Merge v6.19.9linux-rolling-stable

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

16 files changed, 167 insertions, 22 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0a358d94b17c..495ff93fcd1d 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 f5719e630d28..4c49f1195e1b 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,
@@ -6500,6 +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,
@@ -7662,6 +7685,7 @@ 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),
@@ -7758,6 +7782,7 @@ 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/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/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/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/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 aaeeadded7aa..7e9d0b393bf9 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 94cc6376a367..5b8367a966b9 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -712,6 +712,7 @@ 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 e4b21bf39e59..23ba821cd759 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/usb/endpoint.c b/sound/usb/endpoint.c
index 686f09529067..1a020ea55875 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -221,6 +221,7 @@ 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;
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_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 7b31504c5f24..4342d93ab771 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 a89ea2233180..caca0e586d83 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -2363,6 +2363,8 @@ 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 */