Merge tag 'sound-7.1-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of recent small fixes and quirks.

  We still see a bit more changes than wished, but most of them are
  device-specific ones that are pretty safe to apply, while a core fix
  is a typical UAF fix for PCM core that was recently caught by fuzzer;
  so overall nothing looks really worrisome.

  Core:
   - Fix a UAF in PCM OSS proc interface

  HD-audio:
   - Fix memory leaks in CS35L56 driver
   - Various device-specific quirks for Realtek and CS420x codecs

  USB-audio:
   - Quirk for TAE1160 USB Audio
   - Fix for Scarlett2 Gen4 direct monitor gain

  ASoC:
   - Fixes for QCom q6asm-dai, Intel bytcht_es8316, and simple-mux codec

  FireWire:
   - Fix for Motu DSP event queue protection"

* tag 'sound-7.1-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: codecs: simple-mux: Fix enum control bounds check
  ALSA: usb-audio: Add iface reset and delay quirk for TAE1160 USB Audio
  ALSA: hda/cs420x: Add CS4208 fixup for iMac16,1
  ALSA: hda/realtek: add quirk for HP Dragonfly Folio G3 2-in-1
  ALSA: hda/realtek: Fix speaker output on ASUS ROG Strix G615LP
  ASoC: qcom: q6asm-dai: use pointer type with kzalloc_obj()
  ASoC: qcom: q6asm-dai: remove unnecessary braces
  ASoC: qcom: q6asm-dai: fix error handling in prepare and set_params
  ASoC: qcom: q6asm-dai: close stream only when running
  ASoC: qcom: q6asm-dai: do not set stream state in event and trigger callbacks
  ASoC: Intel: bytcht_es8316: Fix MCLK leak on init errors
  ALSA: hda/realtek: Limit mic boost on Positivo DN140
  ALSA: scarlett2: Fix 2i2 Gen 4 direct monitor gain on firmware 2417
  ALSA: pcm: oss: Fix setup list UAF on proc write error
  ALSA: hda: cs35l56: Fix system name string leaks
  ALSA: hda/realtek: Add HDA_CODEC_QUIRK for Lenovo Yoga Slim 7 14AGP11
  ALSA: hda/realtek: Fix incorrect comment for ALC299_FIXUP_PREDATOR_SPK
  ALSA: firewire-motu: Protect register DSP event queue positions

10 files changed, 117 insertions, 50 deletions

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 @@ routing_err:
 	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 */