ALSA: usb-audio: Fix UAC1 rate setup for secondary endpoints

The current sample rate setup function for UAC1 assumes only the first
endpoint retrieved from the interface:altset pair, but the rate set up
may be needed also for the secondary endpoint.  Also, retrieving the
endpoint number from the interface descriptor is redundant; we have
already the target endpoint in the given audioformat object.

This patch simplifies the code and corrects the target endpoint as
described in the above.  It simply refers to fmt->endpoint directly.

Also, this patch drops the pioneer_djm_set_format_quirk() that is
caleld from snd_usb_set_format_quirk(); this function does the sample
rate setup but for the capture endpoint (0x82), and that's exactly
what the change above fixes.

Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

sound/usb/clock.c

@@ -485,18 +485,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip,
 			      const struct audioformat *fmt, int rate)
 {
 	struct usb_device *dev = chip->dev;
-	struct usb_host_interface *alts;
-	unsigned int ep;
 	unsigned char data[3];
 	int err, crate;
 
-	alts = snd_usb_get_host_interface(chip, fmt->iface, fmt->altsetting);
-	if (!alts)
-		return -EINVAL;
-	if (get_iface_desc(alts)->bNumEndpoints < 1)
-		return -EINVAL;
-	ep = get_endpoint(alts, 0)->bEndpointAddress;
-
 	/* if endpoint doesn't have sampling rate control, bail out */
 	if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE))
 		return 0;
@@ -506,11 +497,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip,
 	data[2] = rate >> 16;
 	err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
 			      USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
-			      UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-			      data, sizeof(data));
+			      UAC_EP_CS_ATTR_SAMPLE_RATE << 8,
+			      fmt->endpoint, data, sizeof(data));
 	if (err < 0) {
 		dev_err(&dev->dev, "%d:%d: cannot set freq %d to ep %#x\n",
-			fmt->iface, fmt->altsetting, rate, ep);
+			fmt->iface, fmt->altsetting, rate, fmt->endpoint);
 		return err;
 	}
 
@@ -524,11 +515,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip,
 
 	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
 			      USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
-			      UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-			      data, sizeof(data));
+			      UAC_EP_CS_ATTR_SAMPLE_RATE << 8,
+			      fmt->endpoint, data, sizeof(data));
 	if (err < 0) {
 		dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n",
-			fmt->iface, fmt->altsetting, ep);
+			fmt->iface, fmt->altsetting, fmt->endpoint);
 		chip->sample_rate_read_error++;
 		return 0; /* some devices don't support reading */
 	}

sound/usb/quirks.c

@@ -1470,30 +1470,6 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs,
 	subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0;
 }
 
-
-/*
- * Pioneer DJ DJM-900NXS2
- * Device needs to know the sample rate each time substream is started
- */
-static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs)
-{
-	unsigned int cur_rate = subs->data_endpoint->cur_rate;
-	/* Convert sample rate value to little endian */
-	u8 sr[3];
-
-	sr[0] = cur_rate & 0xff;
-	sr[1] = (cur_rate >> 8) & 0xff;
-	sr[2] = (cur_rate >> 16) & 0xff;
-
-	/* Configure device */
-	usb_set_interface(subs->dev, 0, 1);
-	snd_usb_ctl_msg(subs->stream->chip->dev,
-		usb_rcvctrlpipe(subs->stream->chip->dev, 0),
-		0x01, 0x22, 0x0100, 0x0082, &sr, 0x0003);
-
-	return 0;
-}
-
 void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 			      const struct audioformat *fmt)
 {
@@ -1504,10 +1480,6 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
 		set_format_emu_quirk(subs, fmt);
 		break;
-	case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
-	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
-		pioneer_djm_set_format_quirk(subs);
-		break;
 	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
 		subs->stream_offset_adj = 2;
 		break;