Merge tag 'sound-7.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "There have been continuous flux but most of them are device-specific
  small fixes, while we see a few core fixes at this time (minor PCM fix
  for linked streams and a few ASoC core fixes for delayed work, etc)

  Core:
   - PCM: Fix use-after-free in linked stream drain

  ASoC:
   - core: Fixes for delayed works, empty DMI string handling and DT overlay
   - qcom: qdsp6: Fix ADSP stop/start crash via component removal ordering
   - tegra: Add support for Tegra238 audio graph card
   - amd: Fix missing error checks for clock acquisition
   - rt1011: Fix incorrect DAPM context retrieval helper

  HD-audio:
   - Add quirk for Gigabyte H610M, ASUS UM6702RC, HP 14s-dr5xxx, and
     ThinkPad X390

  USB-audio:
   - Scarlett2: Fix NULL dereference for malformed endpoint descriptors
   - Add quirk for SPACETOUCH"

* tag 'sound-7.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: amd: acp-mach-common: Add missing error check for clock acquisition
  ASoC: detect empty DMI strings
  ASoC: amd: acp3x-rt5682-max9836: Add missing error check for clock acquisition
  ALSA: usb-audio: Add iface reset and delay quirk for SPACETOUCH USB Audio
  ASoC: codecs: rt1011: Use component to get the dapm context in spk_mode_put
  ALSA: usb-audio: Check endpoint numbers at parsing Scarlett2 mixer interfaces
  ASoC: simple-card-utils: fix graph_util_is_ports0() for DT overlays
  ASoC: soc-core: flush delayed work before removing DAIs and widgets
  ASoC: soc-core: drop delayed_work_pending() check before flush
  ASoC: tegra: Add support for Tegra238 soundcard
  ALSA: hda/realtek: Add headset jack quirk for Thinkpad X390
  ALSA: hda/realtek: add HP Laptop 14s-dr5xxx mute LED quirk
  ALSA: hda/realtek: add quirk for ASUS UM6702RC
  ALSA: pcm: fix use-after-free on linked stream runtime in snd_pcm_drain()
  ALSA: hda/realtek: Add quirk for Gigabyte Technology to fix headphone
  firmware: cs_dsp: Fix fragmentation regression in firmware download
  ASoC: qcom: qdsp6: Fix q6apm remove ordering during ADSP stop and start

15 files changed, 102 insertions, 23 deletions

diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c
index b4f1c01e3b5b..5d8be0ac7c5e 100644
--- a/drivers/firmware/cirrus/cs_dsp.c
+++ b/drivers/firmware/cirrus/cs_dsp.c
@@ -1610,11 +1610,17 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware,
 			   region_name);
 
 		if (reg) {
+			/*
+			 * Although we expect the underlying bus does not require
+			 * physically-contiguous buffers, we pessimistically use
+			 * a temporary buffer instead of trusting that the
+			 * alignment of region->data is ok.
+			 */
 			region_len = le32_to_cpu(region->len);
 			if (region_len > buf_len) {
 				buf_len = round_up(region_len, PAGE_SIZE);
-				kfree(buf);
-				buf = kmalloc(buf_len, GFP_KERNEL | GFP_DMA);
+				vfree(buf);
+				buf = vmalloc(buf_len);
 				if (!buf) {
 					ret = -ENOMEM;
 					goto out_fw;
@@ -1643,7 +1649,7 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware,
 
 	ret = 0;
 out_fw:
-	kfree(buf);
+	vfree(buf);
 
 	if (ret == -EOVERFLOW)
 		cs_dsp_err(dsp, "%s: file content overflows file data\n", file);
@@ -2331,11 +2337,17 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 		}
 
 		if (reg) {
+			/*
+			 * Although we expect the underlying bus does not require
+			 * physically-contiguous buffers, we pessimistically use
+			 * a temporary buffer instead of trusting that the
+			 * alignment of blk->data is ok.
+			 */
 			region_len = le32_to_cpu(blk->len);
 			if (region_len > buf_len) {
 				buf_len = round_up(region_len, PAGE_SIZE);
-				kfree(buf);
-				buf = kmalloc(buf_len, GFP_KERNEL | GFP_DMA);
+				vfree(buf);
+				buf = vmalloc(buf_len);
 				if (!buf) {
 					ret = -ENOMEM;
 					goto out_fw;
@@ -2366,7 +2378,7 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 
 	ret = 0;
 out_fw:
-	kfree(buf);
+	vfree(buf);
 
 	if (ret == -EOVERFLOW)
 		cs_dsp_err(dsp, "%s: file content overflows file data\n", file);
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/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c
index 4c49f1195e1b..ab4b22fcb72e 100644
--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -6940,6 +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),
@@ -7273,6 +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),
@@ -7493,6 +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),
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/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/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/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/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/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/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/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index ef3150581eab..fd1fb668929a 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -8251,6 +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/quirks.c b/sound/usb/quirks.c
index d54a1a44a69b..049a94079f9e 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -2243,6 +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) */