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

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

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

8 files changed, 184 insertions, 41 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index ce6581c8ca26..4499bd948860 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -912,7 +912,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     models depending on the codec chip.  The list of available models
     is found in HD-Audio-Models.txt
 
-    The model name "genric" is treated as a special case.  When this
+    The model name "generic" is treated as a special case.  When this
     model is given, the driver uses the generic codec parser without
     "codec-patch".  It's sometimes good for testing and debugging.
 
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
index d9776cf60c07..9663b45f6fde 100644
--- a/Documentation/sound/alsa/seq_oss.html
+++ b/Documentation/sound/alsa/seq_oss.html
@@ -285,7 +285,7 @@ sample data.
 <H4>
 7.2.4 Close Callback</H4>
 The <TT>close</TT> callback is called when this device is closed by the
-applicaion. If any private data was allocated in open callback, it must
+application. If any private data was allocated in open callback, it must
 be released in the close callback. The deletion of ALSA port should be
 done here, too. This callback must not be NULL.
 <H4>
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a9ebcf9e3710..ecdf30eb5879 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3144,7 +3144,7 @@ static unsigned int convert_to_spdif_status(unsigned short val)
         if (val & AC_DIG1_PROFESSIONAL)
                 sbits |= IEC958_AES0_PROFESSIONAL;
         if (sbits & IEC958_AES0_PROFESSIONAL) {
-                if (sbits & AC_DIG1_EMPHASIS)
+                if (val & AC_DIG1_EMPHASIS)
                         sbits |= IEC958_AES0_PRO_EMPHASIS_5015;
         } else {
                 if (val & AC_DIG1_EMPHASIS)
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 78897d05d80f..43c2ea539561 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -995,6 +995,8 @@ enum {
         BAD_NO_EXTRA_SURR_DAC = 0x101,
         /* Primary DAC shared with main surrounds */
         BAD_SHARED_SURROUND = 0x100,
+        /* No independent HP possible */
+        BAD_NO_INDEP_HP = 0x40,
         /* Primary DAC shared with main CLFE */
         BAD_SHARED_CLFE = 0x10,
         /* Primary DAC shared with extra surrounds */
@@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx)
         return snd_hda_get_path_idx(codec, path);
 }
 
+/* check whether the independent HP is available with the current config */
+static bool indep_hp_possible(struct hda_codec *codec)
+{
+        struct hda_gen_spec *spec = codec->spec;
+        struct auto_pin_cfg *cfg = &spec->autocfg;
+        struct nid_path *path;
+        int i, idx;
+
+        if (cfg->line_out_type == AUTO_PIN_HP_OUT)
+                idx = spec->out_paths[0];
+        else
+                idx = spec->hp_paths[0];
+        path = snd_hda_get_path_from_idx(codec, idx);
+        if (!path)
+                return false;
+
+        /* assume no path conflicts unless aamix is involved */
+        if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid))
+                return true;
+
+        /* check whether output paths contain aamix */
+        for (i = 0; i < cfg->line_outs; i++) {
+                if (spec->out_paths[i] == idx)
+                        break;
+                path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]);
+                if (path && is_nid_contained(path, spec->mixer_nid))
+                        return false;
+        }
+        for (i = 0; i < cfg->speaker_outs; i++) {
+                path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]);
+                if (path && is_nid_contained(path, spec->mixer_nid))
+                        return false;
+        }
+
+        return true;
+}
+
 /* fill the empty entries in the dac array for speaker/hp with the
  * shared dac pointed by the paths
  */
@@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
                 badness += BAD_MULTI_IO;
         }
 
+        if (spec->indep_hp && !indep_hp_possible(codec))
+                badness += BAD_NO_INDEP_HP;
+
         /* re-fill the shared DAC for speaker / headphone */
         if (cfg->line_out_type != AUTO_PIN_HP_OUT)
                 refill_shared_dacs(codec, cfg->hp_outs,
@@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec)
                                 cfg->speaker_pins, val);
         }
 
+        /* clear indep_hp flag if not available */
+        if (spec->indep_hp && !indep_hp_possible(codec))
+                spec->indep_hp = 0;
+
         kfree(best_cfg);
         return 0;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4cea6bb6fade..418bfc0eb0a3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -415,6 +415,8 @@ struct azx_dev {
         unsigned int opened :1;
         unsigned int running :1;
         unsigned int irq_pending :1;
+        unsigned int prepared:1;
+        unsigned int locked:1;
         /*
          * For VIA:
          *  A flag to ensure DMA position is 0
@@ -426,8 +428,25 @@ struct azx_dev {
 
         struct timecounter  azx_tc;
         struct cyclecounter azx_cc;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+        struct mutex dsp_mutex;
+#endif
 };
 
+/* DSP lock helpers */
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+#define dsp_lock_init(dev)      mutex_init(&(dev)->dsp_mutex)
+#define dsp_lock(dev)           mutex_lock(&(dev)->dsp_mutex)
+#define dsp_unlock(dev)         mutex_unlock(&(dev)->dsp_mutex)
+#define dsp_is_locked(dev)      ((dev)->locked)
+#else
+#define dsp_lock_init(dev)      do {} while (0)
+#define dsp_lock(dev)           do {} while (0)
+#define dsp_unlock(dev)         do {} while (0)
+#define dsp_is_locked(dev)      0
+#endif
+
 /* CORB/RIRB */
 struct azx_rb {
         u32 *buf;               /* CORB/RIRB buffer
@@ -527,6 +546,10 @@ struct azx {
 
         /* card list (for power_save trigger) */
         struct list_head list;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+        struct azx_dev saved_azx_dev;
+#endif
 };
 
 #define CREATE_TRACE_POINTS
@@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
                 dev = chip->capture_index_offset;
                 nums = chip->capture_streams;
         }