCM capture substream for use. * @substream: The ALSA PCM substream to prepare. * @@ -62,7 +62,7 @@ static int tascam_capture_prepare(struct snd_pcm_substream *substream) return 0; } -/** +/* * tascam_capture_pointer() - Returns the current capture pointer position. * @substream: The ALSA PCM substream. * @@ -91,7 +91,7 @@ tascam_capture_pointer(struct snd_pcm_substream *substream) return do_div(pos, runtime->buffer_size); } -/** +/* * tascam_capture_ops - ALSA PCM operations for capture. * * This structure defines the callback functions for capture stream operations, @@ -109,7 +109,7 @@ const struct snd_pcm_ops tascam_capture_ops = { .pointer = tascam_capture_pointer, }; -/** +/* * decode_tascam_capture_block() - Decodes a raw 512-byte block from the device. * @src_block: Pointer to the 512-byte raw source block. * @dst_block: Pointer to the destination buffer for decoded audio frames. diff --git a/sound/usb/usx2y/us144mkii_controls.c b/sound/usb/usx2y/us144mkii_controls.c index 62055fb8e7ba..81ded11e3709 100644 --- a/sound/usb/usx2y/us144mkii_controls.c +++ b/sound/usb/usx2y/us144mkii_controls.c @@ -3,8 +3,8 @@ #include "us144mkii.h" -/** - * @brief Text descriptions for playback output source options. +/* + * Text descriptions for playback output source options. * * Used by ALSA kcontrol elements to provide user-friendly names for * the playback routing options (e.g., "Playback 1-2", "Playback 3-4"). @@ -12,15 +12,15 @@ static const char *const playback_source_texts[] = { "Playback 1-2", "Playback 3-4" }; -/** - * @brief Text descriptions for capture input source options. +/* + * Text descriptions for capture input source options. * * Used by ALSA kcontrol elements to provide user-friendly names for * the capture routing options (e.g., "Analog In", "Digital In"). */ static const char *const capture_source_texts[] = { "Analog In", "Digital In" }; -/** +/* * tascam_playback_source_info() - ALSA control info callback for playback * source. * @kcontrol: The ALSA kcontrol instance. @@ -38,7 +38,7 @@ static int tascam_playback_source_info(struct snd_kcontrol *kcontrol, return snd_ctl_enum_info(uinfo, 1, 2, playback_source_texts); } -/** +/* * tascam_line_out_get() - ALSA control get callback for Line Outputs Source. * @kcontrol: The ALSA kcontrol instance. * @ucontrol: The ALSA control element value structure to fill. @@ -60,7 +60,7 @@ static int tascam_line_out_get(struct snd_kcontrol *kcontrol, return 0; } -/** +/* * tascam_line_out_put() - ALSA control put callback for Line Outputs Source. * @kcontrol: The ALSA kcontrol instance. * @ucontrol: The ALSA control element value structure containing the new value. @@ -89,7 +89,7 @@ static int tascam_line_out_put(struct snd_kcontrol *kcontrol, return changed; } -/** +/* * tascam_line_out_control - ALSA kcontrol definition for Line Outputs Source. * * This defines a new ALSA mixer control named "Line OUTPUTS Source" that allows @@ -106,7 +106,7 @@ static const struct snd_kcontrol_new tascam_line_out_control = { .put = tascam_line_out_put, }; -/** +/* * tascam_digital_out_get() - ALSA control get callback for Digital Outputs * Source. * @kcontrol: The ALSA kcontrol instance. @@ -129,7 +129,7 @@ static int tascam_digital_out_get(struct snd_kcontrol *kcontrol, return 0; } -/** +/* * tascam_digital_out_put() - ALSA control put callback for Digital Outputs * Source. * @kcontrol: The ALSA kcontrol instance. @@ -159,7 +159,7 @@ static int tascam_digital_out_put(struct snd_kcontrol *kcontrol, return changed; } -/** +/* * tascam_digital_out_control - ALSA kcontrol definition for Digital Outputs * Source. * @@ -177,7 +177,7 @@ static const struct snd_kcontrol_new tascam_digital_out_control = { .put = tascam_digital_out_put, }; -/** +/* * tascam_capture_source_info() - ALSA control info callback for capture source. * @kcontrol: The ALSA kcontrol instance. * @uinfo: The ALSA control element info structure to fill. @@ -194,7 +194,7 @@ static int tascam_capture_source_info(struct snd_kcontrol *kcontrol, return snd_ctl_enum_info(uinfo, 1, 2, capture_source_texts); } -/** +/* * tascam_capture_12_get() - ALSA control get callback for Capture channels 1 * and 2 Source. * @kcontrol: The ALSA kcontrol instance. @@ -217,7 +217,7 @@ static int tascam_capture_12_get(struct snd_kcontrol *kcontrol, return 0; } -/** +/* * tascam_capture_12_put() - ALSA control put callback for Capture channels 1 * and 2 Source. * @kcontrol: The ALSA kcontrol instance. @@ -247,7 +247,7 @@ static int tascam_capture_12_put(struct snd_kcontrol *kcontrol, return changed; } -/** +/* * tascam_capture_12_control - ALSA kcontrol definition for Capture channels 1 * and 2 Source. * @@ -265,7 +265,7 @@ static const struct snd_kcontrol_new tascam_capture_12_control = { .put = tascam_capture_12_put, }; -/** +/* * tascam_capture_34_get() - ALSA control get callback for Capture channels 3 * and 4 Source. * @kcontrol: The ALSA kcontrol instance. @@ -288,7 +288,7 @@ static int tascam_capture_34_get(struct snd_kcontrol *kcontrol, return 0; } -/** +/* * tascam_capture_34_put() - ALSA control put callback for Capture channels 3 * and 4 Source. * @kcontrol: The ALSA kcontrol instance. @@ -318,7 +318,7 @@ static int tascam_capture_34_put(struct snd_kcontrol *kcontrol, return changed; } -/** +/* * tascam_capture_34_control - ALSA kcontrol definition for Capture channels 3 * and 4 Source. * @@ -336,7 +336,7 @@ static const struct snd_kcontrol_new tascam_capture_34_control = { .put = tascam_capture_34_put, }; -/** +/* * tascam_samplerate_info() - ALSA control info callback for Sample Rate. * @kcontrol: The ALSA kcontrol instance. * @uinfo: The ALSA control element info structure to fill. @@ -356,7 +356,7 @@ static int tascam_samplerate_info(struct snd_kcontrol *kcontrol, return 0; } -/** +/* * tascam_samplerate_get() - ALSA control get callback for Sample Rate. * @kcontrol: The ALSA kcontrol instance. * @ucontrol: The ALSA control element value structure to fill. @@ -400,7 +400,7 @@ static int tascam_samplerate_get(struct snd_kcontrol *kcontrol, return 0; } -/** +/* * tascam_samplerate_control - ALSA kcontrol definition for Sample Rate. * * This defines a new ALSA mixer control named "Sample Rate" that displays diff --git a/sound/usb/usx2y/us144mkii_midi.c b/sound/usb/usx2y/us144mkii_midi.c index ed2afec2a89a..4871797b1670 100644 --- a/sound/usb/usx2y/us144mkii_midi.c +++ b/sound/usb/usx2y/us144mkii_midi.c @@ -3,7 +3,7 @@ #include "us144mkii.h" -/** +/* * tascam_midi_in_work_handler() - Deferred work for processing MIDI input. * @work: The work_struct instance. * @@ -75,7 +75,7 @@ out: usb_put_urb(urb); } -/** +/* * tascam_midi_in_open() - Opens the MIDI input substream. * @substream: The ALSA rawmidi substream to open. * @@ -92,7 +92,7 @@ static int tascam_midi_in_open(struct snd_rawmidi_substream *substream) return 0; } -/** +/* * tascam_midi_in_close() - Closes the MIDI input substream. * @substream: The ALSA rawmidi substream to close. * @@ -103,7 +103,7 @@ static int tascam_midi_in_close(struct snd_rawmidi_substream *substream) return 0; } -/** +/* * tascam_midi_in_trigger() - Triggers MIDI input stream activity. * @substream: The ALSA rawmidi substream. * @up: Boolean indicating whether to start (1) or stop (0) the stream. @@ -150,7 +150,7 @@ static void tascam_midi_in_trigger(struct snd_rawmidi_substream *substream, } } -/** +/* * tascam_midi_in_ops - ALSA rawmidi operations for MIDI input. * * This structure defines the callback functions for MIDI input stream @@ -205,7 +205,7 @@ out: usb_put_urb(urb); } -/** +/* * tascam_midi_out_work_handler() - Deferred work for sending MIDI data * @work: The work_struct instance. * @@ -282,7 +282,7 @@ static void tascam_midi_out_work_handler(struct work_struct *work) } } -/** +/* * tascam_midi_out_open() - Opens the MIDI output substream. * @substream: The ALSA rawmidi substream to open. * @@ -301,7 +301,7 @@ static int tascam_midi_out_open(struct snd_rawmidi_substream *substream) return 0; } -/** +/* * tascam_midi_out_close() - Closes the MIDI output substream. * @substream: The ALSA rawmidi substream to close. * @@ -312,7 +312,7 @@ static int tascam_midi_out_close(struct snd_rawmidi_substream *substream) return 0; } -/** +/* * tascam_midi_out_drain() - Drains the MIDI output stream. * @substream: The ALSA rawmidi substream. * @@ -340,7 +340,7 @@ static void tascam_midi_out_drain(struct snd_rawmidi_substream *substream) usb_kill_anchored_urbs(&tascam->midi_out_anchor); } -/** +/* * tascam_midi_out_trigger() - Triggers MIDI output stream activity. * @substream: The ALSA rawmidi substream. * @up: Boolean indicating whether to start (1) or stop (0) the stream. @@ -361,7 +361,7 @@ static void tascam_midi_out_trigger(struct snd_rawmidi_substream *substream, } } -/** +/* * tascam_midi_out_ops - ALSA rawmidi operations for MIDI output. * * This structure defines the callback functions for MIDI output stream diff --git a/sound/usb/usx2y/us144mkii_playback.c b/sound/usb/usx2y/us144mkii_playback.c index 0cb9699ec211..7efaca0a6489 100644 --- a/sound/usb/usx2y/us144mkii_playback.c +++ b/sound/usb/usx2y/us144mkii_playback.c @@ -3,7 +3,7 @@ #include "us144mkii.h" -/** +/* * tascam_playback_open() - Opens the PCM playback substream. * @substream: The ALSA PCM substream to open. * @@ -23,7 +23,7 @@ static int tascam_playback_open(struct snd_pcm_substream *substream) return 0; } -/** +/* * tascam_playback_close() - Closes the PCM playback substream. * @substream: The ALSA PCM substream to close. * @@ -41,7 +41,7 @@ static int tascam_playback_close(struct snd_pcm_substream *substream) return 0; } -/** +/* * tascam_playback_prepare() - Prepares the PCM playback substream for use. * @substream: The ALSA PCM substream to prepare. * @@ -108,7 +108,7 @@ static int tascam_playback_prepare(struct snd_pcm_substream *substream) return 0; } -/** +/* * tascam_playback_pointer() - Returns the current playback pointer position. * @substream: The ALSA PCM substream. * @@ -137,7 +137,7 @@ tascam_playback_pointer(struct snd_pcm_substream *substream) return do_div(pos, runtime->buffer_size); } -/** +/* * tascam_playback_ops - ALSA PCM operations for playback. * * This structure defines the callback functions for playback stream operations, diff --git a/sound/usb/validate.c b/sound/usb/validate.c index 4bb4893f6e74..f62b7cc041dc 100644 --- a/sound/usb/validate.c +++ b/sound/usb/validate.c @@ -281,7 +281,7 @@ static const struct usb_desc_validator audio_validators[] = { /* UAC_VERSION_2, UAC2_SAMPLE_RATE_CONVERTER: not implemented yet */ /* UAC3 */ - FIXED(UAC_VERSION_2, UAC_HEADER, struct uac3_ac_header_descriptor), + FIXED(UAC_VERSION_3, UAC_HEADER, struct uac3_ac_header_descriptor), FIXED(UAC_VERSION_3, UAC_INPUT_TERMINAL, struct uac3_input_terminal_descriptor), FIXED(UAC_VERSION_3, UAC_OUTPUT_TERMINAL, |