Merge tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small, mostly device-specific fixes.

  The significant one is the regression fix for USB-audio implicit
  feedback devices due to the incorrect frame size calculation, which
  landed in 5.8 and stable trees.

  In addition, a few usual HD-audio and USB-audio quirks, Intel HDMI
  fixes, ASoC fsl and rt5682 fixes, as well as the fix in
  compress-offload partial drain operation"

* tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: compress: fix partial_drain completion state
  ALSA: usb-audio: Add implicit feedback quirk for RTX6001
  ALSA: usb-audio: add quirk for MacroSilicon MS2109
  ALSA: hda/realtek: Enable headset mic of Acer Veriton N4660G with ALC269VC
  ALSA: hda/realtek: Enable headset mic of Acer C20-820 with ALC269VC
  ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
  ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
  ALSA: hda/hdmi: improve debug traces for stream lookups
  ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
  ALSA: opl3: fix infoleak in opl3
  ALSA: usb-audio: Replace s/frame/packet/ where appropriate
  ALSA: usb-audio: Fix packet size calculation
  AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module
  ALSA: hda - let hs_mic be picked ahead of hp_mic
  ASoC: rt5682: fix the pop noise while OMTP type headset plugin
  ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
  ASoC: fsl_mqs: Don't check clock is NULL before calling clk API

include/sound/compress_driver.h

@@ -66,6 +66,7 @@ struct snd_compr_runtime {
  * @direction: stream direction, playback/recording
  * @metadata_set: metadata set flag, true when set
  * @next_track: has userspace signal next track transition, true when set
+ * @partial_drain: undergoing partial_drain for stream, true when set
  * @private_data: pointer to DSP private data
  * @dma_buffer: allocated buffer if any
  */
@@ -78,6 +79,7 @@ struct snd_compr_stream {
 	enum snd_compr_direction direction;
 	bool metadata_set;
 	bool next_track;
+	bool partial_drain;
 	void *private_data;
 	struct snd_dma_buffer dma_buffer;
 };
@@ -182,7 +184,13 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream)
 	if (snd_BUG_ON(!stream))
 		return;
 
-	stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+	/* for partial_drain case we are back to running state on success */
+	if (stream->partial_drain) {
+		stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
+		stream->partial_drain = false; /* clear this flag as well */
+	} else {
+		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+	}
 
 	wake_up(&stream->runtime->sleep);
 }

sound/core/compress_offload.c

@@ -764,6 +764,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
 	if (!retval) {
+		/* clear flags and stop any drain wait */
+		stream->partial_drain = false;
+		stream->metadata_set = false;
 		snd_compr_drain_notify(stream);
 		stream->runtime->total_bytes_available = 0;
 		stream->runtime->total_bytes_transferred = 0;
@@ -921,6 +924,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 	if (stream->next_track == false)
 		return -EPERM;
 
+	stream->partial_drain = true;
 	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
 	if (retval) {
 		pr_debug("Partial drain returned failure\n");

sound/drivers/opl3/opl3_synth.c

@@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
 		{
 			struct snd_dm_fm_info info;
 
+			memset(&info, 0, sizeof(info));
+
 			info.fm_mode = opl3->fm_mode;
 			info.rhythm = opl3->rhythm;
 			if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))

sound/pci/hda/hda_auto_parser.c

@@ -72,6 +72,12 @@ static int compare_input_type(const void *ap, const void *bp)
 	if (a->type != b->type)
 		return (int)(a->type - b->type);
 
+	/* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */
+	if (a->is_headset_mic && b->is_headphone_mic)
+		return -1; /* don't swap */
+	else if (a->is_headphone_mic && b->is_headset_mic)
+		return 1; /* swap */
+
 	/* In case one has boost and the other one has not,
 	   pick the one with boost first. */
 	return (int)(b->has_boost_on_pin - a->has_boost_on_pin);

sound/pci/hda/patch_hdmi.c

@@ -259,7 +259,7 @@ static int hinfo_to_pcm_index(struct hda_codec *codec,
 		if (get_pcm_rec(spec, pcm_idx)->stream == hinfo)
 			return pcm_idx;
 
-	codec_warn(codec, "HDMI: hinfo %p not registered\n", hinfo);
+	codec_warn(codec, "HDMI: hinfo %p not tied to a PCM\n", hinfo);
 	return -EINVAL;
 }
 
@@ -277,7 +277,8 @@ static int hinfo_to_pin_index(struct hda_codec *codec,
 			return pin_idx;
 	}
 
-	codec_dbg(codec, "HDMI: hinfo %p not registered\n", hinfo);
+	codec_dbg(codec, "HDMI: hinfo %p (pcm %d) not registered\n", hinfo,
+		  hinfo_to_pcm_index(codec, hinfo));
 	return -EINVAL;
 }
 
@@ -1804,33 +1805,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
 
 static int hdmi_parse_codec(struct hda_codec *codec)
 {
-	hda_nid_t nid;
+	hda_nid_t start_nid;
+	unsigned int caps;
 	int i, nodes;
 
-	nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid);
-	if (!nid || nodes < 0) {
+	nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid);
+	if (!start_nid || nodes < 0) {
 		codec_warn(codec, "HDMI: failed to get afg sub nodes\n");
 		return -EINVAL;
 	}
 
-	for (i = 0; i < nodes; i++, nid++) {
-		unsigned int caps;
-		unsigned int type;
+	/*
+	 * hdmi_add_pin() assumes total amount of converters to
+	 * be known, so first discover all converters
+	 */
+	for (i = 0; i < nodes; i++) {
+		hda_nid_t nid = start_nid + i;
 
 		caps = get_wcaps(codec, nid);
-		type = get_wcaps_type(caps);
 
 		if (!(caps & AC_WCAP_DIGITAL))
 			continue;
 
-		switch (type) {
-		case AC_WID_AUD_OUT:
+		if (get_wcaps_type(caps) == AC_WID_AUD_OUT)
 			hdmi_add_cvt(codec, nid);
-			break;
-		case AC_WID_PIN:
+	}
+
+	/* discover audio pins */
+	for (i = 0; i < nodes; i++) {
+		hda_nid_t nid = start_nid + i;
+
+		caps = get_wcaps(codec, nid);
+
+		if (!(caps & AC_WCAP_DIGITAL))
+			continue;
+
+		if (get_wcaps_type(caps) == AC_WID_PIN)
 			hdmi_add_pin(codec, nid);
-			break;
-		}
 	}
 
 	return 0;

sound/pci/hda/patch_realtek.c

@@ -6149,6 +6149,9 @@ enum {
 	ALC236_FIXUP_HP_MUTE_LED,
 	ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
 	ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+	ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
+	ALC269VC_FIXUP_ACER_HEADSET_MIC,
+	ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -7327,6 +7330,35 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x14, 0x90100120 }, /* use as internal speaker */
+			{ 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */
+			{ 0x1a, 0x01011020 }, /* use as line out */
+			{ },
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC269VC_FIXUP_ACER_HEADSET_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x02a11030 }, /* use as headset mic */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a11130 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7342,10 +7374,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
 	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK),
+	SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS),
+	SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
@@ -7571,8 +7606,7 @@ static const struct snd_pci_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, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
-	SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_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, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),

sound/soc/amd/renoir/Makefile

@@ -2,6 +2,7 @@
 # Renoir platform Support
 snd-rn-pci-acp3x-objs	:= rn-pci-acp3x.o
 snd-acp3x-pdm-dma-objs	:= acp3x-pdm-dma.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR)	 += snd-rn-pci-acp3x.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR)	 += snd-acp3x-pdm-dma.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH)	+= acp3x-rn.o
+snd-acp3x-rn-objs	:= acp3x-rn.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR)	+= snd-rn-pci-acp3x.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR)	+= snd-acp3x-pdm-dma.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH)	+= snd-acp3x-rn.o

sound/soc/codecs/rt5682.c

@@ -932,7 +932,9 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 			RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, RT5682_PWR_CBJ);
-
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1,
+			RT5682_OSW_L_MASK | RT5682_OSW_R_MASK, 0);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH);
 
@@ -956,6 +958,11 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 			rt5682->jack_type = SND_JACK_HEADPHONE;
 			break;
 		}
+
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1,
+			RT5682_OSW_L_MASK | RT5682_OSW_R_MASK,
+			RT5682_OSW_L_EN | RT5682_OSW_R_EN);
 	} else {
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,

sound/soc/fsl/fsl_mqs.c

@@ -265,12 +265,20 @@ static int fsl_mqs_remove(struct platform_device *pdev)
 static int fsl_mqs_runtime_resume(struct device *dev)
 {
 	struct fsl_mqs *mqs_priv = dev_get_drvdata(dev);
+	int ret;
 
-	if (mqs_priv->ipg)
-		clk_prepare_enable(mqs_priv->ipg);
+	ret = clk_prepare_enable(mqs_priv->ipg);
+	if (ret) {
+		dev_err(dev, "failed to enable ipg clock\n");
+		return ret;
+	}
 
-	if (mqs_priv->mclk)
-		clk_prepare_enable(mqs_priv->mclk);
+	ret = clk_prepare_enable(mqs_priv->mclk);
+	if (ret) {
+		dev_err(dev, "failed to enable mclk clock\n");
+		clk_disable_unprepare(mqs_priv->ipg);
+		return ret;
+	}
 
 	if (mqs_priv->use_gpr)
 		regmap_write(mqs_priv->regmap, IOMUXC_GPR2,
@@ -292,11 +300,8 @@ static int fsl_mqs_runtime_suspend(struct device *dev)
 		regmap_read(mqs_priv->regmap, REG_MQS_CTRL,
 			    &mqs_priv->reg_mqs_ctrl);
 
-	if (mqs_priv->mclk)
-		clk_disable_unprepare(mqs_priv->mclk);
-
-	if (mqs_priv->ipg)
-		clk_disable_unprepare(mqs_priv->ipg);
+	clk_disable_unprepare(mqs_priv->mclk);
+	clk_disable_unprepare(mqs_priv->ipg);
 
 	return 0;
 }

sound/usb/card.h

@@ -84,10 +84,10 @@ struct snd_usb_endpoint {
 	dma_addr_t sync_dma;		/* DMA address of syncbuf */
 
 	unsigned int pipe;		/* the data i/o pipe */
-	unsigned int framesize[2];	/* small/large frame sizes in samples */
-	unsigned int sample_rem;	/* remainder from division fs/fps */
+	unsigned int packsize[2];	/* small/large packet sizes in samples */
+	unsigned int sample_rem;	/* remainder from division fs/pps */
 	unsigned int sample_accum;	/* sample accumulator */
-	unsigned int fps;		/* frames per second */
+	unsigned int pps;		/* packets per second */
 	unsigned int freqn;		/* nominal sampling rate in fs/fps in Q16.16 format */
 	unsigned int freqm;		/* momentary sampling rate in fs/fps in Q16.16 format */
 	int	   freqshift;		/* how much to shift the feedback value to get Q16.16 */

sound/usb/endpoint.c

@@ -159,11 +159,11 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
 		return ep->maxframesize;
 
 	ep->sample_accum += ep->sample_rem;
-	if (ep->sample_accum >= ep->fps) {
-		ep->sample_accum -= ep->fps;
-		ret = ep->framesize[1];
+	if (ep->sample_accum >= ep->pps) {
+		ep->sample_accum -= ep->pps;
+		ret = ep->packsize[1];
 	} else {
-		ret = ep->framesize[0];
+		ret = ep->packsize[0];
 	}
 
 	return ret;
@@ -1088,15 +1088,15 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 
 	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) {
 		ep->freqn = get_usb_full_speed_rate(rate);
-		ep->fps = 1000;
+		ep->pps = 1000 >> ep->datainterval;
 	} else {
 		ep->freqn = get_usb_high_speed_rate(rate);
-		ep->fps = 8000;
+		ep->pps = 8000 >> ep->datainterval;
 	}
 
-	ep->sample_rem = rate % ep->fps;
-	ep->framesize[0] = rate / ep->fps;
-	ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps;
+	ep->sample_rem = rate % ep->pps;
+	ep->packsize[0] = rate / ep->pps;
+	ep->packsize[1] = (rate + (ep->pps - 1)) / ep->pps;
 
 	/* calculate the frequency in 16.16 format */
 	ep->freqm = ep->freqn;

sound/usb/pcm.c

@@ -368,6 +368,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
 	case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+	case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;

sound/usb/quirks-table.h

@@ -3633,4 +3633,56 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
 	}
 },
 
+/*
+ * MacroSilicon MS2109 based HDMI capture cards
+ *
+ * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch.
+ * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if
+ * they pretend to be 96kHz mono as a workaround for stereo being broken
+ * by that...
+ *
+ * They also have swapped L-R channels, but that's for userspace to deal
+ * with.
+ */
+{
+	USB_DEVICE(0x534d, 0x2109),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "MacroSilicon",
+		.product_name = "MS2109",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = &(const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_ALIGN_TRANSFER,
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S16_LE,
+					.channels = 2,
+					.iface = 3,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = 0,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC |
+						USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_CONTINUOUS,
+					.rate_min = 48000,
+					.rate_max = 48000,
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+
 #undef USB_DEVICE_VENDOR_SPEC