Surround sound support.

Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.

3 files changed, 404 insertions, 86 deletions

diff --git a/ext/dts/gstdtsdec.c b/ext/dts/gstdtsdec.c
index db4bd7de..edcc8bcc 100644
--- a/ext/dts/gstdtsdec.c
+++ b/ext/dts/gstdtsdec.c
@@ -26,6 +26,8 @@
 #include <stdlib.h>
 
 #include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
 #include <dts.h>
 
 #include "gstdtsdec.h"
@@ -180,42 +182,102 @@ gst_dtsdec_init (GstDtsDec * dtsdec)
 }
 
 static gint
-gst_dtsdec_channels (uint32_t flags)
+gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
 {
   gint chans = 0;
 
   switch (flags & DTS_CHANNEL_MASK) {
     case DTS_MONO:
       chans = 1;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 2);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+      }
       break;
-    case DTS_CHANNEL:
+      /* case DTS_CHANNEL: */
     case DTS_STEREO:
     case DTS_STEREO_SUMDIFF:
     case DTS_STEREO_TOTAL:
     case DTS_DOLBY:
       chans = 2;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 3);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+      }
       break;
     case DTS_3F:
+      chans = 3;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 4);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+      }
+      break;
     case DTS_2F1R:
       chans = 3;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 4);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+      }
       break;
     case DTS_3F1R:
+      chans = 4;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 5);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+      }
+      break;
     case DTS_2F2R:
       chans = 4;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 5);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        *pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+      }
       break;
     case DTS_3F2R:
       chans = 5;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 6);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        *pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+      }
       break;
     case DTS_4F2R:
       chans = 6;
+      if (pos) {
+        *pos = g_new (GstAudioChannelPosition, 7);
+        *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+        *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+        *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        *pos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+      }
       break;
     default:
       /* error */
       g_warning ("dtsdec: invalid flags 0x%x", flags);
       return 0;
   }
-  if (flags & DTS_LFE)
+  if (flags & DTS_LFE) {
+    if (pos) {
+      *pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE;
+    }
     chans += 1;
+  }
 
   return chans;
 }
@@ -223,8 +285,12 @@ gst_dtsdec_channels (uint32_t flags)
 static gboolean
 gst_dtsdec_renegotiate (GstDtsDec * dts)
 {
+  GstAudioChannelPosition *pos;
   GstCaps *caps = gst_caps_from_string (DTS_CAPS);
-  gint channels = gst_dtsdec_channels (dts->using_channels);
+  gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
+
+  if (!channels)
+    return FALSE;
 
   GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
       channels, dts->sample_rate);
@@ -232,6 +298,8 @@ gst_dtsdec_renegotiate (GstDtsDec * dts)
   gst_caps_set_simple (caps,
       "channels", G_TYPE_INT, channels,
       "rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
+  gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
+  g_free (pos);
 
   return gst_pad_set_explicit_caps (dts->srcpad, caps);
 }
@@ -381,7 +449,7 @@ gst_dtsdec_loop (GstElement * element)
     }
 
     samples = dts_samples (dts->state);
-    num_c = gst_dtsdec_channels (dts->using_channels);
+    num_c = gst_dtsdec_channels (dts->using_channels, NULL);
     out = gst_buffer_new_and_alloc ((SAMPLE_WIDTH / 8) * 256 * num_c);
     GST_BUFFER_TIMESTAMP (out) = timestamp;
     GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
@@ -497,7 +565,7 @@ gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-  if (!gst_library_load ("gstbytestream"))
+  if (!gst_library_load ("gstbytestream") || !gst_library_load ("gstaudio"))
     return FALSE;
 
   if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c
index 883790e2..3448f357 100644
--- a/ext/faad/gstfaad.c
+++ b/ext/faad/gstfaad.c
@@ -23,6 +23,8 @@
 
 #include <string.h>
 
+#include <gst/audio/multichannel.h>
+
 #include "gstfaad.h"
 
 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
@@ -31,35 +33,61 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
     );
 
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+#define STATIC_INT_CAPS(bpp) \
+  "audio/x-raw-int, " \
+    "endianness = (int) BYTE_ORDER, " \
+    "signed = (bool) TRUE, " \
+    "width = (int) " G_STRINGIFY (bpp) ", " \
+    "depth = (int) " G_STRINGIFY (bpp) ", " \
+    "rate = (int) [ 8000, 96000 ], " \
+    "channels = (int) [ 1, 8 ]"
+
+#define STATIC_FLOAT_CAPS(bpp) \
+  "audio/x-raw-float, " \
+    "endianness = (int) BYTE_ORDER, " \
+    "depth = (int) " G_STRINGIFY (bpp) ", " \
+    "rate = (int) [ 8000, 96000 ], " \
+    "channels = (int) [ 1, 8 ]"
+
+/*
+ * All except 16-bit integer are disabled until someone fixes FAAD.
+ * FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
+ * for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp
+ * audio, but not for any other. You'll get random segfaults, crashes
+ * and even valgrind goes crazy.
+ */
+
+#define STATIC_CAPS \
+  STATIC_INT_CAPS (16)
+#if 0
+"; "
+STATIC_INT_CAPS (24)
+    "; "
+STATIC_INT_CAPS (32)
+    "; "
+STATIC_FLOAT_CAPS (32)
+    "; "
+STATIC_FLOAT_CAPS (64)
+#endif
+     static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) BYTE_ORDER, "
-        "signed = (bool) TRUE, "
-        "width = (int) { 16, 24, 32 }, "
-        "depth = (int) { 16, 24, 32 }, "
-        "rate = (int) [ 8000, 96000 ], "
-        "channels = (int) [ 1, 6 ]; "
-        "audio/x-raw-float, "
-        "endianness = (int) BYTE_ORDER, "
-        "depth = (int) { 32, 64 }, "
-        "rate = (int) [ 8000, 96000 ], " "channels = (int) [ 1, 6 ]")
+    GST_STATIC_CAPS (STATIC_CAPS)
     );
 
-static void gst_faad_base_init (GstFaadClass * klass);
-static void gst_faad_class_init (GstFaadClass * klass);
-static void gst_faad_init (GstFaad * faad);
+     static void gst_faad_base_init (GstFaadClass * klass);
+     static void gst_faad_class_init (GstFaadClass * klass);
+     static void gst_faad_init (GstFaad * faad);
 
-static GstPadLinkReturn
-gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps);
-static GstPadLinkReturn
-gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);
-static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
-static void gst_faad_chain (GstPad * pad, GstData * data);
-static GstElementStateReturn gst_faad_change_state (GstElement * element);
+     static GstPadLinkReturn
+         gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps);
+     static GstPadLinkReturn
+         gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);
+     static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
+     static void gst_faad_chain (GstPad * pad, GstData * data);
+     static GstElementStateReturn gst_faad_change_state (GstElement * element);
 
-static GstElementClass *parent_class = NULL;
+     static GstElementClass *parent_class = NULL;
 
 /* static guint gst_faad_signals[LAST_SIGNAL] = { 0 }; */
 
@@ -123,6 +151,9 @@ gst_faad_init (GstFaad * faad)
   faad->samplerate = -1;
   faad->channels = -1;
   faad->tempbuf = NULL;
+  faad->need_channel_setup = TRUE;
+  faad->channel_positions = NULL;
+  faad->init = FALSE;
 
   GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE);
 
@@ -142,6 +173,102 @@ gst_faad_init (GstFaad * faad)
   gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);
 }
 
+/*
+ * Channel identifier conversion - caller g_free()s result!
+ */
+
+static guchar *
+gst_faad_chanpos_from_gst (GstAudioChannelPosition * pos, guint num)
+{
+  guchar *fpos = g_new (guchar, num);
+  guint n;
+
+  for (n = 0; n < num; n++) {
+    switch (pos[n]) {
+      case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
+        fpos[n] = FRONT_CHANNEL_LEFT;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
+        fpos[n] = FRONT_CHANNEL_RIGHT;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
+        fpos[n] = FRONT_CHANNEL_CENTER;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
+        fpos[n] = SIDE_CHANNEL_LEFT;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
+        fpos[n] = SIDE_CHANNEL_RIGHT;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
+        fpos[n] = BACK_CHANNEL_LEFT;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
+        fpos[n] = BACK_CHANNEL_RIGHT;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
+        fpos[n] = BACK_CHANNEL_CENTER;
+        break;
+      case GST_AUDIO_CHANNEL_POSITION_LFE:
+        fpos[n] = LFE_CHANNEL;
+        break;
+      default:
+        GST_WARNING ("Unsupported GST channel position 0x%x encountered",
+            pos[n]);
+        g_free (fpos);
+        return NULL;
+    }
+  }
+
+  return fpos;
+}
+
+static GstAudioChannelPosition *
+gst_faad_chanpos_to_gst (guchar * fpos, guint num)
+{
+  GstAudioChannelPosition *pos = g_new (GstAudioChannelPosition, num);
+  guint n;
+
+  for (n = 0; n < num; n++) {
+    switch (fpos[n]) {
+      case FRONT_CHANNEL_LEFT:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        break;
+      case FRONT_CHANNEL_RIGHT:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        break;
+      case FRONT_CHANNEL_CENTER:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        break;
+      case SIDE_CHANNEL_LEFT:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
+        break;
+      case SIDE_CHANNEL_RIGHT:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
+        break;
+      case BACK_CHANNEL_LEFT:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        break;
+      case BACK_CHANNEL_RIGHT:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+        break;
+      case BACK_CHANNEL_CENTER:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+        break;
+      case LFE_CHANNEL:
+        pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE;
+        break;
+      default:
+        GST_WARNING ("Unsupported FAAD channel position 0x%x encountered",
+            fpos[n]);
+        g_free (pos);
+        return NULL;
+    }
+  }
+
+  return pos;
+}
+
 static GstPadLinkReturn
 gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
 {
@@ -160,17 +287,20 @@ gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
             GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0)
       return GST_PAD_LINK_REFUSED;
 
-    faad->samplerate = samplerate;
-    faad->channels = channels;
+    //faad->samplerate = samplerate;
+    //faad->channels = channels;
+    faad->init = TRUE;
 
     if (faad->tempbuf) {
       gst_buffer_unref (faad->tempbuf);
       faad->tempbuf = NULL;
     }
-
-    return GST_PAD_LINK_OK;
+  } else {
+    faad->init = FALSE;
   }
 
+  faad->need_channel_setup = TRUE;
+
   /* if there's no decoderspecificdata, it's all fine. We cannot know
    * much more at this point... */
   return GST_PAD_LINK_OK;
@@ -180,27 +310,45 @@ static GstCaps *
 gst_faad_srcgetcaps (GstPad * pad)
 {
   GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
+  static GstAudioChannelPosition *supported_positions = NULL;
+  static gint num_supported_positions = LFE_CHANNEL - FRONT_CHANNEL_CENTER;
+  GstCaps *templ;
+
+  if (!supported_positions) {
+    guchar *supported_fpos = g_new0 (guchar,
+        LFE_CHANNEL - FRONT_CHANNEL_CENTER);
+    gint n;
+
+    for (n = 0; n < LFE_CHANNEL - FRONT_CHANNEL_CENTER; n++) {
+      supported_fpos[n] = n + FRONT_CHANNEL_CENTER;
+    }
+    supported_positions = gst_faad_chanpos_to_gst (supported_fpos, n);
+    g_free (supported_fpos);
+  }
 
   if (faad->handle != NULL && faad->channels != -1 && faad->samplerate != -1) {
     GstCaps *caps = gst_caps_new_empty ();
     GstStructure *str;
     gint fmt[] = {
       FAAD_FMT_16BIT,
+#if 0
       FAAD_FMT_24BIT,
       FAAD_FMT_32BIT,
       FAAD_FMT_FLOAT,
       FAAD_FMT_DOUBLE,
+#endif
       -1
     }
     , n;
 
     for (n = 0; fmt[n] != -1; n++) {
-      switch (n) {
+      switch (fmt[n]) {
         case FAAD_FMT_16BIT:
           str = gst_structure_new ("audio/x-raw-int",
               "signed", G_TYPE_BOOLEAN, TRUE,
               "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
           break;
+#if 0
         case FAAD_FMT_24BIT:
           str = gst_structure_new ("audio/x-raw-int",
               "signed", G_TYPE_BOOLEAN, TRUE,
@@ -219,6 +367,7 @@ gst_faad_srcgetcaps (GstPad * pad)
           str = gst_structure_new ("audio/x-raw-float",
               "depth", G_TYPE_INT, 64, NULL);
           break;
+#endif
         default:
           str = NULL;
           break;
@@ -234,8 +383,26 @@ gst_faad_srcgetcaps (GstPad * pad)
 
       if (faad->channels != -1) {
         gst_structure_set (str, "channels", G_TYPE_INT, faad->channels, NULL);
+
+        /* put channel information here */
+        if (faad->channel_positions) {
+          GstAudioChannelPosition *pos;
+
+          pos = gst_faad_chanpos_to_gst (faad->channel_positions,
+              faad->channels);
+          if (!pos) {
+            gst_structure_free (str);
+            continue;
+          }
+          gst_audio_set_channel_positions (str, pos);
+          g_free (pos);
+        } else {
+          gst_audio_set_structure_channel_positions_list (str,
+              supported_positions, num_supported_positions);
+        }
       } else {
-        gst_structure_set (str, "channels", GST_TYPE_INT_RANGE, 1, 6, NULL);
+        gst_structure_set (str, "channels", GST_TYPE_INT_RANGE, 1, 8, NULL);
+        /* we set channel positions later */
       }
 
       gst_structure_set (str, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
@@ -243,10 +410,20 @@ gst_faad_srcgetcaps (GstPad * pad)
       gst_caps_append_structure (caps, str);
     }
 
+    if (faad->channels == -1) {
+      gst_audio_set_caps_channel_positions_list (caps,
+          supported_positions, num_supported_positions);
+    }
+
     return caps;
   }
 
-  return gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
+  /* template with channel positions */
+  templ = gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
+  gst_audio_set_caps_channel_positions_list (templ,
+      supported_positions, num_supported_positions);
+
+  return templ;
 }
 
 static GstPadLinkReturn
@@ -258,11 +435,13 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
   gint depth, rate, channels;
   GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
 
-  if (!faad->handle || (faad->samplerate == -1 || faad->channels == -1)) {
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (!faad->handle || (faad->samplerate == -1 || faad->channels == -1) ||
+      !faad->channel_positions) {
     return GST_PAD_LINK_DELAYED;
   }
 
-  structure = gst_caps_get_structure (caps, 0);
   mimetype = gst_structure_get_name (structure);
 
   /* Samplerate and channels are normally provided through
@@ -273,6 +452,30 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
     return GST_PAD_LINK_REFUSED;
   }
 
+  /* Another internal checkup. */
+  if (faad->need_channel_setup) {
+    GstAudioChannelPosition *pos;
+    guchar *fpos;
+    guint i;
+
+    pos = gst_audio_get_channel_positions (structure);
+    if (!pos) {
+      return GST_PAD_LINK_DELAYED;
+    }
+    fpos = gst_faad_chanpos_from_gst (pos, faad->channels);
+    g_free (pos);
+    if (!fpos)
+      return GST_PAD_LINK_REFUSED;
+
+    for (i = 0; i < faad->channels; i++) {
+      if (fpos[i] != faad->channel_positions[i]) {
+        g_free (fpos);
+        return GST_PAD_LINK_REFUSED;
+      }
+    }
+    g_free (fpos);
+  }
+
   if (!strcmp (mimetype, "audio/x-raw-int")) {
     gint width;
 
@@ -286,39 +489,47 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
       case 16:
         fmt = FAAD_FMT_16BIT;
         break;
+#if 0
       case 24:
         fmt = FAAD_FMT_24BIT;
         break;
       case 32:
         fmt = FAAD_FMT_32BIT;
         break;
+#endif
     }
   } else {
     if (!gst_structure_get_int (structure, "depth", &depth))
       return GST_PAD_LINK_REFUSED;
 
     switch (depth) {
+#if 0
       case 32:
         fmt = FAAD_FMT_FLOAT;
         break;
       case 64:
         fmt = FAAD_FMT_DOUBLE;
         break;
+#endif
     }
   }
 
   if (fmt != -1) {
     faacDecConfiguration *conf;
 
+    g_print ("Set format %d\n", fmt);
     conf = faacDecGetCurrentConfiguration (faad->handle);
     conf->outputFormat = fmt;
-    faacDecSetConfiguration (faad->handle, conf);
+    g_print ("Trying to conf\n");
+    if (faacDecSetConfiguration (faad->handle, conf) == 0)
+      return GST_PAD_LINK_REFUSED;
+    g_print ("Done\n");
     /* FIXME: handle return value, how? */
     faad->bps = depth / 8;
 
     return GST_PAD_LINK_OK;
   }
-
+  g_print ("Format not recognized\n");
   return GST_PAD_LINK_REFUSED;
 }
 
@@ -329,7 +540,7 @@ gst_faad_chain (GstPad * pad, GstData * data)
   guchar *input_data;
   GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
   GstBuffer *buf, *outbuf;
-  faacDecFrameInfo info;
+  faacDecFrameInfo *info;
   void *out;
 
   if (GST_IS_EVENT (data)) {
@@ -338,18 +549,8 @@ gst_faad_chain (GstPad * pad, GstData * data)
     switch (GST_EVENT_TYPE (event)) {
       case GST_EVENT_EOS:
         if (faad->tempbuf != NULL) {
-          /* Try to decode the remaining data */
-          out = faacDecDecode (faad->handle, &info,
-              GST_BUFFER_DATA (faad->tempbuf), GST_BUFFER_SIZE (faad->tempbuf));
           gst_buffer_unref (faad->tempbuf);
           faad->tempbuf = NULL;
-          if (out && !info.error && info.samples > 0) {
-            outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
-            /* ugh */
-            memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
-
-            gst_pad_push (faad->srcpad, GST_DATA (outbuf));
-          }
         }
         gst_element_set_eos (GST_ELEMENT (faad));
         gst_pad_push (faad->srcpad, data);
@@ -360,55 +561,89 @@ gst_faad_chain (GstPad * pad, GstData * data)
     }
   }
 
+  info = g_new0 (faacDecFrameInfo, 1);
+
+  /* buffer + remaining data */
   buf = GST_BUFFER (data);
+  if (faad->tempbuf) {
+    buf = gst_buffer_join (faad->tempbuf, buf);
+    faad->tempbuf = NULL;
+  }
 
-  if (faad->samplerate == -1 || faad->channels == -1) {
-    GstPadLinkReturn ret;
+  /* init if not already done during capsnego */
+  if (!faad->init) {
     gulong samplerate;
     guchar channels;
 
     faacDecInit (faad->handle,
         GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate, &channels);
-    faad->samplerate = samplerate;
-    faad->channels = channels;
-
-    ret = gst_pad_renegotiate (faad->srcpad);
-    if (GST_PAD_LINK_FAILED (ret)) {
-      GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
-      gst_buffer_unref (buf);
-      return;
-    }
-  }
+    faad->init = TRUE;
 
-  /* Use the leftovers */
-  if (faad->tempbuf) {
-    buf = gst_buffer_join (faad->tempbuf, buf);
-    faad->tempbuf = NULL;
+    /* store for renegotiation later on */
+    info->samplerate = samplerate;
+    info->channels = channels;
+  } else {
+    info->samplerate = 0;
+    info->channels = 0;
   }
 
+  /* decode cycle */
   input_data = GST_BUFFER_DATA (buf);
   input_size = GST_BUFFER_SIZE (buf);
-  info.bytesconsumed = input_size;
-  while (input_size > (faad->channels * FAAD_MIN_STREAMSIZE)
-      && info.bytesconsumed > 0) {
-    out = faacDecDecode (faad->handle, &info, input_data, input_size);
-    if (info.error) {
+  info->bytesconsumed = input_size;
+  while (input_size >= FAAD_MIN_STREAMSIZE && info->bytesconsumed > 0) {
+    g_print ("Decoding %d bytes of data\n", input_size);
+    out = faacDecDecode (faad->handle, info, input_data, input_size);
+    g_print ("done, rec. %p\n", out);
+    if (info->error) {
       GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
-          ("Failed to decode buffer: %s", faacDecGetErrorMessage (info.error)));
+          ("Failed to decode buffer: %s",
+              faacDecGetErrorMessage (info->error)));
       break;
     }
 
-    input_size -= info.bytesconsumed;
-    input_data += info.bytesconsumed;
+    if (info->bytesconsumed > input_size)
+      info->bytesconsumed = input_size;
+    input_size -= info->bytesconsumed;
+    input_data += info->bytesconsumed;
+
+    if (out && info->samples > 0) {
+      gboolean fmt_change = FALSE;
 
-    if (out) {
+      /* see if we need to renegotiate */
+      if (info->samplerate != faad->samplerate ||
+          info->channels != faad->channels || !faad->channel_positions) {
+        fmt_change = TRUE;
+      } else {
+        gint i;
 
-      if (info.samplerate != faad->samplerate
-          || info.channels != faad->channels) {
+        for (i = 0; i < info->channels; i++) {
+          if (info->channel_position[i] != faad->channel_positions[i])
+            fmt_change = TRUE;
+        }
+      }
+
+      if (fmt_change) {
         GstPadLinkReturn ret;
 
-        faad->samplerate = info.samplerate;
-        faad->channels = info.channels;
+        g_print ("Format change\n");
+        g_print ("To %ld Hz, %d chans, %d/%d/%d/%d/%d/%d\n",
+            info->samplerate, info->channels,
+            info->channel_position[0],
+            info->channel_position[1],
+            info->channel_position[2],
+            info->channel_position[3],
+            info->channel_position[4], info->channel_position[5]);
+        /* store new negotiation information */
+        faad->samplerate = info->samplerate;
+        faad->channels = info->channels;
+        if (faad->channel_positions)
+          g_free (faad->channel_positions);
+        faad->channel_positions = g_new (guint8, faad->channels);
+        memcpy (faad->channel_positions, info->channel_position,
+            faad->channels);
+
+        /* and negotiate */
         ret = gst_pad_renegotiate (faad->srcpad);
         if (GST_PAD_LINK_FAILED (ret)) {
           GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
@@ -416,30 +651,35 @@ gst_faad_chain (GstPad * pad, GstData * data)
         }
       }
 
-      if (info.samples > 0) {
-        outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
+      /* play decoded data */
+      if (info->samples > 0) {
+        g_print ("Playing %ld samples from buf %p\n", info->samples, out);
+        outbuf = gst_buffer_new_and_alloc (info->samples * faad->bps);
         /* ugh */
         memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
+        g_print ("done, to %p\n", GST_BUFFER_DATA (outbuf));
         GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
         GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
 
         gst_pad_push (faad->srcpad, GST_DATA (outbuf));
       }
     }
-  };
+  }
 
   /* Keep the leftovers */
   if (input_size > 0) {
-    if (input_size < GST_BUFFER_SIZE (buf))
+    if (input_size < GST_BUFFER_SIZE (buf)) {
       faad->tempbuf = gst_buffer_create_sub (buf,
           GST_BUFFER_SIZE (buf) - input_size, input_size);
-    else {
+    } else {
       faad->tempbuf = buf;
       gst_buffer_ref (buf);
     }
   }
 
   gst_buffer_unref (buf);
+
+  g_free (info);
 }
 
 static GstElementStateReturn
@@ -463,6 +703,10 @@ gst_faad_change_state (GstElement * element)
     case GST_STATE_PAUSED_TO_READY:
       faad->samplerate = -1;
       faad->channels = -1;
+      faad->need_channel_setup = TRUE;
+      faad->init = FALSE;
+      g_free (faad->channel_positions);
+      faad->channel_positions = NULL;
       break;
     case GST_STATE_READY_TO_NULL:
       faacDecClose (faad->handle);
@@ -485,7 +729,8 @@ gst_faad_change_state (GstElement * element)
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-  return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
+  return gst_library_load ("gstaudio") &&
+      gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
 }
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
diff --git a/ext/faad/gstfaad.h b/ext/faad/gstfaad.h
index e5c66b03..2f048635 100644
--- a/ext/faad/gstfaad.h
+++ b/ext/faad/gstfaad.h
@@ -52,6 +52,11 @@ typedef struct _GstFaad {
 
   /* FAAD object */
   faacDecHandle handle;
+  gboolean init;
+
+  /* FAAD channel setup */
+  guchar *channel_positions;
+  gboolean need_channel_setup;
 } GstFaad;
 
 typedef struct _GstFaadClass {