Merge tag 'v3.9-rc5' into patchwork

Linux 3.9-rc5

* tag 'v3.9-rc5': (1080 commits)
  Linux 3.9-rc5
  Revert "lockdep: check that no locks held at freeze time"
  dw_dmac: adjust slave_id accordingly to request line base
  dmaengine: dw_dma: fix endianess for DT xlate function
  PNP: List Rafael Wysocki as a maintainer
  rbd: don't zero-fill non-image object requests
  ia64 idle: delete stale (*idle)() function pointer
  Btrfs: don't drop path when printing out tree errors in scrub
  target: Fix RESERVATION_CONFLICT status regression for iscsi-target special case
  tcm_vhost: Avoid VIRTIO_RING_F_EVENT_IDX feature bit
  Revert "mm: introduce VM_POPULATE flag to better deal with racy userspace programs"
  usb: ftdi_sio: Add support for Mitsubishi FX-USB-AW/-BD
  mg_disk: fix error return code in mg_probe()
  Btrfs: fix wrong return value of btrfs_lookup_csum()
  Btrfs: fix wrong reservation of csums
  Btrfs: fix double free in the btrfs_qgroup_account_ref()
  Btrfs: limit the global reserve to 512mb
  Btrfs: hold the ordered operations mutex when waiting on ordered extents
  Btrfs: fix space accounting for unlink and rename
  Btrfs: fix space leak when we fail to reserve metadata space
  ...

22 files changed, 335 insertions, 92 deletions

diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c
index 066f5f3e3f4c..c3908862bc8b 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 160b1bd0cd62..24d44b2f61ac 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 857586135d18..0097f3619faa 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 30bcfe470f83..4ff60a6427d9 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 3536b076b529..0aabfedeecba 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 04b57383e8cb..ecdf30eb5879 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -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_generic.c b/sound/pci/hda/hda_generic.c
index 78897d05d80f..43c2ea539561 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -995,6 +995,8 @@ enum {
         BAD_NO_EXTRA_SURR_DAC = 0x101,
         /* Primary DAC shared with main surrounds */
         BAD_SHARED_SURROUND = 0x100,
+        /* No independent HP possible */
+        BAD_NO_INDEP_HP = 0x40,
         /* Primary DAC shared with main CLFE */
         BAD_SHARED_CLFE = 0x10,
         /* Primary DAC shared with extra surrounds */
@@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx)
         return snd_hda_get_path_idx(codec, path);
 }
 
+/* check whether the independent HP is available with the current config */
+static bool indep_hp_possible(struct hda_codec *codec)
+{
+        struct hda_gen_spec *spec = codec->spec;
+        struct auto_pin_cfg *cfg = &spec->autocfg;
+        struct nid_path *path;
+        int i, idx;
+
+        if (cfg->line_out_type == AUTO_PIN_HP_OUT)
+                idx = spec->out_paths[0];
+        else
+                idx = spec->hp_paths[0];
+        path = snd_hda_get_path_from_idx(codec, idx);
+        if (!path)
+                return false;
+
+        /* assume no path conflicts unless aamix is involved */
+        if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid))
+                return true;
+
+        /* check whether output paths contain aamix */
+        for (i = 0; i < cfg->line_outs; i++) {
+                if (spec->out_paths[i] == idx)
+                        break;
+                path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]);
+                if (path && is_nid_contained(path, spec->mixer_nid))
+                        return false;
+        }
+        for (i = 0; i < cfg->speaker_outs; i++) {
+                path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]);
+                if (path && is_nid_contained(path, spec->mixer_nid))
+                        return false;
+        }
+
+        return true;
+}
+
 /* fill the empty entries in the dac array for speaker/hp with the
  * shared dac pointed by the paths
  */
@@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
                 badness += BAD_MULTI_IO;
         }
 
+        if (spec->indep_hp && !indep_hp_possible(codec))
+                badness += BAD_NO_INDEP_HP;
+
         /* re-fill the shared DAC for speaker / headphone */
         if (cfg->line_out_type != AUTO_PIN_HP_OUT)
                 refill_shared_dacs(codec, cfg->hp_outs,
@@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec)
                                 cfg->speaker_pins, val);
         }
 
+        /* clear indep_hp flag if not available */
+        if (spec->indep_hp && !indep_hp_possible(codec))
+                spec->indep_hp = 0;
+
         kfree(best_cfg);
         return 0;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4cea6bb6fade..418bfc0eb0a3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -415,6 +415,8 @@ struct azx_dev {
         unsigned int opened :1;
         unsigned int running :1;
         unsigned int irq_pending :1;
+        unsigned int prepared:1;
+        unsigned int locked:1;
         /*
          * For VIA:
          *  A flag to ensure DMA position is 0
@@ -426,8 +428,25 @@ struct azx_dev {
 
         struct timecounter  azx_tc;
         struct cyclecounter azx_cc;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+        struct mutex dsp_mutex;
+#endif
 };
 
+/* DSP lock helpers */
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+#define dsp_lock_init(dev)      mutex_init(&(dev)->dsp_mutex)
+#define dsp_lock(dev)           mutex_lock(&(dev)->dsp_mutex)
+#define dsp_unlock(dev)         mutex_unlock(&(dev)->dsp_mutex)
+#define dsp_is_locked(dev)      ((dev)->locked)
+#else
+#define dsp_lock_init(dev)      do {} while (0)
+#define dsp_lock(dev)           do {} while (0)
+#define dsp_unlock(dev)         do {} while (0)
+#define dsp_is_locked(dev)      0
+#endif
+
 /* CORB/RIRB */
 struct azx_rb {
         u32 *buf;               /* CORB/RIRB buffer
@@ -527,6 +546,10 @@ struct azx {
 
         /* card list (for power_save trigger) */
         struct list_head list;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+        struct azx_dev saved_azx_dev;
+#endif
 };
 
 #define CREATE_TRACE_POINTS
@@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
                 dev = chip->capture_index_offset;
                 nums = chip->capture_streams;
         }
-        for (i = 0; i < nums; i++, dev++)
-                if (!chip->azx_dev[dev].opened) {
-                        res = &chip->azx_dev[dev];
-                        if (res->assigned_key == key)
-                                break;
+        for (i = 0; i < nums; i++, dev++) {
+                struct azx_dev *azx_dev = &chip->azx_dev[dev];
+                dsp_lock(azx_dev);
+                if (!azx_dev->opened && !dsp_is_locked(azx_dev)) {
+                        res = azx_dev;
+                        if (res->assigned_key == key) {
+                                res->opened = 1;
+                                res->assigned_key = key;
+                                dsp_unlock(azx_dev);
+                                return azx_dev;
+                        }
                 }
+                dsp_unlock(azx_dev);
+        }
         if (res) {
+                dsp_lock(res);
                 res->opened = 1;
                 res->assigned_key = key;
+                dsp_unlock(res);
         }
         return res;
 }
@@ -2009,6 +2042,12 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
         struct azx_dev *azx_dev = get_azx_dev(substream);
         int ret;
 
+        dsp_lock(azx_dev);
+        if (dsp_is_locked(azx_dev)) {
+                ret = -EBUSY;
+                goto unlock;
+        }
+
         mark_runtime_wc(chip, azx_dev, substream, false);
         azx_dev->bufsize = 0;
         azx_dev->period_bytes = 0;
@@ -2016,8 +2055,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
         ret = snd_pcm_lib_malloc_pages(substream,
                                         params_buffer_bytes(hw_params));
         if (ret < 0)
-                return ret;
+                goto unlock;
         mark_runtime_wc(chip, azx_dev, substream, true);
+ unlock:
+        dsp_unlock(azx_dev);
         return ret;
 }
 
@@ -2029,16 +2070,21 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
         struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
 
         /* reset BDL address */
-        azx_sd_writel(azx_dev, SD_BDLPL, 0);
-        azx_sd_writel(azx_dev, SD_BDLPU, 0);
-        azx_sd_writel(azx_dev, SD_CTL, 0);
-        azx_dev->bufsize = 0;
-        azx_dev->period_bytes = 0;
-        azx_dev->format_val = 0;
+        dsp_lock(azx_dev);
+        if (!dsp_is_locked(azx_dev)) {
+                azx_sd_writel(azx_dev, SD_BDLPL, 0);
+                azx_sd_writel(azx_dev, SD_BDLPU, 0);
+                azx_sd_writel(azx_dev, SD_CTL, 0);
+                azx_dev->bufsize = 0;
+                azx_dev->period_bytes = 0;
+                azx_dev->format_val = 0;
+        }
 
         snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
 
         mark_runtime_wc(chip, azx_dev, substream, false);
+        azx_dev->prepared = 0;
+        dsp_unlock(azx_dev);
         return snd_pcm_lib_free_pages(substream);
 }
 
@@ -2055,6 +2101,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
                 snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
         unsigned short ctls = spdif ? spdif->ctls : 0;
 
+        dsp_lock(azx_dev);
+        if (dsp_is_locked(azx_dev)) {
+                err = -EBUSY;
+                goto unlock;
+        }
+
         azx_stream_reset(chip, azx_dev);
         format_val = snd_hda_calc_stream_format(runtime->rate,
                                                 runtime->channels,
@@ -2065,7 +2117,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
                 snd_printk(KERN_ERR SFX
                            "%s: invalid format_val, rate=%d, ch=%d, format=%d\n",
                            pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format);
-                return -EINVAL;
+                err = -EINVAL;
+                goto unlock;
         }
 
         bufsize = snd_pcm_lib_buffer_bytes(substream);
@@ -2084,7 +2137,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
                 azx_dev->no_period_wakeup = runtime->no_period_wakeup;
                 err = azx_setup_periods(chip, substream, azx_dev);
                 if (err < 0)
-                        return err;
+                        goto unlock;
         }
 
         /* wallclk has 24Mhz clock source */
@@ -2101,8 +2154,14 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
         if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) &&
             stream_tag > chip->capture_streams)
                 stream_tag -= chip->capture_streams;
-        return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
+        err = snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
                                      azx_dev->format_val, substream);
+
+ unlock:
+        if (!err)
+                azx_dev->prepared = 1;
+        dsp_unlock(azx_dev);
+        return err;
 }
 
 static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -2117,6 +2176,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
         azx_dev = get_azx_dev(substream);
         trace_azx_pcm_trigger(chip, azx_dev, cmd);
 
+        if (dsp_is_locked(azx_dev) || !azx_dev->prepared)
+                return -EPIPE;
+
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
                 rstart = 1;
@@ -2621,17 +2683,27 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format,
         struct azx_dev *azx_dev;
         int err;
 
-        if (snd_hda_lock_devices(bus))
-                return -EBUSY;
+        azx_dev = azx_get_dsp_loader_dev(chip);
+
+        dsp_lock(azx_dev);
+        spin_lock_irq(&chip->reg_lock);
+        if (azx_dev->running || azx_dev->locked) {
+                spin_unlock_irq(&chip->reg_lock);
+                err = -EBUSY;
+                goto unlock;
+        }
+        azx_dev->prepared = 0;
+        chip->saved_azx_dev = *azx_dev;
+        azx_dev->locked = 1;
+        spin_unlock_irq(&chip->reg_lock);
 
         err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG,
                                   snd_dma_pci_data(chip->pci),
                                   byte_size, bufp);
         if (err < 0)
-                goto unlock;
+                goto err_alloc;
 
         mark_pages_wc(chip, bufp, true);
-        azx_dev = azx_get_dsp_loader_dev(chip);
         azx_dev->bufsize = byte_size;
         azx_dev->period_bytes = byte_size;
         azx_dev->format_val = format;
@@ -2649,13 +2721,20 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format,
                 goto error;
 
         azx_setup_controller(chip, azx_dev);
+        dsp_unlock(azx_dev);
         return azx_dev->stream_tag;
 
  error:
         mark_pages_wc(chip, bufp, false);
         snd_dma_free_pages(bufp);
-unlock:
-        snd_hda_unlock_devices(bus);
+ err_alloc:
+        spin_lock_irq(&chip->reg_lock);
+        if (azx_dev->opened)
+                *azx_dev = chip->saved_azx_dev;
+        azx_dev->locked = 0;
+        spin_unlock_irq(&chip->reg_lock);
+ unlock:
+        dsp_unlock(azx_dev);
         return err;
 }
 
@@ -2677,9 +2756,10 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus,
         struct azx *chip = bus->private_data;
         struct azx_dev *azx_dev = azx_get_dsp_loader_dev(chip);
 
-        if (!dmab->area)
+        if (!dmab->area || !azx_dev->locked)
                 return;
 
+        dsp_lock(azx_dev);
         /* reset BDL address */
         azx_sd_writel(azx_dev, SD_BDLPL, 0);
         azx_sd_writel(azx_dev, SD_BDLPU, 0);
@@ -2692,7 +2772,12 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus,
         snd_dma_free_pages(dmab);
         dmab->area = NULL;
 
-        snd_hda_unlock_devices(bus);
+        spin_lock_irq(&chip->reg_lock);
+        if (azx_dev->opened)
+                *azx_dev = chip->saved_azx_dev;
+        azx_dev->locked = 0;
+        spin_unlock_irq(&chip->reg_lock);
+        dsp_unlock(azx_dev);
 }
 #endif /* CONFIG_SND_HDA_DSP_LOADER */
 
@@ -3481,6 +3566,7 @@ static int azx_first_init(struct azx *chip)
         }
 
         for (i = 0; i < chip->num_streams; i++) {
+                dsp_lock_init(&chip->azx_dev[i]);
                 /* allocate memory for the BDL for each stream */
                 err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
                                           snd_dma_pci_data(chip->pci),
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index db02c1e96b08..0792b5725f9c 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 72ebb8a36b13..0d9c58f13560 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -168,10 +168,10 @@ static void cs_automute(struct hda_codec *codec)
         snd_hda_gen_update_outputs(codec);
 
         if (spec->gpio_eapd_hp) {
-                unsigned int gpio = spec->gen.hp_jack_present ?
+                spec->gpio_data = spec->gen.hp_jack_present ?
                         spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
                 snd_hda_codec_write(codec, 0x01, 0,
-                                    AC_VERB_SET_GPIO_DATA, gpio);
+                                    AC_VERB_SET_GPIO_DATA, spec->gpio_data);
         }
 }
 
@@ -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 941bf6c766ec..2a89d1eefeb6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec)
         }
 
         if (spec->beep_amp)
-                snd_hda_attach_beep_device(codec, spec->beep_amp);
+                snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
         return 0;
 }
@@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec)
         }
 
         if (spec->beep_amp)
-                snd_hda_attach_beep_device(codec, spec->beep_amp);
+                snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
         return 0;
 }
@@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec)
         }
 
         if (spec->beep_amp)
-                snd_hda_attach_beep_device(codec, spec->beep_amp);
+                snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
         return 0;
 }
@@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec)
         return 0;
 }
 
+static void cx_auto_free(struct hda_codec *codec)
+{
+        snd_hda_detach_beep_device(codec);
+        snd_hda_gen_free(codec);
+}
+
 static const struct hda_codec_ops cx_auto_patch_ops = {
         .build_controls = cx_auto_build_controls,
         .build_pcms = snd_hda_gen_build_pcms,
         .init = snd_hda_gen_init,
-        .free = snd_hda_gen_free,
+        .free = cx_auto_free,
         .unsol_event = snd_hda_jack_unsol_event,
 #ifdef CONFIG_PM
         .check_power_status = snd_hda_gen_check_power_status,
@@ -3391,7 +3397,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
         codec->patch_ops = cx_auto_patch_ops;
         if (spec->beep_amp)
-                snd_hda_attach_beep_device(codec, spec->beep_amp);
+                snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
         /* Some laptops with Conexant chips show stalls in S3 resume,
          * which falls into the single-cmd mode.
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2d4237bc0d8e..563c24df4d6f 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;
@@ -3862,6 +3863,7 @@ 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 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 83d5335ac348..dafe04ae8c72 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 2ffdc35d5ffd..806407a3973e 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/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b8d461db369f..b82bbf584146 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -573,6 +573,13 @@ 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)
 {
@@ -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,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index cd17b477781d..cdeb301da1f6 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),
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ec0efc1443ba..0e8b3aaf6c8d 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/wm8960.c b/sound/soc/codecs/wm8960.c
index 9bb927325993..a64b93425ae3 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/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index c27069d24d77..729958713cd4 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 34dc47b9581c..a294d942b9f7 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/usb/card.c b/sound/usb/card.c
index 803953a9bff3..2da8ad75fd96 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/mixer.c b/sound/usb/mixer.c
index 638e7f738018..ca4739c3f650 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
                 case UAC2_CLOCK_SELECTOR: {
                         struct uac_selector_unit_descriptor *d = p1;
                         /* call recursively to retrieve the channel info */
-                        if (check_input_term(state, d->baSourceID[0], term) < 0)
-                                return -ENODEV;
+                        err = check_input_term(state, d->baSourceID[0], term);
+                        if (err < 0)
+                                return err;
                         term->type = d->bDescriptorSubtype << 16; /* virtual type */
                         term->id = id;
                         term->name = uac_selector_unit_iSelector(d);
@@ -725,7 +726,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
                 case UAC1_PROCESSING_UNIT:
                 case UAC1_EXTENSION_UNIT:
                 /* UAC2_PROCESSING_UNIT_V2 */
-                /* UAC2_EFFECT_UNIT */ {
+                /* UAC2_EFFECT_UNIT */
+                case UAC2_EXTENSION_UNIT_V2: {
                         struct uac_processing_unit_descriptor *d = p1;
 
                         if (state->mixer->protocol == UAC_VERSION_2 &&
@@ -1356,8 +1358,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
                 return err;
 
         /* determine the input source type and name */
-        if (check_input_term(state, hdr->bSourceID, &iterm) < 0)
-                return -EINVAL;
+        err = check_input_term(state, hdr->bSourceID, &iterm);
+        if (err < 0)
+                return err;
 
         master_bits = snd_usb_combine_bytes(bmaControls, csize);
         /* master configuration quirks */
@@ -2052,6 +2055,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
                         return parse_audio_extension_unit(state, unitid, p1);
                 else /* UAC_VERSION_2 */
                         return parse_audio_processing_unit(state, unitid, p1);
+        case UAC2_EXTENSION_UNIT_V2:
+                return parse_audio_extension_unit(state, unitid, p1);
         default:
                 snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
                 return -EINVAL;
@@ -2118,7 +2123,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
                         state.oterm.type = le16_to_cpu(desc->wTerminalType);
                         state.oterm.name = desc->iTerminal;
                         err = parse_audio_unit(&state, desc->bSourceID);
-                        if (err < 0)
+                        if (err < 0 && err != -EINVAL)
                                 return err;
                 } else { /* UAC_VERSION_2 */
                         struct uac2_output_terminal_descriptor *desc = p;
@@ -2130,12 +2135,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
                         state.oterm.type = le16_to_cpu(desc->wTerminalType);
                         state.oterm.name = desc->iTerminal;
                         err = parse_audio_unit(&state, desc->bSourceID);
-                        if (err < 0)
+                        if (err < 0 && err != -EINVAL)
                                 return err;
 
                         /* for UAC2, use the same approach to also add the clock selectors */
                         err = parse_audio_unit(&state, desc->bCSourceID);
-                        if (err < 0)
+                        if (err < 0 && err != -EINVAL)
                                 return err;
                 }
         }