diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 33fd34f0d615..746eaf93e1a5 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2974,8 +2974,10 @@ static void snd_pcm_oss_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_str *pstr = entry->private_data; - struct snd_pcm_oss_setup *setup = pstr->oss.setup_list; + struct snd_pcm_oss_setup *setup; + guard(mutex)(&pstr->oss.setup_mutex); + setup = pstr->oss.setup_list; while (setup) { snd_iprintf(buffer, "%s %u %u%s%s%s%s%s%s\n", setup->task_name, @@ -3060,6 +3062,13 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, buffer->error = -ENOMEM; return; } + template.task_name = kstrdup(task_name, GFP_KERNEL); + if (!template.task_name) { + kfree(setup); + buffer->error = -ENOMEM; + return; + } + *setup = template; if (pstr->oss.setup_list == NULL) pstr->oss.setup_list = setup; else { @@ -3067,12 +3076,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, setup1->next; setup1 = setup1->next); setup1->next = setup; } - template.task_name = kstrdup(task_name, GFP_KERNEL); - if (! template.task_name) { - kfree(setup); - buffer->error = -ENOMEM; - return; - } + continue; } *setup = template; } diff --git a/sound/firewire/motu/motu-register-dsp-message-parser.c b/sound/firewire/motu/motu-register-dsp-message-parser.c index a8053e3ef065..4ec23e6880d9 100644 --- a/sound/firewire/motu/motu-register-dsp-message-parser.c +++ b/sound/firewire/motu/motu-register-dsp-message-parser.c @@ -386,6 +386,8 @@ unsigned int snd_motu_register_dsp_message_parser_count_event(struct snd_motu *m { struct msg_parser *parser = motu->message_parser; + guard(spinlock_irqsave)(&parser->lock); + if (parser->pull_pos > parser->push_pos) return EVENT_QUEUE_SIZE - parser->pull_pos + parser->push_pos; else @@ -395,13 +397,14 @@ unsigned int snd_motu_register_dsp_message_parser_count_event(struct snd_motu *m bool snd_motu_register_dsp_message_parser_copy_event(struct snd_motu *motu, u32 *event) { struct msg_parser *parser = motu->message_parser; - unsigned int pos = parser->pull_pos; - - if (pos == parser->push_pos) - return false; + unsigned int pos; guard(spinlock_irqsave)(&parser->lock); + if (parser->pull_pos == parser->push_pos) + return false; + + pos = parser->pull_pos; *event = parser->event_queue[pos]; ++pos; diff --git a/sound/hda/codecs/cirrus/cs420x.c b/sound/hda/codecs/cirrus/cs420x.c index 42559edbba05..85c2ecf46d38 100644 --- a/sound/hda/codecs/cirrus/cs420x.c +++ b/sound/hda/codecs/cirrus/cs420x.c @@ -582,6 +582,7 @@ static const struct hda_quirk cs4208_mac_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7800, "MacPro 6,1", CS4208_MACMINI), SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11), + SND_PCI_QUIRK(0x106b, 0x7f00, "iMac 16,1", CS4208_MBP11), {} /* terminator */ }; diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index f180d6a72021..dcbc669842e0 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -5458,7 +5458,7 @@ static const struct hda_fixup alc269_fixups[] = { [ALC299_FIXUP_PREDATOR_SPK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x21, 0x90170150 }, /* use as headset mic, without its own jack detect */ + { 0x21, 0x90170150 }, /* use as internal speaker */ { } } }, @@ -7070,6 +7070,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 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, 0x8a06, "HP Dragonfly Folio G3 2-in-1", ALC245_FIXUP_CS35L41_SPI_4_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), @@ -7324,6 +7325,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), HDA_CODEC_QUIRK(0x1043, 0x1204, "ASUS Strix G16 G615JMR", ALC287_FIXUP_TXNW2781_I2C_ASUS), SND_PCI_QUIRK(0x1043, 0x1204, "ASUS Strix G615JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C), + HDA_CODEC_QUIRK(0x1043, 0x1214, "ASUS ROG Strix G615LP", ALC287_FIXUP_TXNW2781_I2C_ASUS), SND_PCI_QUIRK(0x1043, 0x1214, "ASUS Strix G615LH_LM_LP", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), @@ -7778,6 +7780,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3929, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x392b, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + HDA_CODEC_QUIRK(0x17aa, 0x394c, "Lenovo Yoga Slim 7 14AGP11", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), @@ -7845,6 +7848,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1e39, 0xca14, "MEDION NM14LNL", ALC233_FIXUP_MEDION_MTL_SPK), SND_PCI_QUIRK(0x1e50, 0x7007, "Positivo DN50E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1e50, 0x7038, "Positivo DN140", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1ee7, 0x2078, "HONOR BRB-X M1010", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1ee7, 0x2081, "HONOR MRB-XXX M1020", ALC256_FIXUP_HONOR_MRB_XXX_M1020_AUDIO), SND_PCI_QUIRK(0x1f4c, 0xe001, "Minisforum V3 (SE)", ALC245_FIXUP_BASS_HP_DAC), diff --git a/sound/hda/codecs/side-codecs/cs35l56_hda.c b/sound/hda/codecs/side-codecs/cs35l56_hda.c index cdbc576569ef..a0ea08eb96a9 100644 --- a/sound/hda/codecs/side-codecs/cs35l56_hda.c +++ b/sound/hda/codecs/side-codecs/cs35l56_hda.c @@ -1025,7 +1025,7 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) u32 values[HDA_MAX_COMPONENTS]; char hid_string[8]; struct acpi_device *adev; - const char *property, *sub; + const char *property; int i, ret; /* @@ -1047,7 +1047,8 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) /* Initialize things that could be overwritten by a fixup */ cs35l56->index = -1; - sub = acpi_get_subsystem_id(ACPI_HANDLE(cs35l56->base.dev)); + const char *sub __free(kfree) = acpi_get_subsystem_id(ACPI_HANDLE(cs35l56->base.dev)); + ret = cs35l56_hda_apply_platform_fixups(cs35l56, sub, &id); if (ret) return ret; @@ -1095,15 +1096,16 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) ret = cirrus_scodec_get_speaker_id(cs35l56->base.dev, cs35l56->index, cs35l56->num_amps, -1); if (ret == -ENOENT) { - cs35l56->system_name = sub; + cs35l56->system_name = devm_kstrdup(cs35l56->base.dev, sub, GFP_KERNEL); } else if (ret >= 0) { - cs35l56->system_name = kasprintf(GFP_KERNEL, "%s-spkid%d", sub, ret); - kfree(sub); - if (!cs35l56->system_name) - return -ENOMEM; + cs35l56->system_name = devm_kasprintf(cs35l56->base.dev, GFP_KERNEL, + "%s-spkid%d", sub, ret); } else { return ret; } + + if (!cs35l56->system_name) + return -ENOMEM; } cs35l56->base.reset_gpio = devm_gpiod_get_index_optional(cs35l56->base.dev, @@ -1254,7 +1256,6 @@ void cs35l56_hda_remove(struct device *dev) cs_dsp_remove(&cs35l56->cs_dsp); - kfree(cs35l56->system_name); pm_runtime_put_noidle(cs35l56->base.dev); gpiod_set_value_cansleep(cs35l56->base.reset_gpio, 0); diff --git a/sound/soc/codecs/simple-mux.c b/sound/soc/codecs/simple-mux.c index 069555f35f73..c2f906a3f074 100644 --- a/sound/soc/codecs/simple-mux.c +++ b/sound/soc/codecs/simple-mux.c @@ -51,7 +51,7 @@ static int simple_mux_control_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *c = snd_soc_dapm_to_component(dapm); struct simple_mux *priv = snd_soc_component_get_drvdata(c); - if (ucontrol->value.enumerated.item[0] > e->items) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; if (priv->mux == ucontrol->value.enumerated.item[0]) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 192e2a394ff3..ea387dc74273 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -40,6 +40,7 @@ struct byt_cht_es8316_private { struct gpio_desc *speaker_en_gpio; struct device *codec_dev; bool speaker_en; + bool mclk_enabled; }; enum { @@ -170,6 +171,15 @@ static struct snd_soc_jack_pin byt_cht_es8316_jack_pins[] = { }, }; +static void byt_cht_es8316_disable_mclk(struct byt_cht_es8316_private *priv) +{ + if (!priv->mclk_enabled) + return; + + clk_disable_unprepare(priv->mclk); + priv->mclk_enabled = false; +} + static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_component *codec = snd_soc_rtd_to_codec(runtime, 0)->component; @@ -227,12 +237,14 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) ret = clk_prepare_enable(priv->mclk); if (ret) dev_err(card->dev, "unable to enable MCLK\n"); + else + priv->mclk_enabled = true; ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(runtime, 0), 0, 19200000, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(card->dev, "can't set codec clock %d\n", ret); - return ret; + goto err_disable_mclk; } ret = snd_soc_card_jack_new_pins(card, "Headset", @@ -241,13 +253,25 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) ARRAY_SIZE(byt_cht_es8316_jack_pins)); if (ret) { dev_err(card->dev, "jack creation failed %d\n", ret); - return ret; + goto err_disable_mclk; } snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_soc_component_set_jack(codec, &priv->jack, NULL); return 0; + +err_disable_mclk: + byt_cht_es8316_disable_mclk(priv); + return ret; +} + +static void byt_cht_es8316_exit(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); + + byt_cht_es8316_disable_mclk(priv); } static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, @@ -353,6 +377,7 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { | SND_SOC_DAIFMT_CBC_CFC, .be_hw_params_fixup = byt_cht_es8316_codec_fixup, .init = byt_cht_es8316_init, + .exit = byt_cht_es8316_exit, SND_SOC_DAILINK_REG(ssp2_port, ssp2_codec, platform), }, }; diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 4f8f7db6c3d3..4f09fdd40905 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -186,12 +186,10 @@ static void event_handler(uint32_t opcode, uint32_t token, case ASM_CLIENT_EVENT_CMD_RUN_DONE: break; case ASM_CLIENT_EVENT_CMD_EOS_DONE: - prtd->state = Q6ASM_STREAM_STOPPED; break; - case ASM_CLIENT_EVENT_DATA_WRITE_DONE: { + case ASM_CLIENT_EVENT_DATA_WRITE_DONE: snd_pcm_period_elapsed(substream); break; - } case ASM_CLIENT_EVENT_DATA_READ_DONE: snd_pcm_period_elapsed(substream); if (prtd->state == Q6ASM_STREAM_RUNNING) @@ -227,9 +225,19 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, /* rate and channels are sent to audio driver */ if (prtd->state == Q6ASM_STREAM_RUNNING) { /* clear the previous setup if any */ - q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); - q6asm_unmap_memory_regions(substream->stream, - prtd->audio_client); + ret = q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); + if (ret < 0) { + dev_err(dev, "Failed to close q6asm stream %d\n", prtd->stream_id); + return ret; + } + + ret = q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); + if (ret < 0) { + dev_err(dev, "Failed to unmap memory regions for q6asm stream %d\n", + prtd->stream_id); + return ret; + } + q6routing_stream_close(soc_prtd->dai_link->id, substream->stream); prtd->state = Q6ASM_STREAM_STOPPED; @@ -297,8 +305,6 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); open_err: q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; return ret; } @@ -341,7 +347,6 @@ static int q6asm_dai_trigger(struct snd_soc_component *component, 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: - prtd->state = Q6ASM_STREAM_STOPPED; ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, CMD_EOS); break; @@ -378,7 +383,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, return -EINVAL; } - prtd = kzalloc_obj(struct q6asm_dai_rtd); + prtd = kzalloc_obj(*prtd); if (prtd == NULL) return -ENOMEM; @@ -457,12 +462,12 @@ static int q6asm_dai_close(struct snd_soc_component *component, struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { - if (prtd->state) + if (prtd->state == Q6ASM_STREAM_RUNNING) { q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); - - q6asm_unmap_memory_regions(substream->stream, + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); + } q6asm_audio_client_free(prtd->audio_client); prtd->audio_client = NULL; } @@ -555,8 +560,6 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, snd_compr_drain_notify(prtd->cstream); prtd->notify_on_drain = false; - } else { - prtd->state = Q6ASM_STREAM_STOPPED; } break; @@ -674,7 +677,7 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = stream->private_data; if (prtd->audio_client) { - if (prtd->state) { + if (prtd->state == Q6ASM_STREAM_RUNNING) { q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); if (prtd->next_track_stream_id) { @@ -682,11 +685,11 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component, prtd->next_track_stream_id, CMD_CLOSE); } - } - snd_dma_free_pages(&prtd->dma_buffer); - q6asm_unmap_memory_regions(stream->direction, + q6asm_unmap_memory_regions(stream->direction, prtd->audio_client); + } + snd_dma_free_pages(&prtd->dma_buffer); q6asm_audio_client_free(prtd->audio_client); prtd->audio_client = NULL; } @@ -916,7 +919,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, prtd->session_id, dir); if (ret) { dev_err(dev, "Stream reg failed ret:%d\n", ret); - goto q6_err; + goto routing_err; } ret = __q6asm_dai_compr_set_codec_params(component, stream, @@ -942,11 +945,11 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, return 0; q6_err: + q6routing_stream_close(rtd->dai_link->id, dir); +routing_err: q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); open_err: - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; return ret; } @@ -1014,7 +1017,6 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component, 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: - prtd->state = Q6ASM_STREAM_STOPPED; ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, CMD_EOS); break; diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 8e80a7165faf..a4fac4652201 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2504,6 +2504,27 @@ static int scarlett2_has_config_item( return !!private->config_set->items[config_item_num].offset; } +/* Return the configuration item's offset, applying any per-firmware + * overrides. + * + * Firmware 2417 for the 2i2 Gen 4 moved DIRECT_MONITOR_GAIN by 4 + * bytes. Apply that shift here so that the rest of the driver can + * keep using the single config set. This override can be removed + * once the multi-config-set framework lands. + */ +static int scarlett2_config_item_offset( + struct scarlett2_data *private, int config_item_num) +{ + int offset = private->config_set->items[config_item_num].offset; + + if (config_item_num == SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN && + private->info == &s2i2_gen4_info && + private->firmware_version >= 2417) + offset = 0x2a4; + + return offset; +} + /* Send a USB message to get configuration parameters; result placed in *buf */ static int scarlett2_usb_get_config( struct usb_mixer_interface *mixer, @@ -2513,6 +2534,7 @@ static int scarlett2_usb_get_config( const struct scarlett2_config *config_item = &private->config_set->items[config_item_num]; int size, err, i; + int item_offset; u8 *buf_8; u8 value; @@ -2522,13 +2544,15 @@ static int scarlett2_usb_get_config( if (!config_item->offset) return -EFAULT; + item_offset = scarlett2_config_item_offset(private, config_item_num); + /* Writes to the parameter buffer are always 1 byte */ size = config_item->size ? config_item->size : 8; /* For byte-sized parameters, retrieve directly into buf */ if (size >= 8) { size = size / 8 * count; - err = scarlett2_usb_get(mixer, config_item->offset, buf, size); + err = scarlett2_usb_get(mixer, item_offset, buf, size); if (err < 0) return err; if (config_item->size == 16) { @@ -2546,7 +2570,7 @@ static int scarlett2_usb_get_config( } /* For bit-sized parameters, retrieve into value */ - err = scarlett2_usb_get(mixer, config_item->offset, &value, 1); + err = scarlett2_usb_get(mixer, item_offset, &value, 1); if (err < 0) return err; @@ -2696,7 +2720,8 @@ static int scarlett2_usb_set_config( */ if (config_item->size >= 8) { size = config_item->size / 8; - offset = config_item->offset + index * size; + offset = scarlett2_config_item_offset(private, config_item_num) + + index * size; /* If updating a bit, retrieve the old value, set/clear the * bit as needed, and update value @@ -2705,7 +2730,7 @@ static int scarlett2_usb_set_config( u8 tmp; size = 1; - offset = config_item->offset; + offset = scarlett2_config_item_offset(private, config_item_num); err = scarlett2_usb_get(mixer, offset, &tmp, 1); if (err < 0) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 31cbe383ae65..3d1b3523b020 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2449,6 +2449,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), + DEVICE_FLG(0x25aa, 0x600b, /* TAE1159 */ + QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x262a, 0x9302, /* ddHiFi TC44C */ QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */