Merge remote-tracking branch 'remotes/fslc/4.9-2.3.x-imx' into toradex_4.9-2.3.x-imx-nextColibri-iMX7_LXDE-Image_2.8b6.184-20190401Colibri-iMX6_LXDE-Image_2.8b6.184-20190401Colibri-iMX6ULL_LXDE-Image_2.8b6.184-20190401Apalis-iMX6_LXDE-Image_2.8b6.184-20190401

34 files changed, 290 insertions, 88 deletions

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 4490a699030b..555df64d46ff 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -529,7 +529,8 @@ static int snd_compress_check_input(struct snd_compr_params *params)
 {
 	/* first let's check the buffer parameter's */
 	if (params->buffer.fragment_size == 0 ||
-	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size)
+	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size ||
+	    params->buffer.fragments == 0)
 		return -EINVAL;
 
 	/* now codec parameters */
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 6bda8f6c5f84..cdff5f976480 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -25,6 +25,7 @@
 #include <linux/time.h>
 #include <linux/mutex.h>
 #include <linux/device.h>
+#include <linux/nospec.h>
 #include <sound/core.h>
 #include <sound/minors.h>
 #include <sound/pcm.h>
@@ -125,6 +126,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card,
 				return -EFAULT;
 			if (stream < 0 || stream > 1)
 				return -EINVAL;
+			stream = array_index_nospec(stream, 2);
 			if (get_user(subdevice, &info->subdevice))
 				return -EFAULT;
 			mutex_lock(&register_mutex);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 79018697b477..3586ab41dec4 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -35,6 +35,7 @@
 #include <sound/timer.h>
 #include <sound/minors.h>
 #include <linux/uio.h>
+#include <linux/delay.h>
 
 /*
  *  Compatibility
@@ -78,12 +79,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem);
  * and this may lead to a deadlock when the code path takes read sem
  * twice (e.g. one in snd_pcm_action_nonatomic() and another in
  * snd_pcm_stream_lock()).  As a (suboptimal) workaround, let writer to
- * spin until it gets the lock.
+ * sleep until all the readers are completed without blocking by writer.
  */
-static inline void down_write_nonblock(struct rw_semaphore *lock)
+static inline void down_write_nonfifo(struct rw_semaphore *lock)
 {
 	while (!down_write_trylock(lock))
-		cond_resched();
+		msleep(1);
 }
 
 /**
@@ -1825,7 +1826,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 		res = -ENOMEM;
 		goto _nolock;
 	}
-	down_write_nonblock(&snd_pcm_link_rwsem);
+	down_write_nonfifo(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
 	    substream->runtime->status->state != substream1->runtime->status->state ||
@@ -1872,7 +1873,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
 	struct snd_pcm_substream *s;
 	int res = 0;
 
-	down_write_nonblock(&snd_pcm_link_rwsem);
+	down_write_nonfifo(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (!snd_pcm_stream_linked(substream)) {
 		res = -EALREADY;
@@ -2224,7 +2225,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
 
 static void pcm_release_private(struct snd_pcm_substream *substream)
 {
-	snd_pcm_unlink(substream);
+	if (snd_pcm_stream_linked(substream))
+		snd_pcm_unlink(substream);
 }
 
 void snd_pcm_release_substream(struct snd_pcm_substream *substream)
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index ab894ed1ff67..8557e54d2659 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -40,6 +40,7 @@ config SND_OXFW
 	   * Mackie(Loud) U.420/U.420d
 	   * TASCAM FireOne
 	   * Stanton Controllers & Systems 1 Deck/Mixer
+	   * APOGEE duet FireWire
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-oxfw.
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index d0dfa822266b..a205b93fd9ac 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -434,7 +434,7 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	/* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */
 	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal),
 	/* Apogee Electronics, Ensemble */
-	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal),
+	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal),
 	/* ESI, Quatafire610 */
 	SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal),
 	/* AcousticReality, eARMasterOne */
@@ -474,7 +474,19 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	/* Focusrite, SaffirePro 26 I/O */
 	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec),
 	/* Focusrite, SaffirePro 10 I/O */
-	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec),
+	{
+		// The combination of vendor_id and model_id is the same as the
+		// same as the one of Liquid Saffire 56.
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID |
+				  IEEE1394_MATCH_SPECIFIER_ID |
+				  IEEE1394_MATCH_VERSION,
+		.vendor_id	= VEN_FOCUSRITE,
+		.model_id	= 0x000006,
+		.specifier_id	= 0x00a02d,
+		.version	= 0x010001,
+		.driver_data	= (kernel_ulong_t)&saffirepro_10_spec,
+	},
 	/* Focusrite, Saffire(no label and LE) */
 	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH,
 			    &saffire_spec),
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 696b6cf35003..b0395c4209ab 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -20,6 +20,7 @@
 #define VENDOR_LACIE		0x00d04b
 #define VENDOR_TASCAM		0x00022e
 #define OUI_STANTON		0x001260
+#define OUI_APOGEE		0x0003db
 
 #define MODEL_SATELLITE		0x00200f
 
@@ -441,6 +442,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = {
 		.vendor_id	= OUI_STANTON,
 		.model_id	= 0x002000,
 	},
+	// APOGEE, duet FireWire
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_APOGEE,
+		.model_id	= 0x01dddd,
+	},
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table);
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index 4a0cbd2241d8..3191666ac129 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -899,6 +899,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
 	int i;
 
+	if (!ins)
+		return 0;
+
 	snd_info_free_entry(ins->proc_sym_info_entry);
 	ins->proc_sym_info_entry = NULL;
 
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 50b216fc369f..5d422d65e62b 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -36,6 +36,7 @@
 #include <linux/init.h>
 #include <linux/mutex.h>
 #include <linux/moduleparam.h>
+#include <linux/nospec.h>
 
 #include <sound/core.h>
 #include <sound/tlv.h>
@@ -1000,6 +1001,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu,
 
 	if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
 		return -EINVAL;
+	ipcm->substream = array_index_nospec(ipcm->substream,
+					     EMU10K1_FX8010_PCM_COUNT);
 	if (ipcm->channels > 32)
 		return -EINVAL;
 	pcm = &emu->fx8010.pcm[ipcm->substream];
@@ -1046,6 +1049,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu,
 
 	if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
 		return -EINVAL;
+	ipcm->substream = array_index_nospec(ipcm->substream,
+					     EMU10K1_FX8010_PCM_COUNT);
 	pcm = &emu->fx8010.pcm[ipcm->substream];
 	mutex_lock(&emu->fx8010.lock);
 	spin_lock_irq(&emu->reg_lock);
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 6efadbfb3fe3..7ea201c05e5d 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -109,7 +109,8 @@ static int hda_codec_driver_probe(struct device *dev)
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
 		goto error_module;
-	if (codec->card->registered) {
+	/* only register after the bus probe finished; otherwise it's racy */
+	if (!codec->bus->bus_probing && codec->card->registered) {
 		err = snd_card_register(codec->card);
 		if (err < 0)
 			goto error_module;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c6b046ddefdd..1b5e217d1bb2 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3004,6 +3004,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 		hda_jackpoll_work(&codec->jackpoll_work.work);
 	else
 		snd_hda_jack_report_sync(codec);
+	codec->core.dev.power.power_state = PMSG_ON;
 	atomic_dec(&codec->core.in_pm);
 }
 
@@ -3036,10 +3037,62 @@ static int hda_codec_runtime_resume(struct device *dev)
 }
 #endif /* CONFIG_PM */
 
+#ifdef CONFIG_PM_SLEEP
+static int hda_codec_force_resume(struct device *dev)
+{
+	int ret;
+
+	/* The get/put pair below enforces the runtime resume even if the
+	 * device hasn't been used at suspend time.  This trick is needed to
+	 * update the jack state change during the sleep.
+	 */
+	pm_runtime_get_noresume(dev);
+	ret = pm_runtime_force_resume(dev);
+	pm_runtime_put(dev);
+	return ret;
+}
+
+static int hda_codec_pm_suspend(struct device *dev)
+{
+	dev->power.power_state = PMSG_SUSPEND;
+	return pm_runtime_force_suspend(dev);
+}
+
+static int hda_codec_pm_resume(struct device *dev)
+{
+	dev->power.power_state = PMSG_RESUME;
+	return hda_codec_force_resume(dev);
+}
+
+static int hda_codec_pm_freeze(struct device *dev)
+{
+	dev->power.power_state = PMSG_FREEZE;
+	return pm_runtime_force_suspend(dev);
+}
+
+static int hda_codec_pm_thaw(struct device *dev)
+{
+	dev->power.power_state = PMSG_THAW;
+	return hda_codec_force_resume(dev);
+}
+
+static int hda_codec_pm_restore(struct device *dev)
+{
+	dev->power.power_state = PMSG_RESTORE;
+	return hda_codec_force_resume(dev);
+}
+#endif /* CONFIG_PM_SLEEP */
+
 /* referred in hda_bind.c */
 const struct dev_pm_ops hda_codec_driver_pm = {
-	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				pm_runtime_force_resume)
+#ifdef CONFIG_PM_SLEEP
+	.suspend = hda_codec_pm_suspend,
+	.resume = hda_codec_pm_resume,
+	.freeze = hda_codec_pm_freeze,
+	.thaw = hda_codec_pm_thaw,
+	.poweroff = hda_codec_pm_suspend,
+	.restore = hda_codec_pm_restore,
+#endif /* CONFIG_PM_SLEEP */
 	SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume,
 			   NULL)
 };
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 776dffa88aee..171e11be938d 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -68,6 +68,7 @@ struct hda_bus {
 	unsigned int response_reset:1;	/* controller was reset */
 	unsigned int in_reset:1;	/* during reset operation */
 	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
+	unsigned int bus_probing :1;	/* during probing process */
 
 	int primary_dig_out_type;	/* primary digital out PCM type */
 	unsigned int mixer_assigned;	/* codec addr for mixer name */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3557e3943ad5..789eca17fc60 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2089,6 +2089,7 @@ static int azx_probe_continue(struct azx *chip)
 	int val;
 	int err;
 
+	to_hda_bus(bus)->bus_probing = 1;
 	hda->probe_continued = 1;
 
 	/* Request display power well for the HDA controller or codec. For
@@ -2189,6 +2190,7 @@ i915_power_fail:
 	if (err < 0)
 		hda->init_failed = 1;
 	complete_all(&hda->probe_wait);
+	to_hda_bus(bus)->bus_probing = 0;
 	return err;
 }
 
@@ -2352,6 +2354,10 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD Stoney */
+	{ PCI_DEVICE(0x1022, 0x157a),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+			 AZX_DCAPS_PM_RUNTIME },
 	/* AMD Raven */
 	{ PCI_DEVICE(0x1022, 0x15e3),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0621920f7617..e85fb04ec7be 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -249,10 +249,12 @@ static int hda_tegra_suspend(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 	struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
+	struct hdac_bus *bus = azx_bus(chip);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 
 	azx_stop_chip(chip);
+	synchronize_irq(bus->irq);
 	azx_enter_link_reset(chip);
 	hda_tegra_disable_clocks(hda);
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d392e867e9ab..447b3a8a83c3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -853,6 +853,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
+	SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+	SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6c2668b4e3bc..0fc05ebdf81a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4489,9 +4489,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec,
 		{ 0x19, 0x21a11010 }, /* dock mic */
 		{ }
 	};
+	/* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
+	 * the speaker output becomes too low by some reason on Thinkpads with
+	 * ALC298 codec
+	 */
+	static hda_nid_t preferred_pairs[] = {
+		0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
+		0
+	};
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gen.preferred_dacs = preferred_pairs;
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		snd_hda_apply_pincfgs(codec, pincfgs);
 	} else if (action == HDA_FIXUP_ACT_INIT) {
@@ -4832,6 +4841,13 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec,
 	}
 }
 
+static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -4938,6 +4954,7 @@ enum {
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
 	ALC255_FIXUP_DELL_SPK_NOISE,
+	ALC225_FIXUP_DISABLE_MIC_VREF,
 	ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC295_FIXUP_DISABLE_DAC3,
 	ALC280_FIXUP_HP_HEADSET_MIC,
@@ -5596,6 +5613,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
+	[ALC225_FIXUP_DISABLE_MIC_VREF] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_mic_vref,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
 	[ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -5605,7 +5628,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{}
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+		.chain_id = ALC225_FIXUP_DISABLE_MIC_VREF
 	},
 	[ALC280_FIXUP_HP_HEADSET_MIC] = {
 		.type = HDA_FIXUP_FUNC,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b94fc6357139..b044dea3c815 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -30,6 +30,7 @@
 #include <linux/math64.h>
 #include <linux/vmalloc.h>
 #include <linux/io.h>
+#include <linux/nospec.h>
 
 #include <sound/core.h>
 #include <sound/control.h>
@@ -4065,15 +4066,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream,
 				    struct snd_pcm_channel_info *info)
 {
 	struct hdsp *hdsp = snd_pcm_substream_chip(substream);
-	int mapped_channel;
+	unsigned int channel = info->channel;
 
-	if (snd_BUG_ON(info->channel >= hdsp->max_channels))
+	if (snd_BUG_ON(channel >= hdsp->max_channels))
 		return -EINVAL;
+	channel = array_index_nospec(channel, hdsp->max_channels);
 
-	if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
+	if (hdsp->channel_map[channel] < 0)
 		return -EINVAL;
 
-	info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES;
+	info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES;
 	info->first = 0;
 	info->step = 32;
 	return 0;
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 09103aab0cb2..7d410e39d1a0 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -253,6 +253,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform)
 
 	rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp),
 			GFP_KERNEL);
+	if (!rt5514_dsp)
+		return -ENOMEM;
 
 	rt5514_dsp->dev = &rt5514_spi->dev;
 	mutex_init(&rt5514_dsp->dma_lock);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 0ba0bf13c3a9..887af25f7367 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -281,7 +281,7 @@ config SND_SOC_PHYCORE_AC97
 
 config SND_SOC_EUKREA_TLV320
 	tristate "Eukrea TLV320"
-	depends on ARCH_MXC && I2C
+	depends on ARCH_MXC && !ARM64 && I2C
 	select SND_SOC_TLV320AIC23_I2C
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_SSI
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 17766f8b09a8..f01f7be66b37 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -88,49 +88,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
-	ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+	ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
 		       pdcr, ptcr);
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS output from %s, ",
 				audmux_port_string((ptcr >> 27) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS input, ");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk output from %s",
 				audmux_port_string((ptcr >> 22) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk input");
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"Port is symmetric");
 	} else {
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS output from %s, ",
 					audmux_port_string((ptcr >> 17) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS input, ");
 
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk output from %s",
 					audmux_port_string((ptcr >> 12) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk input");
 	}
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret,
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 			"\nData received from %s\n",
 			audmux_port_string((pdcr >> 13) & 0x7));
 
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index f5a8050351b5..e83e314a76a5 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	int ret;
+
+	ret =
+		snd_pcm_lib_malloc_pages(substream,
+				params_buffer_bytes(params));
+	if (ret)
+		return ret;
 	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
 	return 0;
 }
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index 33917146d9c4..054b1d514e8a 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst)
 	const struct firmware *fw;
 
 	retval = request_firmware(&fw, sst->firmware_name, sst->dev);
-	if (fw == NULL) {
-		dev_err(sst->dev, "fw is returning as null\n");
-		return -EINVAL;
-	}
 	if (retval) {
 		dev_err(sst->dev, "request fw failed %d\n", retval);
 		return retval;
 	}
+	if (fw == NULL) {
+		dev_err(sst->dev, "fw is returning as null\n");
+		return -EINVAL;
+	}
 	mutex_lock(&sst->sst_lock);
 	retval = sst_cache_and_parse_fw(sst, fw);
 	mutex_unlock(&sst->sst_lock);
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 7486a0022fde..993d2c105ae1 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 863f1d5e2a2c..11d0cc2b0e39 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 89fe95e877db..07af30017b48 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -36,6 +36,8 @@
 #include "../codecs/twl6040.h"
 
 struct abe_twl6040 {
+	struct snd_soc_card card;
+	struct snd_soc_dai_link dai_links[2];
 	int	jack_detection;	/* board can detect jack events */
 	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
 };
@@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
 				ARRAY_SIZE(dmic_audio_map));
 }
 
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
-	{
-		.name = "TWL6040",
-		.stream_name = "TWL6040",
-		.codec_dai_name = "twl6040-legacy",
-		.codec_name = "twl6040-codec",
-		.init = omap_abe_twl6040_init,
-		.ops = &omap_abe_ops,
-	},
-	{
-		.name = "DMIC",
-		.stream_name = "DMIC Capture",
-		.codec_dai_name = "dmic-hifi",
-		.codec_name = "dmic-codec",
-		.init = omap_abe_dmic_init,
-		.ops = &omap_abe_dmic_ops,
-	},
-};
-
-/* Audio machine driver */
-static struct snd_soc_card omap_abe_card = {
-	.owner = THIS_MODULE,
-
-	.dapm_widgets = twl6040_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
-	.dapm_routes = audio_map,
-	.num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
 static int omap_abe_probe(struct platform_device *pdev)
 {
 	struct device_node *node = pdev->dev.of_node;
-	struct snd_soc_card *card = &omap_abe_card;
+	struct snd_soc_card *card;
 	struct device_node *dai_node;
 	struct abe_twl6040 *priv;
 	int num_links = 0;
@@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	card->dev = &pdev->dev;
-
 	priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
 	if (priv == NULL)
 		return -ENOMEM;
 
+	card = &priv->card;
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dapm_widgets = twl6040_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
+	card->dapm_routes = audio_map;
+	card->num_dapm_routes = ARRAY_SIZE(audio_map);
+
 	if (snd_soc_of_parse_card_name(card, "ti,model")) {
 		dev_err(&pdev->dev, "Card name is not provided\n");
 		return -ENODEV;
@@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "McPDM node is not provided\n");
 		return -EINVAL;
 	}
-	abe_twl6040_dai_links[0].cpu_of_node = dai_node;
-	abe_twl6040_dai_links[0].platform_of_node = dai_node;
+
+	priv->dai_links[0].name = "DMIC";
+	priv->dai_links[0].stream_name = "TWL6040";
+	priv->dai_links[0].cpu_of_node = dai_node;
+	priv->dai_links[0].platform_of_node = dai_node;
+	priv->dai_links[0].codec_dai_name = "twl6040-legacy";
+	priv->dai_links[0].codec_name = "twl6040-codec";
+	priv->dai_links[0].init = omap_abe_twl6040_init;
+	priv->dai_links[0].ops = &omap_abe_ops;
 
 	dai_node = of_parse_phandle(node, "ti,dmic", 0);
 	if (dai_node) {
 		num_links = 2;
-		abe_twl6040_dai_links[1].cpu_of_node = dai_node;
-		abe_twl6040_dai_links[1].platform_of_node = dai_node;
+		priv->dai_links[1].name = "TWL6040";
+		priv->dai_links[1].stream_name = "DMIC Capture";
+		priv->dai_links[1].cpu_of_node = dai_node;
+		priv->dai_links[1].platform_of_node = dai_node;
+		priv->dai_links[1].codec_dai_name = "dmic-hifi";
+		priv->dai_links[1].codec_name = "dmic-codec";
+		priv->dai_links[1].init = omap_abe_dmic_init;
+		priv->dai_links[1].ops = &omap_abe_dmic_ops;
 	} else {
 		num_links = 1;
 	}
@@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	card->dai_link = abe_twl6040_dai_links;
+	card->dai_link = priv->dai_links;
 	card->num_links = num_links;
 
 	snd_soc_card_set_drvdata(card, priv);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 09db2aec12a3..776e809a8aab 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -48,6 +48,8 @@ struct omap_dmic {
 	struct device *dev;
 	void __iomem *io_base;
 	struct clk *fclk;
+	struct pm_qos_request pm_qos_req;
+	int latency;
 	int fclk_freq;
 	int out_freq;
 	int clk_div;
@@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
 
 	mutex_lock(&dmic->mutex);
 
+	pm_qos_remove_request(&dmic->pm_qos_req);
+
 	if (!dai->active)
 		dmic->active = 0;
 
@@ -226,6 +230,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	/* packet size is threshold * channels */
 	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 	dma_data->maxburst = dmic->threshold * channels;
+	dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+			params_rate(params);
 
 	return 0;
 }
@@ -236,6 +242,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
 	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
 	u32 ctrl;
 
+	if (pm_qos_request_active(&dmic->pm_qos_req))
+		pm_qos_update_request(&dmic->pm_qos_req, dmic->latency);
+
 	/* Configure uplink threshold */
 	omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
 
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 64609c77a79d..44ffeb71cd1d 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -54,6 +54,8 @@ struct omap_mcpdm {
 	unsigned long phys_base;
 	void __iomem *io_base;
 	int irq;
+	struct pm_qos_request pm_qos_req;
+	int latency[2];
 
 	struct mutex mutex;
 
@@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock(&mcpdm->mutex);
 
@@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 		}
 	}
 
+	if (mcpdm->latency[stream2])
+		pm_qos_update_request(&mcpdm->pm_qos_req,
+				      mcpdm->latency[stream2]);
+	else if (mcpdm->latency[stream1])
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
+	mcpdm->latency[stream1] = 0;
+
 	mutex_unlock(&mcpdm->mutex);
 }
 
@@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 	int stream = substream->stream;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	u32 threshold;
-	int channels;
+	int channels, latency;
 	int link_mask = 0;
 
 	channels = params_channels(params);
@@ -340,14 +353,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 
 		dma_data->maxburst =
 				(MCPDM_DN_THRES_MAX - threshold) * channels;
+		latency = threshold;
 	} else {
 		/* If playback is not running assume a stereo stream to come */
 		if (!mcpdm->config[!stream].link_mask)
 			mcpdm->config[!stream].link_mask = (0x3 << 3);
 
 		dma_data->maxburst = threshold * channels;
+		latency = (MCPDM_DN_THRES_MAX - threshold);
 	}
 
+	/*
+	 * The DMA must act to a DMA request within latency time (usec) to avoid
+	 * under/overflow
+	 */
+	mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+	if (!mcpdm->latency[stream])
+		mcpdm->latency[stream] = 10;
+
 	/* Check if we need to restart McPDM with this stream */
 	if (mcpdm->config[stream].link_mask &&
 	    mcpdm->config[stream].link_mask != link_mask)
@@ -362,6 +386,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+	int latency = mcpdm->latency[stream2];
+
+	/* Prevent omap hardware from hitting off between FIFO fills */
+	if (!latency || mcpdm->latency[stream1] < latency)
+		latency = mcpdm->latency[stream1];
+
+	if (pm_qos_request_active(pm_qos_req))
+		pm_qos_update_request(pm_qos_req, latency);
+	else if (latency)
+		pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency);
 
 	if (!omap_mcpdm_active(mcpdm)) {
 		omap_mcpdm_start(mcpdm);
@@ -423,6 +461,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
 	free_irq(mcpdm->irq, (void *)mcpdm);
 	pm_runtime_disable(mcpdm->dev);
 
+	if (pm_qos_request_active(&mcpdm->pm_qos_req))
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
 	return 0;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e3de566809c..168559b5e9f3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2018,6 +2018,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	card->instantiated = 1;
+	dapm_mark_endpoints_dirty(card);
 	snd_soc_dapm_sync(&card->dapm);
 	mutex_unlock(&card->mutex);
 	mutex_unlock(&client_mutex);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8bfc534e3b34..ab647f1fe11b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1976,19 +1976,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 		out = is_connected_output_ep(w, NULL, NULL);
 	}
 
-	ret = snprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
+	ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
 		       w->name, w->power ? "On" : "Off",
 		       w->force ? " (forced)" : "", in, out);
 
 	if (w->reg >= 0)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				" - R%d(0x%x) mask 0x%x",
 				w->reg, w->reg, w->mask << w->shift);
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (w->sname)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
 				w->sname,
 				w->active ? "active" : "inactive");
 
@@ -2001,7 +2001,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 			if (!p->connect)
 				continue;
 
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					" %s  \"%s\" \"%s\"\n",
 					(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
 					p->name ? p->name : "static",
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index d6b48c796bfc..086fe4d27f60 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1989,6 +1989,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
 	struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id)
 {
 	struct soc_tplg tplg;
+	int ret;
 
 	/* setup parsing context */
 	memset(&tplg, 0, sizeof(tplg));
@@ -2002,7 +2003,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
 	tplg.bytes_ext_ops = ops->bytes_ext_ops;
 	tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count;
 
-	return soc_tplg_load(&tplg);
+	ret = soc_tplg_load(&tplg);
+	/* free the created components if fail to load topology */
+	if (ret)
+		snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load);
 
diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c
index e557946718a9..d9fcae071b47 100644
--- a/sound/synth/emux/emux_hwdep.c
+++ b/sound/synth/emux/emux_hwdep.c
@@ -22,9 +22,9 @@
 #include <sound/core.h>
 #include <sound/hwdep.h>
 #include <linux/uaccess.h>
+#include <linux/nospec.h>
 #include "emux_voice.h"
 
-
 #define TMP_CLIENT_ID	0x1001
 
 /*
@@ -66,13 +66,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg)
 		return -EFAULT;
 	if (info.mode < 0 || info.mode >= EMUX_MD_END)
 		return -EINVAL;
+	info.mode = array_index_nospec(info.mode, EMUX_MD_END);
 
 	if (info.port < 0) {
 		for (i = 0; i < emu->num_ports; i++)
 			emu->portptrs[i]->ctrls[info.mode] = info.value;
 	} else {
-		if (info.port < emu->num_ports)
+		if (info.port < emu->num_ports) {
+			info.port = array_index_nospec(info.port, emu->num_ports);
 			emu->portptrs[info.port]->ctrls[info.mode] = info.value;
+		}
 	}
 	return 0;
 }
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 8906199a83e6..549b9b061694 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -644,9 +644,12 @@ static int usb_audio_probe(struct usb_interface *intf,
 
  __error:
 	if (chip) {
+		/* chip->active is inside the chip->card object,
+		 * decrement before memory is possibly returned.
+		 */
+		atomic_dec(&chip->active);
 		if (!chip->num_interfaces)
 			snd_card_free(chip->card);
-		atomic_dec(&chip->active);
 	}
 	mutex_unlock(&register_mutex);
 	return err;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index db8404e31fae..64b90b8ec661 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1882,7 +1882,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 				char *name)
 {
 	struct uac_processing_unit_descriptor *desc = raw_desc;
-	int num_ins = desc->bNrInPins;
+	int num_ins;
 	struct usb_mixer_elem_info *cval;
 	struct snd_kcontrol *kctl;
 	int i, err, nameid, type, len;
@@ -1897,7 +1897,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 		0, NULL, default_value_info
 	};
 
-	if (desc->bLength < 13 || desc->bLength < 13 + num_ins ||
+	if (desc->bLength < 13) {
+		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
+		return -EINVAL;
+	}
+
+	num_ins = desc->bNrInPins;
+	if (desc->bLength < 13 + num_ins ||
 	    desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) {
 		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
 		return -EINVAL;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index e6ac7b9b4648..497bad9f2789 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -313,6 +313,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
 	return 0;
 }
 
+/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk
+ * applies. Returns 1 if a quirk was found.
+ */
 static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 					 struct usb_device *dev,
 					 struct usb_interface_descriptor *altsd,
@@ -391,7 +394,7 @@ add_sync_ep:
 
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
-	return 0;
+	return 1;
 }
 
 static int set_sync_endpoint(struct snd_usb_substream *subs,
@@ -430,6 +433,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 	if (err < 0)
 		return err;
 
+	/* endpoint set by quirk */
+	if (err > 0)
+		return 0;
+
 	if (altsd->bNumEndpoints < 2)
 		return 0;
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 15cbe2565703..d32727c74a16 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3321,6 +3321,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					}
 				}
 			},
+			{
+				.ifnum = -1
+			},
 		}
 	}
 },