Merge commit '8700c95adb03' into timers/nohz

The full dynticks tree needs the latest RCU and sched
upstream updates in order to fix some dependencies.

Merge a common upstream merge point that has these
updates.

Conflicts:
	include/linux/perf_event.h
	kernel/rcutree.h
	kernel/rcutree_plugin.h

Signed-off-by: Frederic Weisbecker <fweisbec@gmail.com>

46 files changed, 494 insertions, 207 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 71ae86c..eb560fa 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3222,18 +3222,10 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap);
 int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream,
 			   struct vm_area_struct *area)
 {
-	long size;
-	unsigned long offset;
+	struct snd_pcm_runtime *runtime = substream->runtime;;
 
 	area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
-	area->vm_flags |= VM_IO;
-	size = area->vm_end - area->vm_start;
-	offset = area->vm_pgoff << PAGE_SHIFT;
-	if (io_remap_pfn_range(area, area->vm_start,
-				(substream->runtime->dma_addr + offset) >> PAGE_SHIFT,
-				size, area->vm_page_prot))
-		return -EAGAIN;
-	return 0;
+	return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes);
 }
 
 EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem);
diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c
index 066f5f3..c390886 100644
--- a/sound/core/seq/oss/seq_oss_event.c
+++ b/sound/core/seq/oss/seq_oss_event.c
@@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev
 static int
 note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
 {
-	struct seq_oss_synthinfo *info = &dp->synths[dev];
+	struct seq_oss_synthinfo *info;
+
+	if (!snd_seq_oss_synth_is_valid(dp, dev))
+		return -ENXIO;
+
+	info = &dp->synths[dev];
 	switch (info->arg.event_passing) {
 	case SNDRV_SEQ_OSS_PROCESS_EVENTS:
 		if (! info->ch || ch < 0 || ch >= info->nr_voices) {
@@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st
 static int
 note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
 {
-	struct seq_oss_synthinfo *info = &dp->synths[dev];
+	struct seq_oss_synthinfo *info;
+
+	if (!snd_seq_oss_synth_is_valid(dp, dev))
+		return -ENXIO;
+
+	info = &dp->synths[dev];
 	switch (info->arg.event_passing) {
 	case SNDRV_SEQ_OSS_PROCESS_EVENTS:
 		if (! info->ch || ch < 0 || ch >= info->nr_voices) {
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 160b1bd..24d44b2 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -290,10 +290,10 @@ int snd_seq_timer_open(struct snd_seq_queue *q)
 			tid.device = SNDRV_TIMER_GLOBAL_SYSTEM;
 			err = snd_timer_open(&t, str, &tid, q->queue);
 		}
-		if (err < 0) {
-			snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
-			return err;
-		}
+	}
+	if (err < 0) {
+		snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
+		return err;
 	}
 	t->callback = snd_seq_timer_interrupt;
 	t->callback_data = q;
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 8575861..0097f36 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol,
 	}
 	if (!changed)
 		return 0;
-	return slave_put_val(slave, ucontrol);
+	err = slave_put_val(slave, ucontrol);
+	if (err < 0)
+		return err;
+	return 1;
 }
 
 static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 30bcfe4..4ff60a6 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
 		case MIDI_PGM_CHANGE:
 			if (seq_mode == SEQ_2)
 			{
+				if (chn > 15)
+					break;
+
 				synth_devs[dev]->chn_info[chn].pgm_num = p1;
 				if ((int) dev >= num_synths)
 					synth_devs[dev]->set_instr(dev, chn, p1);
@@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
 		case MIDI_PITCH_BEND:
 			if (seq_mode == SEQ_2)
 			{
+				if (chn > 15)
+					break;
+
 				synth_devs[dev]->chn_info[chn].bender_value = w14;
 
 				if ((int) dev < num_synths)
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 3536b07..0aabfed 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi,
 
 static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 {
-	struct snd_card *card = asihpi->card;
+	struct snd_card *card;
 	unsigned int idx = 0;
 	unsigned int subindex = 0;
 	int err;
@@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 
 	if (snd_BUG_ON(!asihpi))
 		return -EINVAL;
+	card = asihpi->card;
 	strcpy(card->mixername, "Asihpi Mixer");
 
 	err =
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 04b5738..4aba764 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg)
 		"Line Out", "Speaker", "HP Out", "CD",
 		"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
 		"Line In", "Aux", "Mic", "Telephony",
-		"SPDIF In", "Digitial In", "Reserved", "Other"
+		"SPDIF In", "Digital In", "Reserved", "Other"
 	};
 
 	return jack_types[(cfg & AC_DEFCFG_DEVICE)
@@ -494,7 +494,7 @@ static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid)
 
 int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid)
 {
-	return get_num_conns(codec, nid) & AC_CLIST_LENGTH;
+	return snd_hda_get_raw_connections(codec, nid, NULL, 0);
 }
 
 /**
@@ -517,9 +517,6 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 	hda_nid_t prev_nid;
 	int null_count = 0;
 
-	if (snd_BUG_ON(!conn_list || max_conns <= 0))
-		return -EINVAL;
-
 	parm = get_num_conns(codec, nid);
 	if (!parm)
 		return 0;
@@ -545,7 +542,8 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 					  AC_VERB_GET_CONNECT_LIST, 0);
 		if (parm == -1 && codec->bus->rirb_error)
 			return -EIO;
-		conn_list[0] = parm & mask;
+		if (conn_list)
+			conn_list[0] = parm & mask;
 		return 1;
 	}
 
@@ -580,14 +578,20 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 				continue;
 			}
 			for (n = prev_nid + 1; n <= val; n++) {
+				if (conn_list) {
+					if (conns >= max_conns)
+						return -ENOSPC;
+					conn_list[conns] = n;
+				}
+				conns++;
+			}
+		} else {
+			if (conn_list) {
 				if (conns >= max_conns)
 					return -ENOSPC;
-				conn_list[conns++] = n;
+				conn_list[conns] = val;
 			}
-		} else {
-			if (conns >= max_conns)
-				return -ENOSPC;
-			conn_list[conns++] = val;
+			conns++;
 		}
 		prev_nid = val;
 	}
@@ -3140,7 +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)
@@ -3334,6 +3338,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec,
 		return -EBUSY;
 	}
 	spdif = snd_array_new(&codec->spdif_out);
+	if (!spdif)
+		return -ENOMEM;
 	for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
 		if (!kctl)
@@ -3431,11 +3437,16 @@ static struct snd_kcontrol_new spdif_share_sw = {
 int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
 				  struct hda_multi_out *mout)
 {
+	struct snd_kcontrol *kctl;
+
 	if (!mout->dig_out_nid)
 		return 0;
+
+	kctl = snd_ctl_new1(&spdif_share_sw, mout);
+	if (!kctl)
+		return -ENOMEM;
 	/* ATTENTION: here mout is passed as private_data, instead of codec */
-	return snd_hda_ctl_add(codec, mout->dig_out_nid,
-			      snd_ctl_new1(&spdif_share_sw, mout));
+	return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl);
 }
 EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw);
 
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 7dd8463..d0d7ac1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
 		     unsigned char *buf, int *eld_size)
 {
 	int i;
-	int ret;
+	int ret = 0;
 	int size;
 
 	/*
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 78897d0..2dbe767 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path);
 static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
 {
 	struct hda_gen_spec *spec = codec->spec;
-	bool changed;
+	bool changed = false;
 	int i;
 
 	if (!spec->power_down_unused || path->active)
@@ -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 4cea6bb6..bcd40ee 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
  * this may give more power-saving, but will take longer time to
  * wake up.
  */
-static int power_save_controller = -1;
-module_param(power_save_controller, bint, 0644);
+static bool power_save_controller = 1;
+module_param(power_save_controller, bool, 0644);
 MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
 #endif /* CONFIG_PM */
 
@@ -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 */
 
@@ -2846,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
-	if (power_save_controller > 0)
-		return 0;
 	if (!power_save_controller ||
 	    !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
 		return -EBUSY;
@@ -3481,6 +3564,7 @@ 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_ca0132.c b/sound/pci/hda/patch_ca0132.c
index db02c1e..0792b57 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec,
 	hda_frame_size_words = ((sample_rate_div == 0) ? 0 :
 			(num_chans * sample_rate_mul / sample_rate_div));
 
+	if (hda_frame_size_words == 0) {
+		snd_printdd(KERN_ERR "frmsz zero\n");
+		return -EINVAL;
+	}
+
 	buffer_size_words = min(buffer_size_words,
 				(unsigned int)(UC_RANGE(chip_addx, 1) ?
 				65536 : 32768));
@@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec,
 		   chip_addx, hda_frame_size_words, num_chans,
 		   sample_rate_mul, sample_rate_div, buffer_size_words);
 
-	if ((buffer_addx == NULL) || (hda_frame_size_words == 0) ||
-	    (buffer_size_words < hda_frame_size_words)) {
+	if (buffer_size_words < hda_frame_size_words) {
 		snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n");
 		return -EINVAL;
 	}
@@ -3235,7 +3239,7 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val)
 	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
 
-	if (!dspload_is_loaded(codec))
+	if (spec->dsp_state != DSP_DOWNLOADED)
 		return 0;
 
 	/* if CrystalVoice if off, vipsource should be 0 */
@@ -4263,11 +4267,12 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
  */
 static void ca0132_setup_defaults(struct hda_codec *codec)
 {
+	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
 	int num_fx;
 	int idx, i;
 
-	if (!dspload_is_loaded(codec))
+	if (spec->dsp_state != DSP_DOWNLOADED)
 		return;
 
 	/* out, in effects + voicefx */
@@ -4347,12 +4352,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
 		return false;
 
 	dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
-	dspload_image(codec, dsp_os_image, 0, 0, true, 0);
+	if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
+		pr_err("ca0132 dspload_image failed.\n");
+		goto exit_download;
+	}
+
 	dsp_loaded = dspload_wait_loaded(codec);
 
+exit_download:
 	release_firmware(fw_entry);
 
-
 	return dsp_loaded;
 }
 
@@ -4363,16 +4372,13 @@ static void ca0132_download_dsp(struct hda_codec *codec)
 #ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP
 	return; /* NOP */
 #endif
-	spec->dsp_state = DSP_DOWNLOAD_INIT;
 
-	if (spec->dsp_state == DSP_DOWNLOAD_INIT) {
-		chipio_enable_clocks(codec);
-		spec->dsp_state = DSP_DOWNLOADING;
-		if (!ca0132_download_dsp_images(codec))
-			spec->dsp_state = DSP_DOWNLOAD_FAILED;
-		else
-			spec->dsp_state = DSP_DOWNLOADED;
-	}
+	chipio_enable_clocks(codec);
+	spec->dsp_state = DSP_DOWNLOADING;
+	if (!ca0132_download_dsp_images(codec))
+		spec->dsp_state = DSP_DOWNLOAD_FAILED;
+	else
+		spec->dsp_state = DSP_DOWNLOADED;
 
 	if (spec->dsp_state == DSP_DOWNLOADED)
 		ca0132_set_dsp_msr(codec, true);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 72ebb8a..0d9c58f 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);
 	}
 }
 
@@ -506,6 +506,8 @@ static int patch_cs420x(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 
+	spec->gen.automute_hook = cs_automute;
+
 	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
 			   cs420x_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -893,6 +895,8 @@ static int patch_cs4210(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 
+	spec->gen.automute_hook = cs_automute;
+
 	snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl,
 			   cs421x_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 941bf6c..2a89d1ee 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/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 78e1827..de8ac5c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 
 	_snd_printd(SND_PR_VERBOSE,
 		"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
+		codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid);
 
 	if (eld->eld_valid) {
 		if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2d4237b..f15c36b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3163,6 +3163,7 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
+	case 0x10ec0233:
 	case 0x10ec0282:
 	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
@@ -3439,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	const hda_nid_t *ssids;
 
 	if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
-	    codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
+	    codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 ||
+	    codec->vendor_id == 0x10ec0671)
 		ssids = alc663_ssids;
 	else
 		ssids = alc662_ssids;
@@ -3862,6 +3864,7 @@ static int patch_alc680(struct hda_codec *codec)
  */
 static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
+	{ .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
 	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
 	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
@@ -3892,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
 	{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
 	{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
+	{ .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
 	{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
 	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 83d5335..dafe04a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
 	return 0;
 }
 
+/* check whether a built-in speaker is included in parsed pins */
+static bool has_builtin_speaker(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t *nid_pin;
+	int nids, i;
+
+	if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) {
+		nid_pin = spec->gen.autocfg.line_out_pins;
+		nids = spec->gen.autocfg.line_outs;
+	} else {
+		nid_pin = spec->gen.autocfg.speaker_pins;
+		nids = spec->gen.autocfg.speaker_outs;
+	}
+
+	for (i = 0; i < nids; i++) {
+		unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]);
+		if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT)
+			return true;
+	}
+	return false;
+}
+
 /*
  * PC beep controls
  */
@@ -3890,6 +3913,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 		return err;
 	}
 
+	/* Don't GPIO-mute speakers if there are no internal speakers, because
+	 * the GPIO might be necessary for Headphone
+	 */
+	if (spec->eapd_switch && !has_builtin_speaker(codec))
+		spec->eapd_switch = 0;
+
 	codec->proc_widget_hook = stac92hd7x_proc_hook;
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 2ffdc35..806407a 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2594,6 +2594,8 @@ static int snd_ice1712_create(struct snd_card *card,
 	snd_ice1712_proc_init(ice);
 	synchronize_irq(pci->irq);
 
+	card->private_data = ice;
+
 	err = pci_request_regions(pci, "ICE1712");
 	if (err < 0) {
 		kfree(ice);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index f9b5229..8f489de 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -295,18 +295,5 @@ static struct pcmcia_driver pdacf_cs_driver = {
 	.suspend	= pdacf_suspend,
 	.resume		= pdacf_resume,
 #endif
-
 };
-
-static int __init init_pdacf(void)
-{
-	return pcmcia_register_driver(&pdacf_cs_driver);
-}
-
-static void __exit exit_pdacf(void)
-{
-	pcmcia_unregister_driver(&pdacf_cs_driver);
-}
-
-module_init(init_pdacf);
-module_exit(exit_pdacf);
+module_pcmcia_driver(pdacf_cs_driver);
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 8f93504..d4db7ec 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -367,16 +367,4 @@ static struct pcmcia_driver vxp_cs_driver = {
 	.resume		= vxp_resume,
 #endif
 };
-
-static int __init init_vxpocket(void)
-{
-	return pcmcia_register_driver(&vxp_cs_driver);
-}
-
-static void __exit exit_vxpocket(void)
-{
-	pcmcia_unregister_driver(&vxp_cs_driver);
-}
-
-module_init(init_vxpocket);
-module_exit(exit_vxpocket);
+module_pcmcia_driver(vxp_cs_driver);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index ac948a6..e7d3471 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -364,6 +364,39 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 }
 EXPORT_SYMBOL_GPL(arizona_out_ev);
 
+int arizona_hp_ev(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol,
+		   int event)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
+	unsigned int mask = 1 << w->shift;
+	unsigned int val;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		val = mask;
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		val = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Store the desired state for the HP outputs */
+	priv->arizona->hp_ena &= ~mask;
+	priv->arizona->hp_ena |= val;
+
+	/* Force off if HPDET magic is active */
+	if (priv->arizona->hpdet_magic)
+		val = 0;
+
+	snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val);
+
+	return arizona_out_ev(w, kcontrol, event);
+}
+EXPORT_SYMBOL_GPL(arizona_hp_ev);
+
 static unsigned int arizona_sysclk_48k_rates[] = {
 	6144000,
 	12288000,
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 116372c..13dd291 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -184,6 +184,9 @@ extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
 extern int arizona_out_ev(struct snd_soc_dapm_widget *w,
 			  struct snd_kcontrol *kcontrol,
 			  int event);
+extern int arizona_hp_ev(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol,
+			 int event);
 
 extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			      int source, unsigned int freq, int dir);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index fc17604..fc17604 100755..100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 7e103f2..7e103f2 100755..100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index f2d61a1..566ea32 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		width = SI476X_PCM_FORMAT_S8;
+		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		width = SI476X_PCM_FORMAT_S16_LE;
 		break;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b8d461d..15bc31f 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -573,11 +573,18 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = {
 	{ 0x025e, 0x0112 },
 };
 
+static const struct reg_default wm5102_sysclk_revb_patch[] = {
+	{ 0x3081, 0x08FE },
+	{ 0x3083, 0x00ED },
+	{ 0x30C1, 0x08FE },
+	{ 0x30C3, 0x00ED },
+};
+
 static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
 			    struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct arizona *arizona = dev_get_drvdata(codec->dev);
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
 	struct regmap *regmap = codec->control_data;
 	const struct reg_default *patch = NULL;
 	int i, patch_size;
@@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
 		patch = wm5102_sysclk_reva_patch;
 		patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch);
 		break;
+	default:
+		patch = wm5102_sysclk_revb_patch;
+		patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch);
+		break;
 	}
 
 	switch (event) {
@@ -755,7 +766,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
 
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
 SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
 	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
@@ -767,7 +778,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
 SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
@@ -1120,11 +1131,11 @@ ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
 		       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
 
-SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
-SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
 		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index cd17b47..7841b42 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -213,9 +213,9 @@ ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
 
 SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
 	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
 	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("OUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
 	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
 SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
 	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
@@ -226,9 +226,9 @@ SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
 
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+SOC_DOUBLE_R("HPOUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
@@ -240,10 +240,10 @@ SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L,
 SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
@@ -260,11 +260,11 @@ SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
 		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
 		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
 		       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
 		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
 		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
 		       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("OUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
 		       ARIZONA_OUTPUT_PATH_CONFIG_3R,
 		       ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
 
@@ -551,11 +551,11 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
 SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
 		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
 
-SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
-SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
 		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ec0efc1..0e8b3aa 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1301,7 +1301,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&priv->hpl.work, 200);
+	schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
@@ -1318,7 +1318,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&priv->hpr.work, 200);
+	schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 134e41c..f8a31ad 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = {
 	{ "ROP", NULL, "Right Speaker PGA" },
 	{ "RON", NULL, "Right Speaker PGA" },
 
+	{ "Charge Pump", NULL, "CLK_DSP" },
+
 	{ "Left Headphone Output PGA", NULL, "Charge Pump" },
 	{ "Right Headphone Output PGA", NULL, "Charge Pump" },
 	{ "Left Line Output PGA", NULL, "Charge Pump" },
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 9bb9273..a64b934 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -53,8 +53,8 @@
  * using 2 wire for device control, so we cache them instead.
  */
 static const struct reg_default wm8960_reg_defaults[] = {
-	{  0x0, 0x0097 },
-	{  0x1, 0x0097 },
+	{  0x0, 0x00a7 },
+	{  0x1, 0x00a7 },
 	{  0x2, 0x0000 },
 	{  0x3, 0x0000 },
 	{  0x4, 0x0000 },
@@ -323,8 +323,8 @@ SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
 SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
 		   wm8960_rin, ARRAY_SIZE(wm8960_rin)),
 
-SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
-SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER1, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0),
 
 SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
 SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f3f7e75..9af1bdd 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 						&buf_list);
 			if (!buf) {
 				adsp_err(dsp, "Out of memory\n");
-				return -ENOMEM;
+				ret = -ENOMEM;
+				goto out_fw;
 			}
 
 			adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
@@ -865,7 +866,7 @@ out_fw:
 	wm_adsp_buf_free(&buf_list);
 out:
 	kfree(file);
-	return 0;
+	return ret;
 }
 
 int wm_adsp1_init(struct wm_adsp *adsp)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 55464a5..810c7ee 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97)
 
 	if (imx_ssi->ac97_reset)
 		imx_ssi->ac97_reset(ac97);
+	/* First read sometimes fails, do a dummy read */
+	imx_ssi_ac97_read(ac97, 0);
 }
 
 static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
@@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
 
 	if (imx_ssi->ac97_warm_reset)
 		imx_ssi->ac97_warm_reset(ac97);
+
+	/* First read sometimes fails, do a dummy read */
+	imx_ssi_ac97_read(ac97, 0);
 }
 
 struct snd_ac97_bus_ops soc_ac97_ops = {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 8e52c14..eb43738 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = {
 	.num_links = ARRAY_SIZE(pcm030_fabric_dai),
 };
 
-static int __init pcm030_fabric_probe(struct platform_device *op)
+static int pcm030_fabric_probe(struct platform_device *op)
 {
 	struct device_node *np = op->dev.of_node;
 	struct device_node *platform_np;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index d7231e3..6bbeb0b 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = {
 static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
 {
 	struct i2s_dai *i2s;
+	int ret;
 
 	i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL);
 	if (i2s == NULL)
@@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
 		i2s->i2s_dai_drv.capture.channels_max = 2;
 		i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES;
 		i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS;
+		dev_set_drvdata(&i2s->pdev->dev, i2s);
 	} else {	/* Create a new platform_device for Secondary */
-		i2s->pdev = platform_device_register_resndata(NULL,
-				"samsung-i2s-sec", -1, NULL, 0, NULL, 0);
+		i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1);
 		if (IS_ERR(i2s->pdev))
 			return NULL;
-	}
 
-	/* Pre-assign snd_soc_dai_set_drvdata */
-	dev_set_drvdata(&i2s->pdev->dev, i2s);
+		platform_set_drvdata(i2s->pdev, i2s);
+		ret = platform_device_add(i2s->pdev);
+		if (ret < 0)
+			return NULL;
+	}
 
 	return i2s;
 }
@@ -1107,6 +1110,10 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 
 	if (samsung_dai_type == TYPE_SEC) {
 		sec_dai = dev_get_drvdata(&pdev->dev);
+		if (!sec_dai) {
+			dev_err(&pdev->dev, "Unable to get drvdata\n");
+			return -EFAULT;
+		}
 		snd_soc_register_dai(&sec_dai->pdev->dev,
 			&sec_dai->i2s_dai_drv);
 		asoc_dma_platform_register(&pdev->dev);
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 19eff8f..1a8b03e 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static struct snd_soc_platform sh7760_soc_platform = {
-	.pcm_ops 	= &camelot_pcm_ops,
+static struct snd_soc_platform_driver sh7760_soc_platform = {
+	.ops		= &camelot_pcm_ops,
 	.pcm_new	= camelot_pcm_new,
 	.pcm_free	= camelot_pcm_free,
 };
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index b5b3db7..ed0bfb0 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
 	if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) {
 		ret = platform->driver->compr_ops->set_params(cstream, params);
 		if (ret < 0)
-			goto out;
+			goto err;
 	}
 
 	if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) {
 		ret = rtd->dai_link->compr_ops->set_params(cstream);
 		if (ret < 0)
-			goto out;
+			goto err;
 	}
 
 	snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
 				SND_SOC_DAPM_STREAM_START);
 
-out:
+	/* cancel any delayed stream shutdown that is pending */
+	rtd->pop_wait = 0;
+	mutex_unlock(&rtd->pcm_mutex);
+
+	cancel_delayed_work_sync(&rtd->delayed_work);
+
+	return ret;
+
+err:
 	mutex_unlock(&rtd->pcm_mutex);
 	return ret;
 }
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b7e84a7..ff4b45a5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 	val = val << shift;
 
 	ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
-	if (ret != 0)
+	if (ret < 0)
 		return ret;
 
 	if (snd_soc_volsw_is_stereo(mc)) {
@@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 	if (params->mask) {
 		ret = regmap_read(codec->control_data, params->base, &val);
 		if (ret != 0)
-			return ret;
+			goto out;
 
 		val &= params->mask;
 
@@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 			((u32 *)data)[0] |= cpu_to_be32(val);
 			break;
 		default:
-			return -EINVAL;
+			ret = -EINVAL;
+			goto out;
 		}
 	}
 
 	ret = regmap_raw_write(codec->control_data, params->base,
 			       data, len);
 
+out:
 	kfree(data);
 
 	return ret;
@@ -4197,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			dev_err(card->dev,
 				"ASoC: Property '%s' index %d could not be read: %d\n",
 				propname, 2 * i, ret);
-			kfree(routes);
 			return -EINVAL;
 		}
 		ret = of_property_read_string_index(np, propname,
@@ -4206,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			dev_err(card->dev,
 				"ASoC: Property '%s' index %d could not be read: %d\n",
 				propname, (2 * i) + 1, ret);
-			kfree(routes);
 			return -EINVAL;
 		}
 	}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d6a9b3..d6d9ba2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 		if (path->weak)
 			continue;
 
+		if (path->walking)
+			return 1;
+
 		if (path->walked)
 			continue;
 
@@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 
 		if (path->sink && path->connect) {
 			path->walked = 1;
+			path->walking = 1;
 
 			/* do we need to add this widget to the list ? */
 			if (list) {
@@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 					dev_err(widget->dapm->dev,
 						"ASoC: could not add widget %s\n",
 						widget->name);
+					path->walking = 0;
 					return con;
 				}
 			}
 
 			con += is_connected_output_ep(path->sink, list);
+
+			path->walking = 0;
 		}
 	}
 
@@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 		if (path->weak)
 			continue;
 
+		if (path->walking)
+			return 1;
+
 		if (path->walked)
 			continue;
 
@@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 
 		if (path->source && path->connect) {
 			path->walked = 1;
+			path->walking = 1;
 
 			/* do we need to add this widget to the list ? */
 			if (list) {
@@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 					dev_err(widget->dapm->dev,
 						"ASoC: could not add widget %s\n",
 						widget->name);
+					path->walking = 0;
 					return con;
 				}
 			}
 
 			con += is_connected_input_ep(path->source, list);
+
+			path->walking = 0;
 		}
 	}
 
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 9b76cc5..5e7aebe 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm)
 
 static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int spear_pcm_new(struct snd_card *card,
-		struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
 	int ret;
 
 	if (!card->dev->dma_mask)
@@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
-		ret = spear_pcm_preallocate_dma_buffer(pcm,
+	if (rtd->cpu_dai->driver->playback.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
 				SNDRV_PCM_STREAM_PLAYBACK,
 				spear_pcm_hardware.buffer_bytes_max);
 		if (ret)
 			return ret;
 	}
 
-	if (dai->driver->capture.channels_min) {
-		ret = spear_pcm_preallocate_dma_buffer(pcm,
+	if (rtd->cpu_dai->driver->capture.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
 				SNDRV_PCM_STREAM_CAPTURE,
 				spear_pcm_hardware.buffer_bytes_max);
 		if (ret)
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index c27069d..7299587 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -121,7 +121,7 @@
 
 #define TEGRA20_I2S_TIMING_NON_SYM_ENABLE		(1 << 12)
 #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
-#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7ff
 #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK	(TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
 
 /* Fields in TEGRA20_I2S_FIFO_SCR */
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 34dc47b..a294d94 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -110,7 +110,7 @@
 
 #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE		(1 << 12)
 #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
-#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7ff
 #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK	(TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
 
 /* Fields in TEGRA30_I2S_OFFSET */
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index c925ab0..5e2c55c 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -43,8 +43,6 @@
 static const struct snd_pcm_hardware tegra_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
-				  SNDRV_PCM_INFO_PAUSE |
-				  SNDRV_PCM_INFO_RESUME |
 				  SNDRV_PCM_INFO_INTERLEAVED,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.channels_min		= 2,
@@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		return snd_dmaengine_pcm_trigger(substream,
-					SNDRV_PCM_TRIGGER_START);
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		return snd_dmaengine_pcm_trigger(substream,
-					SNDRV_PCM_TRIGGER_STOP);
-	default:
-		return -EINVAL;
-	}
-	return 0;
-}
-
 static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
 				struct vm_area_struct *vma)
 {
@@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = {
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= tegra_pcm_hw_params,
 	.hw_free	= tegra_pcm_hw_free,
-	.trigger	= tegra_pcm_trigger,
+	.trigger	= snd_dmaengine_pcm_trigger,
 	.pointer	= snd_dmaengine_pcm_pointer,
 	.mmap		= tegra_pcm_mmap,
 };
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 803953a..2da8ad7 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -244,6 +244,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 			usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
 
 		if (!assoc) {
+			/*
+			 * Firmware writers cannot count to three.  So to find
+			 * the IAD on the NuForce UDH-100, also check the next
+			 * interface.
+			 */
+			struct usb_interface *iface =
+				usb_ifnum_to_if(dev, ctrlif + 1);
+			if (iface &&
+			    iface->intf_assoc &&
+			    iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO &&
+			    iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2)
+				assoc = iface->intf_assoc;
+		}
+
+		if (!assoc) {
 			snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n");
 			return -EINVAL;
 		}
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 5e634a2..9e2703a 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 {
 	struct usb_device *dev = chip->dev;
 	unsigned char data[4];
-	int err, crate;
+	int err, cur_rate, prev_rate;
 	int clock = snd_usb_clock_find_source(chip, fmt->clock);
 
 	if (clock < 0)
@@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 		return -ENXIO;
 	}
 
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
+			      data, sizeof(data));
+	if (err < 0) {
+		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+			   dev->devnum, iface, fmt->altsetting);
+		prev_rate = 0;
+	} else {
+		prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
+	}
+
 	data[0] = rate;
 	data[1] = rate >> 8;
 	data[2] = rate >> 16;
@@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 		return err;
 	}
 
-	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
-				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8,
-				   snd_usb_ctrl_intf(chip) | (clock << 8),
-				   data, sizeof(data))) < 0) {
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
+			      data, sizeof(data));
+	if (err < 0) {
 		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);
-		return err;
+		cur_rate = 0;
+	} else {
+		cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
 	}
 
-	crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
-	if (crate != rate)
-		snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+	if (cur_rate != rate) {
+		snd_printd(KERN_WARNING
+			   "current rate %d is different from the runtime rate %d\n",
+			   cur_rate, rate);
+	}
+
+	/* Some devices doesn't respond to sample rate changes while the
+	 * interface is active. */
+	if (rate != prev_rate) {
+		usb_set_interface(dev, iface, 0);
+		usb_set_interface(dev, iface, fmt->altsetting);
+	}
 
 	return 0;
 }
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 638e7f7..ca4739c 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;
 		}
 	}
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 497d274..ebe9144 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol,
 	else
 		ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest,
 				  USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN,
-				  0, cpu_to_le16(wIndex),
+				  0, wIndex,
 				  &tmp, sizeof(tmp), 1000);
 	up_read(&mixer->chip->shutdown_rwsem);
 
@@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol,
 	else
 		ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest,
 				  USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT,
-				  cpu_to_le16(wValue), cpu_to_le16(wIndex),
+				  wValue, wIndex,
 				  NULL, 0, 1000);
 	up_read(&mixer->chip->shutdown_rwsem);
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 5325a38..9c5ab22 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -486,7 +486,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev)
 {
 	int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
 				  0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE,
-				  cpu_to_le16(1), 0, NULL, 0, 1000);
+				  1, 0, NULL, 0, 1000);
 
 	if (ret < 0)
 		return ret;