Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Mostly HD-audio and USB-audio regression fixes:
   - Oops fix at unloading of snd-hda-codec-conexant module
   - A few trivial regression fixes for Cirrus and Conexant HD-audio
     codecs
   - Relax the USB-audio descriptor parse errors as non-fatal
   - Fix locking of HD-audio CA0132 DSP loader
   - Fix the generic HD-audio parser for VIA codecs"

* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix DAC assignment for independent HP
  ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
  ALSA: hda - Fix typo in checking IEC958 emphasis bit
  ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
  ALSA: snd-usb: mixer: propagate errors up the call chain
  ALSA: usb: Parse UAC2 extension unit like for UAC1
  ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
  ALSA: hda/cirrus - Fix the digital beep registration
  ALSA: hda - Fix missing beep detach in patch_conexant.c
  ALSA: documentation: Fix typo in Documentation/sound

8 files changed, 184 insertions, 41 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index ce6581c8ca26..4499bd948860 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -912,7 +912,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     models depending on the codec chip.  The list of available models
     is found in HD-Audio-Models.txt
 
-    The model name "genric" is treated as a special case.  When this
+    The model name "generic" is treated as a special case.  When this
     model is given, the driver uses the generic codec parser without
     "codec-patch".  It's sometimes good for testing and debugging.
 
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
index d9776cf60c07..9663b45f6fde 100644
--- a/Documentation/sound/alsa/seq_oss.html
+++ b/Documentation/sound/alsa/seq_oss.html
@@ -285,7 +285,7 @@ sample data.
 <H4>
 7.2.4 Close Callback</H4>
 The <TT>close</TT> callback is called when this device is closed by the
-applicaion. If any private data was allocated in open callback, it must
+application. If any private data was allocated in open callback, it must
 be released in the close callback. The deletion of ALSA port should be
 done here, too. This callback must not be NULL.
 <H4>
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a9ebcf9e3710..ecdf30eb5879 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3144,7 +3144,7 @@ static unsigned int convert_to_spdif_status(unsigned short val)
 	if (val & AC_DIG1_PROFESSIONAL)
 		sbits |= IEC958_AES0_PROFESSIONAL;
 	if (sbits & IEC958_AES0_PROFESSIONAL) {
-		if (sbits & AC_DIG1_EMPHASIS)
+		if (val & AC_DIG1_EMPHASIS)
 			sbits |= IEC958_AES0_PRO_EMPHASIS_5015;
 	} else {
 		if (val & AC_DIG1_EMPHASIS)
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 78897d05d80f..43c2ea539561 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -995,6 +995,8 @@ enum {
 	BAD_NO_EXTRA_SURR_DAC = 0x101,
 	/* Primary DAC shared with main surrounds */
 	BAD_SHARED_SURROUND = 0x100,
+	/* No independent HP possible */
+	BAD_NO_INDEP_HP = 0x40,
 	/* Primary DAC shared with main CLFE */
 	BAD_SHARED_CLFE = 0x10,
 	/* Primary DAC shared with extra surrounds */
@@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx)
 	return snd_hda_get_path_idx(codec, path);
 }
 
+/* check whether the independent HP is available with the current config */
+static bool indep_hp_possible(struct hda_codec *codec)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct nid_path *path;
+	int i, idx;
+
+	if (cfg->line_out_type == AUTO_PIN_HP_OUT)
+		idx = spec->out_paths[0];
+	else
+		idx = spec->hp_paths[0];
+	path = snd_hda_get_path_from_idx(codec, idx);
+	if (!path)
+		return false;
+
+	/* assume no path conflicts unless aamix is involved */
+	if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid))
+		return true;
+
+	/* check whether output paths contain aamix */
+	for (i = 0; i < cfg->line_outs; i++) {
+		if (spec->out_paths[i] == idx)
+			break;
+		path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]);
+		if (path && is_nid_contained(path, spec->mixer_nid))
+			return false;
+	}
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]);
+		if (path && is_nid_contained(path, spec->mixer_nid))
+			return false;
+	}
+
+	return true;
+}
+
 /* fill the empty entries in the dac array for speaker/hp with the
  * shared dac pointed by the paths
  */
@@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 		badness += BAD_MULTI_IO;
 	}
 
+	if (spec->indep_hp && !indep_hp_possible(codec))
+		badness += BAD_NO_INDEP_HP;
+
 	/* re-fill the shared DAC for speaker / headphone */
 	if (cfg->line_out_type != AUTO_PIN_HP_OUT)
 		refill_shared_dacs(codec, cfg->hp_outs,
@@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec)
 				cfg->speaker_pins, val);
 	}
 
+	/* clear indep_hp flag if not available */
+	if (spec->indep_hp && !indep_hp_possible(codec))
+		spec->indep_hp = 0;
+
 	kfree(best_cfg);
 	return 0;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4cea6bb6fade..418bfc0eb0a3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -415,6 +415,8 @@ struct azx_dev {
 	unsigned int opened :1;
 	unsigned int running :1;
 	unsigned int irq_pending :1;
+	unsigned int prepared:1;
+	unsigned int locked:1;
 	/*
 	 * For VIA:
 	 *  A flag to ensure DMA position is 0
@@ -426,8 +428,25 @@ struct azx_dev {
 
 	struct timecounter  azx_tc;
 	struct cyclecounter azx_cc;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+	struct mutex dsp_mutex;
+#endif
 };
 
+/* DSP lock helpers */
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+#define dsp_lock_init(dev)	mutex_init(&(dev)->dsp_mutex)
+#define dsp_lock(dev)		mutex_lock(&(dev)->dsp_mutex)
+#define dsp_unlock(dev)		mutex_unlock(&(dev)->dsp_mutex)
+#define dsp_is_locked(dev)	((dev)->locked)
+#else
+#define dsp_lock_init(dev)	do {} while (0)
+#define dsp_lock(dev)		do {} while (0)
+#define dsp_unlock(dev)		do {} while (0)
+#define dsp_is_locked(dev)	0
+#endif
+
 /* CORB/RIRB */
 struct azx_rb {
 	u32 *buf;		/* CORB/RIRB buffer
@@ -527,6 +546,10 @@ struct azx {
 
 	/* card list (for power_save trigger) */
 	struct list_head list;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+	struct azx_dev saved_azx_dev;
+#endif
 };
 
 #define CREATE_TRACE_POINTS
@@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
 		dev = chip->capture_index_offset;
 		nums = chip->capture_streams;
 	}
-	for (i = 0; i < nums; i++, dev++)
-		if (!chip->azx_dev[dev].opened) {
-			res = &chip->azx_dev[dev];
-			if (res->assigned_key == key)
-				break;
+	for (i = 0; i < nums; i++, dev++) {
+		struct azx_dev *azx_dev = &chip->azx_dev[dev];
+		dsp_lock(azx_dev);
+		if (!azx_dev->opened && !dsp_is_locked(azx_dev)) {
+			res = azx_dev;
+			if (res->assigned_key == key) {
+				res->opened = 1;
+				res->assigned_key = key;
+				dsp_unlock(azx_dev);
+				return azx_dev;
+			}
 		}
+		dsp_unlock(azx_dev);
+	}
 	if (res) {
+		dsp_lock(res);
 		res->opened = 1;
 		res->assigned_key = key;
+		dsp_unlock(res);
 	}
 	return res;
 }
@@ -2009,6 +2042,12 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct azx_dev *azx_dev = get_azx_dev(substream);
 	int ret;
 
+	dsp_lock(azx_dev);
+	if (dsp_is_locked(azx_dev)) {
+		ret = -EBUSY;
+		goto unlock;
+	}
+
 	mark_runtime_wc(chip, azx_dev, substream, false);
 	azx_dev->bufsize = 0;
 	azx_dev->period_bytes = 0;
@@ -2016,8 +2055,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
 	ret = snd_pcm_lib_malloc_pages(substream,
 					params_buffer_bytes(hw_params));
 	if (ret < 0)
-		return ret;
+		goto unlock;
 	mark_runtime_wc(chip, azx_dev, substream, true);
+ unlock:
+	dsp_unlock(azx_dev);
 	return ret;
 }
 
@@ -2029,16 +2070,21 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
 
 	/* reset BDL address */
-	azx_sd_writel(azx_dev, SD_BDLPL, 0);
-	azx_sd_writel(azx_dev, SD_BDLPU, 0);
-	azx_sd_writel(azx_dev, SD_CTL, 0);
-	azx_dev->bufsize = 0;
-	azx_dev->period_bytes = 0;
-	azx_dev->format_val = 0;
+	dsp_lock(azx_dev);
+	if (!dsp_is_locked(azx_dev)) {
+		azx_sd_writel(azx_dev, SD_BDLPL, 0);
+		azx_sd_writel(azx_dev, SD_BDLPU, 0);
+		azx_sd_writel(azx_dev, SD_CTL, 0);
+		azx_dev->bufsize = 0;
+		azx_dev->period_bytes = 0;
+		azx_dev->format_val = 0;
+	}
 
 	snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
 
 	mark_runtime_wc(chip, azx_dev, substream, false);
+	azx_dev->prepared = 0;
+	dsp_unlock(azx_dev);
 	return snd_pcm_lib_free_pages(substream);
 }
 
@@ -2055,6 +2101,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 		snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
 	unsigned short ctls = spdif ? spdif->ctls : 0;
 
+	dsp_lock(azx_dev);
+	if (dsp_is_locked(azx_dev)) {
+		err = -EBUSY;
+		goto unlock;
+	}
+
 	azx_stream_reset(chip, azx_dev);
 	format_val = snd_hda_calc_stream_format(runtime->rate,
 						runtime->channels,
@@ -2065,7 +2117,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 		snd_printk(KERN_ERR SFX
 			   "%s: invalid format_val, rate=%d, ch=%d, format=%d\n",
 			   pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format);
-		return -EINVAL;
+		err = -EINVAL;
+		goto unlock;
 	}
 
 	bufsize = snd_pcm_lib_buffer_bytes(substream);
@@ -2084,7 +2137,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 		azx_dev->no_period_wakeup = runtime->no_period_wakeup;
 		err = azx_setup_periods(chip, substream, azx_dev);
 		if (err < 0)
-			return err;
+			goto unlock;
 	}
 
 	/* wallclk has 24Mhz clock source */
@@ -2101,8 +2154,14 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 	if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) &&
 	    stream_tag > chip->capture_streams)
 		stream_tag -= chip->capture_streams;
-	return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
+	err = snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
 				     azx_dev->format_val, substream);
+
+ unlock:
+	if (!err)
+		azx_dev->prepared = 1;
+	dsp_unlock(azx_dev);
+	return err;
 }
 
 static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -2117,6 +2176,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	azx_dev = get_azx_dev(substream);
 	trace_azx_pcm_trigger(chip, azx_dev, cmd);
 
+	if (dsp_is_locked(azx_dev) || !azx_dev->prepared)
+		return -EPIPE;
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		rstart = 1;
@@ -2621,17 +2683,27 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format,
 	struct azx_dev *azx_dev;
 	int err;
 
-	if (snd_hda_lock_devices(bus))
-		return -EBUSY;
+	azx_dev = azx_get_dsp_loader_dev(chip);
+
+	dsp_lock(azx_dev);
+	spin_lock_irq(&chip->reg_lock);
+	if (azx_dev->running || azx_dev->locked) {
+		spin_unlock_irq(&chip->reg_lock);
+		err = -EBUSY;
+		goto unlock;
+	}
+	azx_dev->prepared = 0;
+	chip->saved_azx_dev = *azx_dev;
+	azx_dev->locked = 1;
+	spin_unlock_irq(&chip->reg_lock);
 
 	err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG,
 				  snd_dma_pci_data(chip->pci),
 				  byte_size, bufp);
 	if (err < 0)
-		goto unlock;
+		goto err_alloc;
 
 	mark_pages_wc(chip, bufp, true);
-	azx_dev = azx_get_dsp_loader_dev(chip);
 	azx_dev->bufsize = byte_size;
 	azx_dev->period_bytes = byte_size;
 	azx_dev->format_val = format;
@@ -2649,13 +2721,20 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format,
 		goto error;
 
 	azx_setup_controller(chip, azx_dev);
+	dsp_unlock(azx_dev);
 	return azx_dev->stream_tag;
 
  error:
 	mark_pages_wc(chip, bufp, false);
 	snd_dma_free_pages(bufp);
-unlock:
-	snd_hda_unlock_devices(bus);
+ err_alloc:
+	spin_lock_irq(&chip->reg_lock);
+	if (azx_dev->opened)
+		*azx_dev = chip->saved_azx_dev;
+	azx_dev->locked = 0;
+	spin_unlock_irq(&chip->reg_lock);
+ unlock:
+	dsp_unlock(azx_dev);
 	return err;
 }
 
@@ -2677,9 +2756,10 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus,
 	struct azx *chip = bus->private_data;
 	struct azx_dev *azx_dev = azx_get_dsp_loader_dev(chip);
 
-	if (!dmab->area)
+	if (!dmab->area || !azx_dev->locked)
 		return;
 
+	dsp_lock(azx_dev);
 	/* reset BDL address */
 	azx_sd_writel(azx_dev, SD_BDLPL, 0);
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
@@ -2692,7 +2772,12 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus,
 	snd_dma_free_pages(dmab);
 	dmab->area = NULL;
 
-	snd_hda_unlock_devices(bus);
+	spin_lock_irq(&chip->reg_lock);
+	if (azx_dev->opened)
+		*azx_dev = chip->saved_azx_dev;
+	azx_dev->locked = 0;
+	spin_unlock_irq(&chip->reg_lock);
+	dsp_unlock(azx_dev);
 }
 #endif /* CONFIG_SND_HDA_DSP_LOADER */
 
@@ -3481,6 +3566,7 @@ static int azx_first_init(struct azx *chip)
 	}
 
 	for (i = 0; i < chip->num_streams; i++) {
+		dsp_lock_init(&chip->azx_dev[i]);
 		/* allocate memory for the BDL for each stream */
 		err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
 					  snd_dma_pci_data(chip->pci),
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 60d08f669f0c..0d9c58f13560 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -168,10 +168,10 @@ static void cs_automute(struct hda_codec *codec)
 	snd_hda_gen_update_outputs(codec);
 
 	if (spec->gpio_eapd_hp) {
-		unsigned int gpio = spec->gen.hp_jack_present ?
+		spec->gpio_data = spec->gen.hp_jack_present ?
 			spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
 		snd_hda_codec_write(codec, 0x01, 0,
-				    AC_VERB_SET_GPIO_DATA, gpio);
+				    AC_VERB_SET_GPIO_DATA, spec->gpio_data);
 	}
 }
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 941bf6c766ec..2a89d1eefeb6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	}
 
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	return 0;
 }
@@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 	}
 
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	return 0;
 }
@@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 	}
 
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	return 0;
 }
@@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
+static void cx_auto_free(struct hda_codec *codec)
+{
+	snd_hda_detach_beep_device(codec);
+	snd_hda_gen_free(codec);
+}
+
 static const struct hda_codec_ops cx_auto_patch_ops = {
 	.build_controls = cx_auto_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = snd_hda_gen_init,
-	.free = snd_hda_gen_free,
+	.free = cx_auto_free,
 	.unsol_event = snd_hda_jack_unsol_event,
 #ifdef CONFIG_PM
 	.check_power_status = snd_hda_gen_check_power_status,
@@ -3391,7 +3397,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	codec->patch_ops = cx_auto_patch_ops;
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	/* Some laptops with Conexant chips show stalls in S3 resume,
 	 * which falls into the single-cmd mode.
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 638e7f738018..ca4739c3f650 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
 		case UAC2_CLOCK_SELECTOR: {
 			struct uac_selector_unit_descriptor *d = p1;
 			/* call recursively to retrieve the channel info */
-			if (check_input_term(state, d->baSourceID[0], term) < 0)
-				return -ENODEV;
+			err = check_input_term(state, d->baSourceID[0], term);
+			if (err < 0)
+				return err;
 			term->type = d->bDescriptorSubtype << 16; /* virtual type */
 			term->id = id;
 			term->name = uac_selector_unit_iSelector(d);
@@ -725,7 +726,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
 		case UAC1_PROCESSING_UNIT:
 		case UAC1_EXTENSION_UNIT:
 		/* UAC2_PROCESSING_UNIT_V2 */
-		/* UAC2_EFFECT_UNIT */ {
+		/* UAC2_EFFECT_UNIT */
+		case UAC2_EXTENSION_UNIT_V2: {
 			struct uac_processing_unit_descriptor *d = p1;
 
 			if (state->mixer->protocol == UAC_VERSION_2 &&
@@ -1356,8 +1358,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 		return err;
 
 	/* determine the input source type and name */
-	if (check_input_term(state, hdr->bSourceID, &iterm) < 0)
-		return -EINVAL;
+	err = check_input_term(state, hdr->bSourceID, &iterm);
+	if (err < 0)
+		return err;
 
 	master_bits = snd_usb_combine_bytes(bmaControls, csize);
 	/* master configuration quirks */
@@ -2052,6 +2055,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 			return parse_audio_extension_unit(state, unitid, p1);
 		else /* UAC_VERSION_2 */
 			return parse_audio_processing_unit(state, unitid, p1);
+	case UAC2_EXTENSION_UNIT_V2:
+		return parse_audio_extension_unit(state, unitid, p1);
 	default:
 		snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
 		return -EINVAL;
@@ -2118,7 +2123,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 			state.oterm.type = le16_to_cpu(desc->wTerminalType);
 			state.oterm.name = desc->iTerminal;
 			err = parse_audio_unit(&state, desc->bSourceID);
-			if (err < 0)
+			if (err < 0 && err != -EINVAL)
 				return err;
 		} else { /* UAC_VERSION_2 */
 			struct uac2_output_terminal_descriptor *desc = p;
@@ -2130,12 +2135,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 			state.oterm.type = le16_to_cpu(desc->wTerminalType);
 			state.oterm.name = desc->iTerminal;
 			err = parse_audio_unit(&state, desc->bSourceID);
-			if (err < 0)
+			if (err < 0 && err != -EINVAL)
 				return err;
 
 			/* for UAC2, use the same approach to also add the clock selectors */
 			err = parse_audio_unit(&state, desc->bCSourceID);
-			if (err < 0)
+			if (err < 0 && err != -EINVAL)
 				return err;
 		}
 	}