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

4 files changed, 65 insertions, 12 deletions

diff --git a/sound/usb/card.h b/sound/usb/card.h
index d6219fba9699..de43267b9c8a 100644
--- a/sound/usb/card.h
+++ b/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 */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 9bea7d3f99f8..88760268fb55 100644
--- a/sound/usb/endpoint.c
+++ b/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;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a777d36c4f5a..40b7cd13fed9 100644
--- a/sound/usb/pcm.c
+++ b/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;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 4ec491011b19..9092cc0aa807 100644
--- a/sound/usb/quirks-table.h
+++ b/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