diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 42980da45fbf..06a7ced218e2 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -403,8 +403,6 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal),
 	/* ESI, Quatafire610 */
 	SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal),
-	// AcousticReality, eARMasterOne. Terratec OEM.
-	SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000002, &spec_normal),
 	/* CME, MatrixKFW */
 	SND_BEBOB_DEV_ENTRY(VEN_CME, 0x00030000, &spec_normal),
 	// Phonic Helix Board 12 FireWire MkII.
@@ -434,7 +432,8 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000007, &yamaha_terratec_spec),
 	/* TerraTec Electronic GmbH, EWS MIC2/MIC8 */
 	SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000005, &spec_normal),
-	/* Terratec Electronic GmbH, Aureon 7.1 Firewire */
+	// Terratec Electronic GmbH, Aureon 7.1 Firewire.
+	// AcousticReality, eAR Master One, Eroica, Figaro, and Ciaccona. Perhaps Terratec OEM.
 	SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000002, &spec_normal),
 	/* Yamaha, GO44 */
 	SND_BEBOB_DEV_ENTRY(VEN_YAMAHA, 0x0010000b, &yamaha_terratec_spec),
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index 6c59a3eec59b..ef6d0a20efd8 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -551,7 +551,7 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev)
 	}
 	if (acard->sb->hardware != SB_HW_16) {
 		snd_printk(KERN_ERR PFX "SB16 not found during probe\n");
-		return err;
+		return -ENODEV;
 	}
 
 	snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 493d8ee2642a..1389cfd5e0db 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8442,6 +8442,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 2d1bfbcba933..df3ba5c70de9 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -694,7 +694,7 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich
 	int status, civ, i, step;
 	int ack = 0;
 
-	if (!ichdev->prepared || ichdev->suspended)
+	if (!(ichdev->prepared || chip->in_measurement) || ichdev->suspended)
 		return;
 
 	spin_lock_irqsave(&chip->reg_lock, flags);
diff --git a/sound/synth/emux/emux_nrpn.c b/sound/synth/emux/emux_nrpn.c
index 7eed5791972c..8056422ed7c5 100644
--- a/sound/synth/emux/emux_nrpn.c
+++ b/sound/synth/emux/emux_nrpn.c
@@ -69,7 +69,7 @@ static const int gs_sense[] =
 	DEF_FX_VIBRATE, DEF_FX_VIBDEPTH, DEF_FX_VIBDELAY
 };
 
-/* effect sensitivies for XG controls:
+/* effect sensitivities for XG controls:
  * adjusted for chaos 8MB soundfonts
  */
 static const int xg_sense[] =
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index da3cf8912463..16f00097cb95 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -108,7 +108,7 @@ snd_soundfont_close_check(struct snd_sf_list *sflist, int client)
  * Deal with a soundfont patch.  Any driver could use these routines
  * although it was designed for the AWE64.
  *
- * The sample_write and callargs pararameters allow a callback into
+ * The sample_write and callargs parameters allow a callback into
  * the actual driver to write sample data to the board or whatever
  * it wants to do with it.
  */
@@ -799,7 +799,7 @@ snd_sf_linear_to_log(unsigned int amount, int offset, int ratio)
 		amount <<= 1;
 	s = (amount >> 24) & 0x7f;
 	low = (amount >> 16) & 0xff;
-	/* linear approxmimation by lower 8 bit */
+	/* linear approximation by lower 8 bit */
 	v = (log_tbl[s + 1] * low + log_tbl[s] * (0x100 - low)) >> 8;
 	v -= offset;
 	v = (v * ratio) >> 16;
@@ -1433,7 +1433,7 @@ snd_sf_free(struct snd_sf_list *sflist)
 
 /*
  * Remove all samples
- * The soundcard should be silet before calling this function.
+ * The soundcard should be silent before calling this function.
  */
 int
 snd_soundfont_remove_samples(struct snd_sf_list *sflist)
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 5577a776561b..6c0a052a28f9 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -19,7 +19,7 @@ struct audioformat {
 	unsigned char iface;		/* interface number */
 	unsigned char altsetting;	/* corresponding alternate setting */
 	unsigned char ep_idx;		/* endpoint array index */
-	unsigned char altset_idx;	/* array index of altenate setting */
+	unsigned char altset_idx;	/* array index of alternate setting */
 	unsigned char attributes;	/* corresponding attributes of cs endpoint */
 	unsigned char endpoint;		/* endpoint */
 	unsigned char ep_attr;		/* endpoint attributes */
@@ -158,6 +158,7 @@ struct snd_usb_substream {
 	unsigned int stream_offset_adj;	/* Bytes to drop from beginning of stream (for non-compliant devices) */
 
 	unsigned int running: 1;	/* running status */
+	unsigned int period_elapsed_pending;	/* delay period handling */
 
 	unsigned int buffer_bytes;	/* buffer size in bytes */
 	unsigned int inflight_bytes;	/* in-flight data bytes on buffer (for playback) */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index da649211bff3..4f856771216b 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -645,7 +645,7 @@ static bool endpoint_compatible(struct snd_usb_endpoint *ep,
 }
 
 /*
- * Check whether the given fp and hw params are compatbile with the current
+ * Check whether the given fp and hw params are compatible with the current
  * setup of the target EP for implicit feedback sync
  */
 bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip,
@@ -1245,7 +1245,7 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip,
  *
  * This function sets up the EP to be fully usable state.
  * It's called either from hw_params or prepare callback.
- * The function checks need_setup flag, and perfoms nothing unless needed,
+ * The function checks need_setup flag, and performs nothing unless needed,
  * so it's safe to call this multiple times.
  *
  * This returns zero if unchanged, 1 if the configuration has changed,
diff --git a/sound/usb/media.c b/sound/usb/media.c
index 812017eacbcf..840f42cb9272 100644
--- a/sound/usb/media.c
+++ b/sound/usb/media.c
@@ -285,7 +285,7 @@ snd_mixer_init:
 			ret);
 
 	if (!media_devnode_is_registered(mdev->devnode)) {
-		/* dont'register if snd_media_mixer_init() failed */
+		/* don't register if snd_media_mixer_init() failed */
 		if (ret)
 			goto create_fail;
 
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
index c53a9773f310..0255089c9efb 100644
--- a/sound/usb/mixer_s1810c.c
+++ b/sound/usb/mixer_s1810c.c
@@ -163,7 +163,7 @@ snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a,
 }
 
 /*
- * When opening Universal Control the program periodicaly
+ * When opening Universal Control the program periodically
  * sends and receives state packets for syncinc state between
  * the device and the host.
  *
diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c
index 691b95466d0f..0d6e4f15bf77 100644
--- a/sound/usb/mixer_scarlett.c
+++ b/sound/usb/mixer_scarlett.c
@@ -21,7 +21,7 @@
  * Auto-detection via UAC2 is not feasible to properly discover the vast
  * majority of features. It's related to both Linux/ALSA's UAC2 as well as
  * Focusrite's implementation of it. Eventually quirks may be sufficient but
- * right now it's a major headache to work arount these things.
+ * right now it's a major headache to work around these things.
  *
  * NB. Neither the OSX nor the win driver provided by Focusrite performs
  * discovery, they seem to operate the same as this driver.
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index e26d37365f02..4e5031a68064 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -611,13 +611,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 	subs->hwptr_done = 0;
 	subs->transfer_done = 0;
 	subs->last_frame_number = 0;
+	subs->period_elapsed_pending = 0;
 	runtime->delay = 0;
 
-	/* for playback, submit the URBs now; otherwise, the first hwptr_done
-	 * updates for all URBs would happen at the same time when starting */
-	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
-		ret = start_endpoints(subs);
-
  unlock:
 	snd_usb_unlock_shutdown(chip);
 	return ret;
@@ -1398,6 +1394,10 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 		subs->trigger_tstamp_pending_update = false;
 	}
 
+	if (period_elapsed && !subs->running) {
+		subs->period_elapsed_pending = 1;
+		period_elapsed = 0;
+	}
 	spin_unlock_irqrestore(&subs->lock, flags);
 	urb->transfer_buffer_length = bytes;
 	if (period_elapsed)
@@ -1413,6 +1413,7 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 {
 	unsigned long flags;
 	struct snd_urb_ctx *ctx = urb->context;
+	bool period_elapsed = false;
 
 	spin_lock_irqsave(&subs->lock, flags);
 	if (ctx->queued) {
@@ -1423,13 +1424,20 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 	}
 
 	subs->last_frame_number = usb_get_current_frame_number(subs->dev);
+	if (subs->running) {
+		period_elapsed = subs->period_elapsed_pending;
+		subs->period_elapsed_pending = 0;
+	}
 	spin_unlock_irqrestore(&subs->lock, flags);
+	if (period_elapsed)
+		snd_pcm_period_elapsed(subs->pcm_substream);
 }
 
 static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream,
 					      int cmd)
 {
 	struct snd_usb_substream *subs = substream->runtime->private_data;
+	int err;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -1440,6 +1448,14 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
 					      prepare_playback_urb,
 					      retire_playback_urb,
 					      subs);
+		if (cmd == SNDRV_PCM_TRIGGER_START) {
+			err = start_endpoints(subs);
+			if (err < 0) {
+				snd_usb_endpoint_set_callback(subs->data_endpoint,
+							      NULL, NULL, NULL);
+				return err;
+			}
+		}
 		subs->running = 1;
 		dev_dbg(&subs->dev->dev, "%d:%d Start Playback PCM\n",
 			subs->cur_audiofmt->iface,
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 6154662d3097..c39cc6851e2d 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -526,7 +526,7 @@ static int usx2y_urbs_start(struct snd_usx2y_substream *subs)
  cleanup:
 	if (err) {
 		usx2y_subs_startup_finish(usx2y);
-		usx2y_clients_stop(usx2y);		// something is completely wroong > stop evrything
+		usx2y_clients_stop(usx2y);	// something is completely wrong > stop everything
 	}
 	return err;
 }
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
index da643c2dbb12..db83522c1b49 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.c
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -480,7 +480,7 @@ static int usx2y_usbpcm_urbs_start(struct snd_usx2y_substream *subs)
  cleanup:
 	if (err) {
 		usx2y_subs_startup_finish(usx2y);	// Call it now
-		usx2y_clients_stop(usx2y);		// something is completely wroong > stop evrything
+		usx2y_clients_stop(usx2y);	// something is completely wrong > stop everything
 	}
 	return err;
 }
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index 33b12aa67cf5..a34d7d9c2a57 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -236,7 +236,7 @@ static void had_write_register(struct snd_intelhad *ctx, u32 reg, u32 val)
  * updating AUD_CONFIG register.
  * This is because:
  * Bit6 of AUD_CONFIG register is writeonly due to a silicon bug on VLV2
- * HDMI IP. As a result a read-modify of AUD_CONFIG regiter will always
+ * HDMI IP. As a result a read-modify of AUD_CONFIG register will always
  * clear bit6. AUD_CONFIG[6:4] represents the "channels" field of the
  * register. This field should be 1xy binary for configuration with 6 or
  * more channels. Read-modify of AUD_CONFIG (Eg. for enabling audio)
@@ -342,7 +342,7 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream,
 
 /*
  * function to initialize audio
- * registers and buffer confgiuration registers
+ * registers and buffer configuration registers
  * This function is called in the prepare callback
  */
 static int had_init_audio_ctrl(struct snd_pcm_substream *substream,
@@ -1790,7 +1790,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev)
 		pcm->private_data = ctx;
 		pcm->info_flags = 0;
 		strscpy(pcm->name, card->shortname, strlen(card->shortname));
-		/* setup the ops for playabck */
+		/* setup the ops for playback */
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops);
 
 		/* allocate dma pages;
diff --git a/sound/x86/intel_hdmi_audio.h b/sound/x86/intel_hdmi_audio.h
index 0d91bb5dbab7..bb3853195922 100644
--- a/sound/x86/intel_hdmi_audio.h
+++ b/sound/x86/intel_hdmi_audio.h
@@ -96,7 +96,7 @@ struct pcm_stream_info {
  * @had_spinlock: driver lock
  * @aes_bits: IEC958 status bits
  * @buff_done: id of current buffer done intr
- * @dev: platoform device handle
+ * @dev: platform device handle
  * @chmap: holds channel map info
  */
 struct snd_intelhad {