%patch
Index: gst-plugins/ext/a52dec/gsta52dec.c
===================================================================
--- gst-plugins.orig/ext/a52dec/gsta52dec.c	2004-10-02 17:47:41.000000000 +0200
+++ gst-plugins/ext/a52dec/gsta52dec.c	2004-11-12 12:37:19.000000000 +0100
@@ -27,6 +27,8 @@
 #include "_stdint.h"
 
 #include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
 #include <a52dec/a52.h>
 #include <a52dec/mm_accel.h>
 #include "gsta52dec.h"
@@ -39,6 +41,12 @@
   "David I. Lehn <dlehn@users.sourceforge.net>",
 };
 
+#ifdef LIBA52_DOUBLE
+#define SAMPLE_WIDTH 64
+#else
+#define SAMPLE_WIDTH 32
+#endif
+
 GST_DEBUG_CATEGORY_STATIC (a52dec_debug);
 #define GST_CAT_DEFAULT (a52dec_debug)
 
@@ -68,12 +76,10 @@
 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw-int, "
+    GST_STATIC_CAPS ("audio/x-raw-float, "
         "endianness = (int) BYTE_ORDER, "
-        "signed = (boolean) true, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
-        "rate = (int) [ 4000, 48000 ], " "channels = (int) [ 1, 6 ]")
+        "width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", "
+        "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
     );
 
 static void gst_a52dec_base_init (gpointer g_class);
@@ -170,176 +176,108 @@
   a52dec->dynamic_range_compression = FALSE;
 }
 
-/* BEGIN modified a52dec conversion code */
-
-static inline int16_t
-convert (int32_t i)
-{
-  if (i > 0x43c07fff)
-    return 32767;
-  else if (i < 0x43bf8000)
-    return -32768;
-  else
-    return i - 0x43c00000;
-}
-
-static inline void
-float_to_int (float *_f, int16_t * s16, int flags)
-{
-  int i;
-  int32_t *f = (int32_t *) _f;
-
-  switch (flags) {
-    case A52_MONO:
-      for (i = 0; i < 256; i++) {
-        s16[5 * i] = s16[5 * i + 1] = s16[5 * i + 2] = s16[5 * i + 3] = 0;
-        s16[5 * i + 4] = convert (f[i]);
-      }
-      break;
-    case A52_CHANNEL:
-    case A52_STEREO:
-    case A52_DOLBY:
-      for (i = 0; i < 256; i++) {
-        s16[2 * i] = convert (f[i]);
-        s16[2 * i + 1] = convert (f[i + 256]);
-      }
-      break;
-    case A52_3F:
-      for (i = 0; i < 256; i++) {
-        s16[5 * i] = convert (f[i]);
-        s16[5 * i + 1] = convert (f[i + 512]);
-        s16[5 * i + 2] = s16[5 * i + 3] = 0;
-        s16[5 * i + 4] = convert (f[i + 256]);
-      }
-      break;
-    case A52_2F2R:
-      for (i = 0; i < 256; i++) {
-        s16[4 * i] = convert (f[i]);
-        s16[4 * i + 1] = convert (f[i + 256]);
-        s16[4 * i + 2] = convert (f[i + 512]);
-        s16[4 * i + 3] = convert (f[i + 768]);
-      }
-      break;
-    case A52_3F2R:
-      for (i = 0; i < 256; i++) {
-        s16[5 * i] = convert (f[i]);
-        s16[5 * i + 1] = convert (f[i + 512]);
-        s16[5 * i + 2] = convert (f[i + 768]);
-        s16[5 * i + 3] = convert (f[i + 1024]);
-        s16[5 * i + 4] = convert (f[i + 256]);
-      }
-      break;
-    case A52_MONO | A52_LFE:
-      for (i = 0; i < 256; i++) {
-        s16[6 * i] = s16[6 * i + 1] = s16[6 * i + 2] = s16[6 * i + 3] = 0;
-        s16[6 * i + 4] = convert (f[i + 256]);
-        s16[6 * i + 5] = convert (f[i]);
-      }
-      break;
-    case A52_CHANNEL | A52_LFE:
-    case A52_STEREO | A52_LFE:
-    case A52_DOLBY | A52_LFE:
-      for (i = 0; i < 256; i++) {
-        s16[6 * i] = convert (f[i + 256]);
-        s16[6 * i + 1] = convert (f[i + 512]);
-        s16[6 * i + 2] = s16[6 * i + 3] = s16[6 * i + 4] = 0;
-        s16[6 * i + 5] = convert (f[i]);
-      }
-      break;
-    case A52_3F | A52_LFE:
-      for (i = 0; i < 256; i++) {
-        s16[6 * i] = convert (f[i + 256]);
-        s16[6 * i + 1] = convert (f[i + 768]);
-        s16[6 * i + 2] = s16[6 * i + 3] = 0;
-        s16[6 * i + 4] = convert (f[i + 512]);
-        s16[6 * i + 5] = convert (f[i]);
-      }
-      break;
-    case A52_2F2R | A52_LFE:
-      for (i = 0; i < 256; i++) {
-        s16[6 * i] = convert (f[i + 256]);
-        s16[6 * i + 1] = convert (f[i + 512]);
-        s16[6 * i + 2] = convert (f[i + 768]);
-        s16[6 * i + 3] = convert (f[i + 1024]);
-        s16[6 * i + 4] = 0;
-        s16[6 * i + 5] = convert (f[i]);
-      }
-      break;
-    case A52_3F2R | A52_LFE:
-      for (i = 0; i < 256; i++) {
-        s16[6 * i] = convert (f[i + 256]);
-        s16[6 * i + 1] = convert (f[i + 768]);
-        s16[6 * i + 2] = convert (f[i + 1024]);
-        s16[6 * i + 3] = convert (f[i + 1280]);
-        s16[6 * i + 4] = convert (f[i + 512]);
-        s16[6 * i + 5] = convert (f[i]);
-      }
-      break;
-  }
-}
-
 static int
-gst_a52dec_channels (int flags)
+gst_a52dec_channels (int flags, GstAudioChannelPosition ** pos)
 {
   int chans = 0;
 
+  /* allocated just for safety. Number makes no sense */
+  if (pos) {
+    *pos = g_new (GstAudioChannelPosition, 6);
+  }
+
   if (flags & A52_LFE) {
     chans += 1;
+    *pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE;
   }
   flags &= A52_CHANNEL_MASK;
   switch (flags) {
     case A52_3F2R:
+      if (pos) {
+        *pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        *pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        *pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+      }
       chans += 5;
       break;
     case A52_2F2R:
+      if (pos) {
+        *pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+        *pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+      }
+      chans += 4;
+      break;
     case A52_3F1R:
+      if (pos) {
+        *pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        *pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+      }
       chans += 4;
       break;
     case A52_2F1R:
+      if (pos) {
+        *pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+        *pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+      }
+      chans += 3;
+      break;
     case A52_3F:
+      if (pos) {
+        *pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+        *pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+      }
       chans += 3;
-    case A52_CHANNEL:
+      break;
+    /*case A52_CHANNEL:*/
     case A52_STEREO:
     case A52_DOLBY:
+      if (pos) {
+        *pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+        *pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+      }
       chans += 2;
       break;
     default:
       /* error */
       g_warning ("a52dec invalid flags %d", flags);
+      g_free (pos);
       return 0;
   }
+
   return chans;
 }
 
 static int
-gst_a52dec_push (GstPad * srcpad, int flags, sample_t * _samples,
+gst_a52dec_push (GstPad * srcpad, int flags, sample_t * samples,
     GstClockTime timestamp)
 {
   GstBuffer *buf;
-  int chans;
-
-#ifdef LIBA52_DOUBLE
-  float samples[256 * 6];
-  int i;
-
-  for (i = 0; i < 256 * 6; i++)
-    samples[i] = _samples[i];
-#else
-  float *samples = _samples;
-#endif
-
-  flags &= A52_CHANNEL_MASK | A52_LFE;
+  int chans, n, c;
 
-  chans = gst_a52dec_channels (flags);
+  flags &= (A52_CHANNEL_MASK | A52_LFE);
+  chans = gst_a52dec_channels (flags, NULL);
   if (!chans) {
     return 1;
   }
 
   buf = gst_buffer_new ();
-  GST_BUFFER_SIZE (buf) = sizeof (int16_t) * 256 * chans;
+  GST_BUFFER_SIZE (buf) = 256 * chans * (SAMPLE_WIDTH / 8);
   GST_BUFFER_DATA (buf) = g_malloc (GST_BUFFER_SIZE (buf));
+  for (n = 0; n < 256; n++) {
+    for (c = 0; c < chans; c++) {
+      ((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
+          samples[c * 256 + n];
+    }
+  }
   GST_BUFFER_TIMESTAMP (buf) = timestamp;
-  float_to_int (samples, (int16_t *) GST_BUFFER_DATA (buf), flags);
 
   gst_pad_push (srcpad, GST_DATA (buf));
 
@@ -348,18 +286,29 @@
 
 /* END modified a52dec conversion code */
 
-static void
-gst_a52dec_reneg (GstPad * pad, int channels, int rate)
+static gboolean
+gst_a52dec_reneg (GstPad * pad)
 {
-  GST_INFO ("a52dec: reneg channels:%d rate:%d\n", channels, rate);
+  GstAudioChannelPosition *pos;
+  GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
+  gint channels = gst_a52dec_channels (a52dec->using_channels, &pos);
+  GstCaps *caps;
+
+  if (!channels)
+    return FALSE;
+
+  GST_INFO ("a52dec: reneg channels:%d rate:%d\n",
+      channels, a52dec->sample_rate);
+
+  caps = gst_caps_new_simple ("audio/x-raw-float",
+      "endianness", G_TYPE_INT, G_BYTE_ORDER,
+      "width", G_TYPE_INT, SAMPLE_WIDTH,
+      "channels", G_TYPE_INT, channels,
+      "rate", G_TYPE_INT, a52dec->sample_rate, NULL);
+  gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
+  g_free (pos);
 
-  gst_pad_set_explicit_caps (pad,
-      gst_caps_new_simple ("audio/x-raw-int",
-          "endianness", G_TYPE_INT, G_BYTE_ORDER,
-          "signed", G_TYPE_BOOLEAN, TRUE,
-          "width", G_TYPE_INT, 16,
-          "depth", G_TYPE_INT, 16,
-          "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, rate, NULL));
+  return gst_pad_set_explicit_caps (pad, caps);
 }
 
 static void
@@ -487,8 +436,8 @@
   if (need_reneg == TRUE) {
     GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d\n",
         a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
-    gst_a52dec_reneg (a52dec->srcpad,
-        gst_a52dec_channels (a52dec->using_channels), a52dec->sample_rate);
+    if (!gst_a52dec_reneg (a52dec->srcpad))
+      goto end;
   }
 
   if (a52dec->dynamic_range_compression == FALSE) {
@@ -621,7 +570,7 @@
 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, "a52dec", GST_RANK_PRIMARY,
Index: gst-plugins/ext/alsa/gstalsa.c
===================================================================
--- gst-plugins.orig/ext/alsa/gstalsa.c	2004-11-12 11:17:01.000000000 +0100
+++ gst-plugins/ext/alsa/gstalsa.c	2004-11-12 12:37:19.000000000 +0100
@@ -26,6 +26,7 @@
 #include <sys/time.h>
 
 #include "gst/gst-i18n-plugin.h"
+#include <gst/audio/multichannel.h>
 #include "gst/propertyprobe/propertyprobe.h"
 #include "gstalsa.h"
 #include "gstalsaclock.h"
@@ -853,14 +854,56 @@
 
       /* we can never use a format we can't set caps for */
       if (caps != NULL) {
+        gint n;
+
         g_assert (gst_caps_get_size (caps) == 1);
         add_channels (gst_caps_get_structure (caps, 0), min_rate, max_rate,
             min_channels, max_channels);
-        if (ret) {
-          gst_caps_append (ret, caps);
-        } else {
-          ret = caps;
+
+        /* channel configuration */
+        for (n = min_channels; n < max_channels; n++) {
+          if (snd_pcm_hw_params_test_channels (this->handle,
+                  hw_params, n) == 0) {
+            GstStructure *str;
+            GstAudioChannelPosition pos[8] = {
+              GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+              GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+              GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+              GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+              GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+              GST_AUDIO_CHANNEL_POSITION_LFE,
+              GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+              GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
+            };
+
+            switch (n) {
+              case 1:
+                pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+                break;
+              case 2:
+              case 4:
+              case 6:
+              case 8:
+                /* keep above */
+                break;
+              default:
+                /* unsupported */
+                pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
+                break;
+            }
+
+            if (pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
+              str = gst_structure_copy (gst_caps_get_structure (caps, 0));
+              gst_structure_set (str, "channels", G_TYPE_INT, n, NULL);
+              gst_audio_set_channel_positions (str, pos);
+              if (!ret) {
+                ret = gst_caps_new_empty ();
+              }
+              gst_caps_append_structure (ret, str);
+            }
+          }
         }
+        gst_caps_free (caps);
       }
     }
   }
Index: gst-plugins/ext/alsa/gstalsaplugin.c
===================================================================
--- gst-plugins.orig/ext/alsa/gstalsaplugin.c	2004-10-02 17:47:41.000000000 +0200
+++ gst-plugins/ext/alsa/gstalsaplugin.c	2004-11-12 12:37:19.000000000 +0100
@@ -56,6 +56,9 @@
 {
   int err;
 
+  if (!gst_library_load ("gstaudio"))
+    return FALSE;
+
   if (!gst_element_register (plugin, "alsamixer", GST_RANK_NONE,
           GST_TYPE_ALSA_MIXER))
     return FALSE;
Index: gst-plugins/ext/dts/gstdtsdec.c
===================================================================
--- gst-plugins.orig/ext/dts/gstdtsdec.c	2004-10-02 17:47:41.000000000 +0200
+++ gst-plugins/ext/dts/gstdtsdec.c	2004-11-12 12:37:19.000000000 +0100
@@ -26,6 +26,8 @@
 #include <stdlib.h>
 
 #include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
 #include <dts.h>
 
 #include "gstdtsdec.h"
@@ -180,42 +182,103 @@
 }
 
 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 +286,12 @@
 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 +299,8 @@
   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 +450,7 @@
     }
 
     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 +566,7 @@
 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,
Index: gst-plugins/ext/faad/gstfaad.c
===================================================================
--- gst-plugins.orig/ext/faad/gstfaad.c	2004-11-07 19:42:14.000000000 +0100
+++ gst-plugins/ext/faad/gstfaad.c	2004-11-12 16:03:02.011867944 +0100
@@ -23,6 +23,8 @@
 
 #include <string.h>
 
+#include <gst/audio/multichannel.h>
+
 #include "gstfaad.h"
 
 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
@@ -31,21 +33,45 @@
     GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
     );
 
+#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);
@@ -123,6 +149,9 @@
   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 +171,102 @@
   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 +285,20 @@
             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 +308,45 @@
 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 +365,7 @@
           str = gst_structure_new ("audio/x-raw-float",
               "depth", G_TYPE_INT, 64, NULL);
           break;
+#endif
         default:
           str = NULL;
           break;
@@ -234,8 +381,26 @@
 
       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 +408,20 @@
       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 +433,13 @@
   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 +450,30 @@
     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 +487,46 @@
       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 +537,7 @@
   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 +546,8 @@
     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 +558,89 @@
     }
   }
 
+  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;
+
+      /* 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 (out) {
+        for (i = 0; i < info->channels; i++) {
+          if (info->channel_position[i] != faad->channel_positions[i])
+            fmt_change = TRUE;
+        }
+      }
 
-      if (info.samplerate != faad->samplerate
-          || info.channels != faad->channels) {
+      if (fmt_change) {
         GstPadLinkReturn ret;
+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);
 
-        faad->samplerate = info.samplerate;
-        faad->channels = info.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 +648,35 @@
         }
       }
 
-      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 +700,10 @@
     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 +726,8 @@
 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,
Index: gst-plugins/ext/faad/gstfaad.h
===================================================================
--- gst-plugins.orig/ext/faad/gstfaad.h	2004-10-02 17:57:19.000000000 +0200
+++ gst-plugins/ext/faad/gstfaad.h	2004-11-12 12:43:05.000000000 +0100
@@ -52,6 +52,11 @@
 
   /* FAAD object */
   faacDecHandle handle;
+  gboolean init;
+
+  /* FAAD channel setup */
+  guchar *channel_positions;
+  gboolean need_channel_setup;
 } GstFaad;
 
 typedef struct _GstFaadClass {
Index: gst-plugins/ext/vorbis/vorbisdec.c
===================================================================
--- gst-plugins.orig/ext/vorbis/vorbisdec.c	2004-11-12 17:28:56.272301256 +0100
+++ gst-plugins/ext/vorbis/vorbisdec.c	2004-11-12 17:37:38.337935216 +0100
@@ -24,6 +24,7 @@
 #include "vorbisdec.h"
 #include <string.h>
 #include <gst/tag/tag.h>
+#include <gst/audio/multichannel.h>
 
 GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug);
 #define GST_CAT_DEFAULT vorbisdec_debug
@@ -53,7 +54,7 @@
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-float, "
         "rate = (int) [ 8000, 50000 ], "
-        "channels = (int) [ 1, 2 ], " "endianness = (int) BYTE_ORDER, "
+        "channels = (int) [ 1, 6 ], " "endianness = (int) BYTE_ORDER, "
 /* no ifdef in macros, please
 #ifdef GST_VORBIS_DEC_SEQUENTIAL
       "layout = \"sequential\", "
@@ -431,6 +432,7 @@
       gst_element_found_tags_for_pad (GST_ELEMENT (vd), vd->srcpad, 0, list);
     } else if (packet.packetno == 2) {
       GstCaps *caps;
+      const GstAudioChannelPosition *pos = NULL;
 
       /* done */
       vorbis_synthesis_init (&vd->vd, &vd->vi);
@@ -440,6 +442,64 @@
           "channels", G_TYPE_INT, vd->vi.channels,
           "endianness", G_TYPE_INT, G_BYTE_ORDER,
           "width", G_TYPE_INT, 32, "buffer-frames", G_TYPE_INT, 0, NULL);
+      switch (vd->vi.channels) {
+        case 1:
+        case 2:
+          /* nothing */
+          break;
+        case 3: {
+          static GstAudioChannelPosition pos3[] = {
+            GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
+          };
+          pos = pos3;
+          break;
+        }
+        case 4: {
+          static GstAudioChannelPosition pos4[] = {
+            GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+            GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
+          };
+          pos = pos4;
+          break;
+        }
+        case 5: {
+          static GstAudioChannelPosition pos5[] = {
+            GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+            GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
+          };
+          pos = pos5;
+          break;
+        }
+        case 6: {
+          static GstAudioChannelPosition pos6[] = {
+            GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+            GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+            GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+            GST_AUDIO_CHANNEL_POSITION_LFE
+          };
+          pos = pos6;
+          break;
+        }
+        default:
+          gst_data_unref (data);
+          gst_caps_free (caps);
+          GST_ELEMENT_ERROR (vd, STREAM, NOT_IMPLEMENTED, (NULL),
+              ("Unsupported channel count %d", vd->vi.channels));
+          return;
+      }
+      if (pos) {
+        gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0),
+            pos);
+      }
       if (!gst_pad_set_explicit_caps (vd->srcpad, caps)) {
         gst_caps_free (caps);
         return;
Index: gst-plugins/gst-libs/gst/audio/Makefile.am
===================================================================
--- gst-plugins.orig/gst-libs/gst/audio/Makefile.am	2004-10-05 19:07:41.000000000 +0200
+++ gst-plugins/gst-libs/gst/audio/Makefile.am	2004-11-12 12:37:19.000000000 +0100
@@ -1,15 +1,32 @@
+# variables used for enum/marshal generation
+glib_enum_headers=multichannel.h
+glib_enum_define=GST_AUDIO
+glib_enum_prefix=gst_audio
+
+built_sources = multichannel-enumtypes.c
+built_headers = multichannel-enumtypes.h
+BUILT_SOURCES = $(built_sources) $(built_headers)
+
 librarydir = $(libdir)/gstreamer-@GST_MAJORMINOR@
 
 library_LTLIBRARIES = libgstaudio.la libgstaudiofilter.la
 noinst_LTLIBRARIES = libgstaudiofilterexample.la
 
 EXTRA_DIST = gstaudiofiltertemplate.c make_filter
-CLEANFILES = gstaudiofilterexample.c
+CLEANFILES = gstaudiofilterexample.c \
+	$(BUILT_SOURCES)
 
-libgstaudio_la_SOURCES = audio.c audioclock.c
+libgstaudio_la_SOURCES = audio.c audioclock.c \
+	multichannel.c
+nodist_libgstaudio_la_SOURCES = $(built_sources)
 
 libgstaudioincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/audio
-libgstaudioinclude_HEADERS = audio.h audioclock.h gstaudiofilter.h
+libgstaudioinclude_HEADERS = \
+	audio.h \
+	audioclock.h \
+	gstaudiofilter.h \
+	multichannel.h \
+	multichannel-enumtypes.h
 
 libgstaudio_la_LIBADD =
 libgstaudio_la_CFLAGS = $(GST_CFLAGS)
@@ -25,3 +42,10 @@
 
 gstaudiofilterexample.c: $(srcdir)/make_filter $(srcdir)/gstaudiofiltertemplate.c
 	$(srcdir)/make_filter AudiofilterExample $(srcdir)/gstaudiofiltertemplate.c
+
+noinst_PROGRAMS = testchannels
+testchannels_SOURCES = testchannels.c
+testchannels_CFLAGS = $(GST_CFLAGS)
+testchannels_LDFLAGS = $(GST_LIBS)
+
+include $(top_srcdir)/common/glib-gen.mak
Index: gst-plugins/gst-libs/gst/audio/audio.c
===================================================================
--- gst-plugins.orig/gst-libs/gst/audio/audio.c	2004-10-05 19:07:41.000000000 +0200
+++ gst-plugins/gst-libs/gst/audio/audio.c	2004-11-12 12:37:19.000000000 +0100
@@ -22,6 +22,7 @@
 #endif
 
 #include "audio.h"
+#include "multichannel-enumtypes.h"
 
 #include <gst/gststructure.h>
 
@@ -267,6 +268,8 @@
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
+  gst_audio_channel_position_get_type ();
+
   return TRUE;
 }
 
Index: gst-plugins/gst-libs/gst/audio/multichannel.c
===================================================================
--- gst-plugins.orig/gst-libs/gst/audio/multichannel.c	2004-02-23 22:02:56.000000000 +0100
+++ gst-plugins/gst-libs/gst/audio/multichannel.c	2004-11-12 12:37:19.000000000 +0100
@@ -0,0 +1,625 @@
+/* GStreamer Multichannel-Audio helper functions
+ * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "multichannel.h"
+
+#define GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME "channel-positions"
+
+/*
+ * This function checks if basic assumptions apply:
+ *  - does each position occur at most once?
+ *  - do conflicting positions occur?
+ *     + front_mono vs. front_left/right
+ *     + front_center vs. front_left/right_of_center
+ *     + rear_center vs. rear_left/right
+ * It also adds some hacks that 0.8.x needs for compatibility:
+ *  - if channels == 1, are we really mono?
+ *  - if channels == 2, are we really stereo?
+ */
+
+static gboolean
+gst_audio_check_channel_positions (const GstAudioChannelPosition * pos,
+    gint channels)
+{
+  gint i, n;
+  struct {
+    GstAudioChannelPosition pos1[2];
+    GstAudioChannelPosition pos2[1];
+  } conf[] = {
+    /* front: mono <-> stereo */
+    { { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_MONO } },
+    /* front center: 2 <-> 1 */
+    { { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER },
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER } },
+    /* rear: 2 <-> 1 */
+    { { GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_REAR_CENTER } },
+    { { GST_AUDIO_CHANNEL_POSITION_INVALID } }
+  };
+
+  /* check for invalid channel positions */
+  for (n = 0; n < channels; n++) {
+    if (pos[n] == GST_AUDIO_CHANNEL_POSITION_INVALID) {
+      g_warning ("Position %d is invalid, not allowed", n);
+      return FALSE;
+    }
+  }
+
+  /* check for multiple position occurrences */
+  for (i = GST_AUDIO_CHANNEL_POSITION_INVALID + 1;
+       i < GST_AUDIO_CHANNEL_POSITION_NUM; i++) {
+    gint count = 0;
+
+    for (n = 0; n < channels; n++) {
+      if (pos[n] == i)
+        count++;
+    }
+
+    if (count > 1) {
+      g_warning ("Channel position %d occurred %d times, not allowed",
+          i, count);
+      return FALSE;
+    }
+  }
+
+  /* check for position conflicts */
+  for (i = 0; conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; i++) {
+    gboolean found1 = FALSE, found2 = FALSE;
+                                                                                
+    for (n = 0; n < channels; n++) {
+      if (pos[n] == conf[i].pos1[0] || pos[n] == conf[i].pos1[1])
+        found1 = TRUE;
+      else if (pos[n] == conf[i].pos2[0])
+        found2 = TRUE;
+    }
+
+    if (found1 && found2) {
+      g_warning ("Found conflicting channel positions %d/%d and %d",
+           conf[i].pos1[0], conf[i].pos1[1], conf[i].pos2[0]);
+      return FALSE;
+    }
+  }
+
+  /* 0.8.x evilry */
+  if ((channels == 1 && pos[0] != GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) ||
+      (channels == 2 && (pos[0] != GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT ||
+                         pos[1] != GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT))) {
+    g_warning ("0.8.x: channels=1 implies mono; channels=2 implies stereo");
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+/**
+ * gst_audio_get_channel_positions:
+ * @str: A #GstStructure to retrieve channel positions from.
+ *
+ * Retrieves a number of (fixed!) audio channel positions from
+ * the provided #GstStructure and returns it as a newly allocated
+ * array. The caller should g_free () this array. The caller
+ * should also check that the members in this #GstStructure are
+ * indeed "fixed" before calling this function.
+ *
+ * Returns: a newly allocated array containing the channel
+ * positions as provided in the given #GstStructure. Returns
+ * NULL on error.
+ */
+
+GstAudioChannelPosition *
+gst_audio_get_channel_positions (GstStructure * str)
+{
+  GstAudioChannelPosition *pos;
+  gint channels, n;
+  const GValue *pos_val_arr, *pos_val_entry;
+  gboolean res;
+  GType t;
+
+  /* get number of channels, general type checkups */
+  g_return_val_if_fail (str != NULL, NULL);
+  res = gst_structure_get_int (str, "channels", &channels);
+  g_return_val_if_fail (res, NULL);
+  g_return_val_if_fail (channels > 0, NULL);
+  pos_val_arr = gst_structure_get_value (str,
+      GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME);
+
+  /* The following checks are here to retain compatibility for plugins not
+   * implementing this property. They expect that channels=1 implies mono
+   * and channels=2 implies stereo, so we follow that.
+   * This might be removed during 0.9.x. */
+  if (!pos_val_arr && (channels == 1 || channels == 2)) {
+    pos = g_new (GstAudioChannelPosition, channels);
+    if (channels == 1) {
+      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+    } else {
+      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+      pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+    }
+    return pos;
+  }
+  g_return_val_if_fail (pos_val_arr != NULL, NULL);
+  g_return_val_if_fail (gst_value_list_get_size (pos_val_arr) == channels,
+      NULL);
+  for (n = 0; n < channels; n++) {
+    t = G_VALUE_TYPE (gst_value_list_get_value (pos_val_arr, n));
+    g_return_val_if_fail (t == GST_TYPE_AUDIO_CHANNEL_POSITION, NULL);
+  }
+
+  /* ... and fill array */
+  pos = g_new (GstAudioChannelPosition, channels);
+  for (n = 0; n < channels; n++) {
+    pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
+    pos[n] = g_value_get_enum (pos_val_entry);
+  }
+
+  if (!gst_audio_check_channel_positions (pos, channels)) {
+    g_free (pos);
+    return NULL;
+  }
+
+  return pos;
+}
+
+/**
+ * gst_audio_set_channel_positions:
+ * @str: A #GstStructure to retrieve channel positions from.
+ * @pos: an array of channel positions. The number of members
+ *       in this array should be equal to the (fixed!) number
+ *       of the "channels" property in the given #GstStructure.
+ *
+ * Adds a "channel-positions" property to the given #GstStructure,
+ * which will represent the channel positions as given in the
+ * provided #GstAudioChannelPosition array.
+ */
+
+void
+gst_audio_set_channel_positions (GstStructure * str,
+    const GstAudioChannelPosition * pos)
+{
+  GValue pos_val_arr = { 0 }, pos_val_entry = { 0 };
+  gint channels, n;
+  gboolean res;
+
+  /* get number of channels, checkups */
+  g_return_if_fail (str != NULL);
+  g_return_if_fail (pos != NULL);
+  res = gst_structure_get_int (str, "channels", &channels);
+  g_return_if_fail (res);
+  g_return_if_fail (channels > 0);
+  if (!gst_audio_check_channel_positions (pos, channels))
+    return;
+
+  /* build gvaluearray from positions */
+  g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
+  g_value_init (&pos_val_arr, GST_TYPE_FIXED_LIST);
+  for (n = 0; n < channels; n++) {
+    g_value_set_enum (&pos_val_entry, pos[n]);
+    gst_value_list_append_value (&pos_val_arr, &pos_val_entry);
+  }
+  g_value_unset (&pos_val_entry);
+
+  /* add to structure */
+  gst_structure_set_value (str,
+      GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME, &pos_val_arr);
+  g_value_unset (&pos_val_arr);
+}
+
+/**
+ * gst_audio_set_structure_channel_positions_list:
+ * @str: #GstStructure to set the list of channel positions
+ *       on.
+ * @pos: the array containing one or more possible audio
+ *       channel positions that we should add in each value
+ *       of the array in the given structure.
+ * @num_positions: the number of values in pos.
+ *
+ * Sets a (possibly non-fixed) list of possible audio channel
+ * positions (given in pos) on the given structure. The
+ * structure, after this function has been called, will contain
+ * a "channel-positions" property with an array of the size of
+ * the "channels" property value in the given structure (note
+ * that this means that the channels property in the provided
+ * structure should be fixed!). Each value in the array will
+ * contain each of the values given in the pos array.
+ */
+
+void
+gst_audio_set_structure_channel_positions_list (GstStructure * str,
+    const GstAudioChannelPosition * pos, gint num_positions)
+{
+  gint channels, n, c;
+  GValue pos_val_arr = { 0 }, pos_val_list = { 0 }, pos_val_entry = { 0 };
+  gboolean res;
+
+  /* get number of channels, general type checkups */
+  g_return_if_fail (str != NULL);
+  g_return_if_fail (num_positions > 0);
+  g_return_if_fail (pos != NULL);
+  res = gst_structure_get_int (str, "channels", &channels);
+  g_return_if_fail (res);
+  g_return_if_fail (channels > 0);
+
+  /* 0.8.x: channels=1 or channels=2 is mono/stereo, no positions needed
+   * there (we discard them anyway) */
+  if (channels == 1 || channels == 2)
+    return;
+
+  /* create the array of lists */
+  g_value_init (&pos_val_arr, GST_TYPE_FIXED_LIST);
+  g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
+  for (n = 0; n < channels; n++) {
+    g_value_init (&pos_val_list, GST_TYPE_LIST);
+    for (c = 0; c < num_positions; c++) {
+      g_value_set_enum (&pos_val_entry, pos[c]);
+      gst_value_list_append_value (&pos_val_list, &pos_val_entry);
+    }
+    gst_value_list_append_value (&pos_val_arr, &pos_val_list);
+    g_value_unset (&pos_val_list);
+  }
+  g_value_unset (&pos_val_entry);
+  gst_structure_set_value (str, GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME,
+    &pos_val_arr);
+  g_value_unset (&pos_val_arr);
+}
+
+/*
+ * Helper function for below. The structure will be conserved,
+ * but might be cut down. Any additional structures that were
+ * created will be stored in the returned caps.
+ */
+
+static GstCaps *
+add_list_to_struct (GstStructure *str,
+    const GstAudioChannelPosition * pos, gint num_positions)
+{
+  GstCaps *caps = gst_caps_new_empty ();
+  const GValue *chan_val;
+
+  chan_val = gst_structure_get_value (str, "channels");
+  if (G_VALUE_TYPE (chan_val) == G_TYPE_INT) {
+    gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
+  } else if (G_VALUE_TYPE (chan_val) == GST_TYPE_LIST) {
+    gint size;
+    const GValue *sub_val;
+
+    size = gst_value_list_get_size (chan_val);
+    sub_val = gst_value_list_get_value (chan_val, 0);
+    gst_structure_set_value (str, "channels", sub_val);
+    gst_caps_append (caps, add_list_to_struct (str, pos, num_positions));
+    while (--size > 0) {
+      str = gst_structure_copy (str);
+      sub_val = gst_value_list_get_value (chan_val, size);
+      gst_structure_set_value (str, "channels", sub_val);
+      gst_caps_append (caps, add_list_to_struct (str, pos, num_positions));
+      gst_caps_append_structure (caps, str);
+    }
+  } else if (G_VALUE_TYPE (chan_val) == GST_TYPE_INT_RANGE) {
+    gint min, max;
+
+    min = gst_value_get_int_range_min (chan_val);
+    max = gst_value_get_int_range_max (chan_val);
+
+    gst_structure_set (str, "channels", G_TYPE_INT, min, NULL);
+    gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
+    for (++min; min < max; min++) {
+      str = gst_structure_copy (str);
+      gst_structure_set (str, "channels", G_TYPE_INT, min, NULL);
+      gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
+      gst_caps_append_structure (caps, str);
+    }
+  } else {
+    g_warning ("Unknown value type for channels property");
+  }
+
+  return caps;
+}
+
+/**
+ * gst_audio_set_caps_channel_positions_list:
+ * @caps: #GstCaps to set the list of channel positions on.
+ * @pos: the array containing one or more possible audio
+ *       channel positions that we should add in each value
+ *       of the array in the given structure.
+ * @num_positions: the number of values in pos.
+ *
+ * Sets a (possibly non-fixed) list of possible audio channel
+ * positions (given in pos) on the given caps. Each of the
+ * structures of the caps, after this function has been called,
+ * will contain a "channel-positions" property with an array.
+ * Each value in the array will contain each of the values given
+ * in the pos array. Note that the size of the caps might be
+ * increased by this, since each structure with a "channel-
+ * positions" property needs to have a fixed "channels" property.
+ * The input caps is not required to have this.
+ */
+
+void
+gst_audio_set_caps_channel_positions_list (GstCaps * caps,
+    const GstAudioChannelPosition * pos, gint num_positions)
+{
+  gint size, n;
+
+  /* get number of channels, general type checkups */
+  g_return_if_fail (caps != NULL);
+  g_return_if_fail (num_positions > 0);
+  g_return_if_fail (pos != NULL);
+
+  size = gst_caps_get_size (caps);
+  for (n = 0; n < size; n++) {
+    gst_caps_append (caps, add_list_to_struct (gst_caps_get_structure (caps,
+        n), pos, num_positions));
+  }
+}
+
+/**
+ * gst_audio_fixate_channel_positions:
+ * @str: a #GstStructure containing a (possibly unfixed)
+ *       "channel-positions" property.
+ *
+ * Custom fixate function. Elements that implement some sort of
+ * channel conversion algorhithm should use this function for
+ * fixating on GstAudioChannelPosition properties. It will take
+ * care of equal channel positioning (left/right). Caller g_free()s
+ * the return value. The input properties may be (and are supposed
+ * to be) unfixed.
+ * Note that this function is mostly a hack because we currently
+ * have no way to add default fixation functions for new GTypes.
+ *
+ * Returns: fixed values that the caller could use as a fixed
+ * set of #GstAudioChannelPosition values.
+ */
+
+GstAudioChannelPosition *
+gst_audio_fixate_channel_positions (GstStructure * str)
+{
+  GstAudioChannelPosition *pos;
+  gint channels, n, num_unfixed = 0, i, c;
+  const GValue *pos_val_arr, *pos_val_entry, *pos_val;
+  gboolean res, is_stereo = TRUE;
+  GType t;
+  /*
+   * We're going to do this cluelessly. We'll make an array of values that
+   * conflict with each other and, for each iteration in this array, pick
+   * either one until all unknown values are filled. This might not work in
+   * corner cases but should work OK for the general case.
+   */
+  struct {
+    GstAudioChannelPosition pos1[2];
+    GstAudioChannelPosition pos2[1];
+  } conf[] = {
+    /* front: mono <-> stereo */
+    { { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_MONO } },
+    /* front center: 2 <-> 1 */
+    { { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER },
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER } },
+    /* rear: 2 <-> 1 */
+    { { GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_REAR_CENTER } },
+    { { GST_AUDIO_CHANNEL_POSITION_INVALID,
+        GST_AUDIO_CHANNEL_POSITION_INVALID },
+      { GST_AUDIO_CHANNEL_POSITION_LFE } },
+    { { GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_INVALID } },
+    { { GST_AUDIO_CHANNEL_POSITION_INVALID,
+        GST_AUDIO_CHANNEL_POSITION_INVALID },
+      { GST_AUDIO_CHANNEL_POSITION_INVALID } }
+  };
+  struct {
+    gint num_opt[3];
+    guint num_opts[3];
+    gboolean is_fixed[3];
+    gint choice; /* -1 is none, 0 is the two, 1 is the one */
+  } opt;
+
+  /* get number of channels, general type checkups */
+  g_return_val_if_fail (str != NULL, NULL);
+  res = gst_structure_get_int (str, "channels", &channels);
+  g_return_val_if_fail (res, NULL);
+  g_return_val_if_fail (channels > 0, NULL);
+
+  /* 0.8.x mono/stereo checks */
+  pos_val_arr = gst_structure_get_value (str,
+      GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME);
+  if (!pos_val_arr && (channels == 1 || channels == 2)) {
+    pos = g_new (GstAudioChannelPosition, channels);
+    if (channels == 1) {
+      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+    } else {
+      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+      pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+    }
+    return pos;
+  }
+  g_return_val_if_fail (pos_val_arr != NULL, NULL);
+  g_return_val_if_fail (gst_value_list_get_size (pos_val_arr) == channels,
+      NULL);
+  for (n = 0; n < channels; n++) {
+    t = G_VALUE_TYPE (gst_value_list_get_value (pos_val_arr, n));
+    g_return_val_if_fail (t == GST_TYPE_LIST ||
+        t == GST_TYPE_AUDIO_CHANNEL_POSITION, NULL);
+  }
+
+  /* all unknown, to start with */
+  pos = g_new (GstAudioChannelPosition, channels);
+  for (n = 0; n < channels; n++)
+    pos[n] = GST_AUDIO_CHANNEL_POSITION_INVALID;
+  num_unfixed = channels;
+
+  /* Iterate the array of conflicting values */
+  for (i = 0; conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID ||
+              conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; i++) {
+    /* front/center only important if not mono (obviously) */
+    if (conf[i].pos1[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER &&
+        !is_stereo)
+      continue;
+
+    /* init values */
+    for (n = 0; n < 3; n++) {
+      opt.num_opt[n] = -1;
+      opt.num_opts[n] = -1;
+      opt.is_fixed[n] = FALSE;
+    }
+
+    /* Now, we'll see for each channel if it allows for any of the values in
+     * the set of conflicting audio channel positions and keep scores. */
+    for (n = 0; n < channels; n++) {
+      /* if the channel is already taken, don't bother */
+      if (pos[n] != GST_AUDIO_CHANNEL_POSITION_INVALID)
+        continue;
+
+      pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
+      t = G_VALUE_TYPE (pos_val_entry);
+      if (t == GST_TYPE_LIST) {
+        /* This algorhythm is suboptimal. */
+        for (c = 0; c < gst_value_list_get_size (pos_val_entry); c++) {
+          pos_val = gst_value_list_get_value (pos_val_entry, c);
+          if (g_value_get_enum (pos_val) == conf[i].pos1[0] &&
+              opt.num_opts[0] > gst_value_list_get_size (pos_val_entry) &&
+              !opt.is_fixed[0]) {
+            /* Now test if the old position of num_opt[0] also allows for
+             * the other channel (which was skipped previously). If so,
+             * keep score. */
+            if (opt.num_opt[0] != -1) {
+              gint c1;
+
+              pos_val_entry = gst_value_list_get_value (pos_val_arr,
+                  opt.num_opt[0]);
+              if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) {
+                for (c1 = 0; c1 < gst_value_list_get_size (pos_val_entry);
+                     c1++) {
+                  pos_val = gst_value_list_get_value (pos_val_entry, c1);
+                  if (g_value_get_enum (pos_val) == conf[i].pos1[1] &&
+                      opt.num_opts[1] > opt.num_opts[0] &&
+                      !opt.is_fixed[1]) {
+                    opt.num_opts[1] = opt.num_opts[0];
+                    opt.num_opt[1] = opt.num_opt[0];
+                  }
+                }
+                pos_val = gst_value_list_get_value (pos_val_entry, c);
+              }
+              pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
+            }
+
+            /* and save values */
+            opt.num_opts[0] = gst_value_list_get_size (pos_val_entry);
+            opt.num_opt[0] = n;
+          } else if (g_value_get_enum (pos_val) == conf[i].pos1[1] &&
+              opt.num_opts[1] > gst_value_list_get_size (pos_val_entry) &&
+              !opt.is_fixed[1] && n != opt.num_opt[0]) {
+            opt.num_opts[1] = gst_value_list_get_size (pos_val_entry);
+            opt.num_opt[1] = n;
+          }
+
+          /* 2 goes separately, because 0/1 vs. 2 are separate */
+          if (g_value_get_enum (pos_val) == conf[i].pos2[0] &&
+              opt.num_opts[2] > gst_value_list_get_size (pos_val_entry) &&
+              !opt.is_fixed[2]) {
+            opt.num_opts[2] = gst_value_list_get_size (pos_val_entry);
+            opt.num_opt[2] = n;
+          }
+        }
+      } else {
+        if (g_value_get_enum (pos_val_entry) == conf[i].pos1[0]) {
+          opt.num_opt[0] = n;
+          opt.is_fixed[0] = TRUE;
+        } else if (g_value_get_enum (pos_val_entry) == conf[i].pos1[1]) {
+          opt.num_opt[1] = n;
+          opt.is_fixed[1] = TRUE;
+        } else if (g_value_get_enum (pos_val_entry) == conf[i].pos2[0]) {
+          opt.num_opt[2] = n;
+          opt.is_fixed[2] = TRUE;
+        }
+      }
+    }
+
+    /* check our results and choose either one */
+    if ((opt.is_fixed[0] || opt.is_fixed[1]) && opt.is_fixed[2]) {
+      g_warning ("Pre-fixated on both %d/%d and %d - conflict!",
+           conf[i].pos1[0], conf[i].pos1[1], conf[i].pos2[0]);
+      g_free (pos);
+      return NULL;
+    } else if ((opt.is_fixed[0] && opt.num_opt[1] == -1) ||
+        (opt.is_fixed[1] && opt.num_opt[0] == -1)) {
+      g_warning ("Pre-fixated one side, but other side n/a of %d/%d",
+          conf[i].pos1[0], conf[i].pos1[1]);
+      g_free (pos);
+      return NULL;
+    } else if (opt.is_fixed[0] || opt.is_fixed[1]) {
+      opt.choice = 0;
+    } else if (opt.is_fixed[2]) {
+      opt.choice = 1;
+    } else if (opt.num_opt[0] != -1 && opt.num_opt[1] != -1) {
+      opt.choice = 0;
+    } else if (opt.num_opt[2] != -1) {
+      opt.choice = 1;
+    } else {
+      opt.choice = -1;
+    }
+
+    /* stereo? Note that we keep is_stereo to TRUE if we didn't decide on
+     * any arrangement. The mono/stereo channels might be handled elsewhere
+     * which is clearly outside the scope of this element, so we cannot
+     * know and expect the application to handle that then. */
+    if (conf[i].pos2[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO &&
+        opt.choice == 1) {
+      is_stereo = FALSE;
+    }
+
+    /* now actually decide what we'll do and fixate on that */
+    if (opt.choice == 0) {
+      g_assert (conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID &&
+          conf[i].pos1[1] != GST_AUDIO_CHANNEL_POSITION_INVALID);
+      pos[opt.num_opt[0]] = conf[i].pos1[0];
+      pos[opt.num_opt[1]] = conf[i].pos1[1];
+      num_unfixed -= 2;
+    } else if (opt.choice == 1) {
+      g_assert (conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID);
+      pos[opt.num_opt[2]] = conf[i].pos2[0];
+      num_unfixed--;
+    }
+  }
+
+  /* safety check */
+  if (num_unfixed > 0) {
+    g_warning ("%d unfixed channel positions left after fixation!",
+        num_unfixed);
+    g_free (pos);
+    return NULL;
+  }
+
+  if (!gst_audio_check_channel_positions (pos, channels)) {
+    g_free (pos);
+    return NULL;
+  }
+
+  return pos;
+}
Index: gst-plugins/gst-libs/gst/audio/multichannel.h
===================================================================
--- gst-plugins.orig/gst-libs/gst/audio/multichannel.h	2004-02-23 22:02:56.000000000 +0100
+++ gst-plugins/gst-libs/gst/audio/multichannel.h	2004-11-12 12:37:19.000000000 +0100
@@ -0,0 +1,86 @@
+/* GStreamer Multichannel-Audio helper functions
+ * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_AUDIO_MULTICHANNEL_H__
+#define __GST_AUDIO_MULTICHANNEL_H__
+
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel-enumtypes.h>
+
+typedef enum {
+  GST_AUDIO_CHANNEL_POSITION_INVALID = -1,
+
+  /* Main front speakers. Mono and left/right are mututally exclusive! */
+  GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
+  GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+  GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+
+  /* rear. Left/right and center are mututally exclusive! */
+  GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
+  GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+  GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+
+  /* subwoofer/low-frequency */
+  GST_AUDIO_CHANNEL_POSITION_LFE,
+
+  /* Center front speakers. Center and left/right_of_center cannot be
+   * used together! */
+  GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+  GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
+  GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
+
+  /* sides */
+  GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+  GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
+
+  /* don't use - counter */
+  GST_AUDIO_CHANNEL_POSITION_NUM
+} GstAudioChannelPosition;
+
+/* Retrieves or sets the positions from/to a GstStructure. Only
+ * works with fixed caps, caller should check for that! Caller
+ * g_free()s result of the getter. */
+GstAudioChannelPosition *
+	gst_audio_get_channel_positions	(GstStructure *str);
+void	gst_audio_set_channel_positions	(GstStructure *str,
+					 const GstAudioChannelPosition *pos);
+
+/* Sets a (non-fixed) list of possible audio channel positions
+ * on a structure (this requires the "channels" property to
+ * be fixed!) or on a caps (here, the "channels" property may be
+ * unfixed and the caps may even contain multiple structures). */
+void	gst_audio_set_structure_channel_positions_list
+					(GstStructure *str,
+					 const GstAudioChannelPosition *pos,
+					 gint          num_positions);
+void	gst_audio_set_caps_channel_positions_list
+					(GstCaps      *caps,
+					 const GstAudioChannelPosition *pos,
+					 gint          num_positions);
+
+/* Custom fixate function. Elements that implement some sort of
+ * channel conversion algorhithm should use this function for
+ * fixating on GstAudioChannelPosition properties. It will take
+ * care of equal channel positioning (left/right). Caller g_free()s
+ * the return value. The input properties may be (and are supposed
+ * to be) unfixed. */
+GstAudioChannelPosition *
+	gst_audio_fixate_channel_positions (GstStructure *str);
+
+#endif /* __GST_AUDIO_MULTICHANNEL_H__ */
Index: gst-plugins/gst-libs/gst/audio/testchannels.c
===================================================================
--- gst-plugins.orig/gst-libs/gst/audio/testchannels.c	2004-02-23 22:02:56.000000000 +0100
+++ gst-plugins/gst-libs/gst/audio/testchannels.c	2004-11-12 12:37:19.000000000 +0100
@@ -0,0 +1,55 @@
+/* GStreamer Multichannel Test
+ * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+
+#include <multichannel.c>
+#include <multichannel-enumtypes.c>
+
+gint
+main (gint   argc,
+      gchar *argv[])
+{
+  gchar *str;
+  GstCaps *caps;
+  GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT };
+
+  /* register multichannel type */
+  gst_init (&argc, &argv);
+  gst_audio_channel_position_get_type ();
+
+  /* test some caps-string conversions */
+  caps = gst_caps_new_simple ("audio/x-raw-int",
+      "channels", G_TYPE_INT, 2, NULL);
+  str = gst_caps_to_string (caps);
+  g_print ("Test caps #1: %s\n", str);
+  g_free (str);
+  gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
+  str = gst_caps_to_string (caps);
+  g_print ("Test caps #2: %s\n", str);
+  g_free (str);
+  gst_caps_free (caps);
+
+  return 0;
+}
Index: gst-plugins/gst/audioconvert/gstaudioconvert.c
===================================================================
--- gst-plugins.orig/gst/audioconvert/gstaudioconvert.c	2004-10-29 14:59:14.000000000 +0200
+++ gst-plugins/gst/audioconvert/gstaudioconvert.c	2004-11-12 12:37:19.000000000 +0100
@@ -37,6 +37,7 @@
 #endif
 
 #include <gst/gst.h>
+#include <gst/audio/multichannel.h>
 #include <string.h>
 #include "plugin.h"
 
@@ -64,6 +65,7 @@
   gint width;
   gint rate;
   gint channels;
+  GstAudioChannelPosition *pos;
 
   /* int audio caps */
   gboolean sign;
@@ -104,6 +106,7 @@
 static void gst_audio_convert_base_init (gpointer g_class);
 static void gst_audio_convert_class_init (GstAudioConvertClass * klass);
 static void gst_audio_convert_init (GstAudioConvert * audio_convert);
+static void gst_audio_convert_dispose (GObject * obj);
 
 /* gstreamer functions */
 static void gst_audio_convert_chain (GstPad * pad, GstData * _data);
@@ -147,33 +150,35 @@
 GST_STATIC_CAPS ( \
   "audio/x-raw-int, " \
     "rate = (int) [ 1, MAX ], " \
-    "channels = (int) [ 1, 2 ], " \
+    "channels = (int) [ 1, 8 ], " \
     "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
     "width = (int) 8, " \
     "depth = (int) [ 1, 8 ], " \
     "signed = (boolean) { true, false }; " \
   "audio/x-raw-int, " \
     "rate = (int) [ 1, MAX ], " \
-    "channels = (int) [ 1, 2 ], " \
+    "channels = (int) [ 1, 8 ], " \
     "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
     "width = (int) 16, " \
     "depth = (int) [ 1, 16 ], " \
     "signed = (boolean) { true, false }; " \
   "audio/x-raw-int, " \
     "rate = (int) [ 1, MAX ], " \
-    "channels = (int) [ 1, 2 ], " \
+    "channels = (int) [ 1, 8 ], " \
     "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
     "width = (int) 32, " \
     "depth = (int) [ 1, 32 ], " \
     "signed = (boolean) { true, false }; " \
   "audio/x-raw-float, " \
     "rate = (int) [ 1, MAX ], " \
-    "channels = (int) [ 1, 2 ], " \
+    "channels = (int) [ 1, 8 ], " \
     "endianness = (int) BYTE_ORDER, " \
     "width = (int) 32, " \
     "buffer-frames = (int) [ 0, MAX ]" \
 )
 
+static GstAudioChannelPosition *supported_positions;
+
 static GstStaticPadTemplate gst_audio_convert_src_template =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
@@ -204,8 +209,16 @@
 gst_audio_convert_class_init (GstAudioConvertClass * klass)
 {
   GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  gint i;
 
   gstelement_class->change_state = gst_audio_convert_change_state;
+  gobject_class->dispose = gst_audio_convert_dispose;
+
+  supported_positions = g_new0 (GstAudioChannelPosition,
+      GST_AUDIO_CHANNEL_POSITION_NUM);
+  for (i = 0; i < GST_AUDIO_CHANNEL_POSITION_NUM; i++)
+    supported_positions[i] = i;
 }
 
 static void
@@ -233,6 +246,24 @@
 
   /* clear important variables */
   this->convert_internal = NULL;
+  this->sinkcaps.pos = NULL;
+  this->srccaps.pos = NULL;
+}
+
+static void
+gst_audio_convert_dispose (GObject * obj)
+{
+  GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
+
+  if (this->sinkcaps.pos) {
+    g_free (this->sinkcaps.pos);
+    this->sinkcaps.pos = NULL;
+  }
+
+  if (this->srccaps.pos) {
+    g_free (this->srccaps.pos);
+    this->srccaps.pos = NULL;
+  }
 }
 
 /*** GSTREAMER FUNCTIONS ******************************************************/
@@ -310,6 +341,7 @@
   for (i = size - 1; i >= 0; i--) {
     structure = gst_caps_get_structure (othercaps, i);
     gst_structure_remove_field (structure, "channels");
+    gst_structure_remove_field (structure, "channel-positions");
     gst_structure_remove_field (structure, "endianness");
     gst_structure_remove_field (structure, "width");
     gst_structure_remove_field (structure, "depth");
@@ -332,6 +364,10 @@
   caps = gst_caps_intersect (othercaps, templcaps);
   gst_caps_free (othercaps);
 
+  /* Get the channel positions in as well. */
+  gst_audio_set_caps_channel_positions_list (caps, supported_positions,
+      GST_AUDIO_CHANNEL_POSITION_NUM);
+
   return caps;
 }
 
@@ -344,10 +380,17 @@
   g_return_val_if_fail (gst_caps_is_fixed (gst_caps), FALSE);
   g_return_val_if_fail (caps != NULL, FALSE);
 
+  /* cleanup old */
+  if (caps->pos) {
+    g_free (caps->pos);
+    caps->pos = NULL;
+  }
+
   caps->endianness = G_BYTE_ORDER;
   caps->is_int =
       (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0);
   if (!gst_structure_get_int (structure, "channels", &caps->channels)
+      || !(caps->pos = gst_audio_get_channel_positions (structure))
       || !gst_structure_get_int (structure, "width", &caps->width)
       || !gst_structure_get_int (structure, "rate", &caps->rate)
       || (caps->is_int
@@ -359,10 +402,14 @@
           && !gst_structure_get_int (structure, "buffer-frames",
               &caps->buffer_frames))) {
     GST_DEBUG ("could not get some values from structure");
+    g_free (caps->pos);
+    caps->pos = NULL;
     return FALSE;
   }
   if (caps->is_int && caps->depth > caps->width) {
     GST_DEBUG ("width > depth, not allowed - make us advertise correct caps");
+    g_free (caps->pos);
+    caps->pos = NULL;
     return FALSE;
   }
   return TRUE;
@@ -386,6 +433,7 @@
   this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
   otherpad = (pad == this->src ? this->sink : this->src);
 
+  ac_caps.pos = NULL;
   if (!gst_audio_convert_parse_caps (caps, &ac_caps))
     return GST_PAD_LINK_REFUSED;
 
@@ -406,8 +454,10 @@
     }
   }
   if (this->sink == pad) {
+    g_free (this->sinkcaps.pos);
     this->sinkcaps = ac_caps;
   } else {
+    g_free (this->srccaps.pos);
     this->srccaps = ac_caps;
   }
   GST_LOG_OBJECT (this, "trying to set caps to %" GST_PTR_FORMAT, othercaps);
@@ -419,18 +469,23 @@
   /* woohoo, got it */
   othercaps = (GstCaps *) gst_pad_get_negotiated_caps (otherpad);
   if (othercaps) {
+    other_ac_caps.pos = NULL;
     if (!gst_audio_convert_parse_caps (othercaps, &other_ac_caps)) {
       g_critical ("internal negotiation error");
       return GST_PAD_LINK_REFUSED;
     }
   } else {
     other_ac_caps = ac_caps;
+    other_ac_caps.pos = g_memdup (ac_caps.pos,
+        ac_caps.channels * sizeof (GstAudioChannelPosition));
   }
 
   if (this->sink == pad) {
+    g_free (this->srccaps.pos);
     this->srccaps = other_ac_caps;
     this->sinkcaps = ac_caps;
   } else {
+    g_free (this->sinkcaps.pos);
     this->srccaps = ac_caps;
     this->sinkcaps = other_ac_caps;
   }
@@ -489,6 +544,7 @@
 static GstCaps *
 gst_audio_convert_fixate (GstPad * pad, const GstCaps * caps)
 {
+  const GValue *pos_val;
   GstAudioConvert *this =
       GST_AUDIO_CONVERT (gst_object_get_parent (GST_OBJECT (pad)));
   GstPad *otherpad = (pad == this->sink ? this->src : this->sink);
@@ -516,7 +572,27 @@
     return copy;
   if (_fixate_caps_to_int (&copy, "endianness", try.endianness))
     return copy;
-
+  if ((pos_val = gst_structure_get_value (gst_caps_get_structure (copy, 0),
+          "channel-positions")) != NULL) {
+    GstAudioChannelPosition *pos;
+    const GValue *pos_val_entry;
+    gint i;
+
+    for (i = 0; i < gst_value_list_get_size (pos_val); i++) {
+      pos_val_entry = gst_value_list_get_value (pos_val, i);
+      if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) {
+        /* unfixed */
+        pos = gst_audio_fixate_channel_positions (
+            gst_caps_get_structure (copy, 0));
+        if (pos) {
+          gst_audio_set_channel_positions (
+              gst_caps_get_structure (copy, 0), pos);
+          g_free (pos);
+          return copy;
+        }
+      }
+    }
+  }
 
   gst_caps_free (copy);
   return NULL;
@@ -786,28 +862,128 @@
 gst_audio_convert_channels (GstAudioConvert * this, GstBuffer * buf)
 {
   GstBuffer *ret;
-  gint i, count;
+  gint c, i, count, ci, co;
   gint32 *src, *dest;
-
-  if (this->sinkcaps.channels == this->srccaps.channels)
-    return buf;
+  /* Conversions from one-channel to compatible two-channel configs */
+  struct {
+    GstAudioChannelPosition pos1[2];
+    GstAudioChannelPosition pos2[1];
+  } conv[] = {
+    /* front: mono <-> stereo */
+    { { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_MONO } },
+    /* front center: 2 <-> 1 */
+    { { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER },
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER } },
+    /* rear: 2 <-> 1 */
+    { { GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT },
+      { GST_AUDIO_CHANNEL_POSITION_REAR_CENTER } },
+    { { GST_AUDIO_CHANNEL_POSITION_INVALID } }
+  };
+  gboolean set[8] = { FALSE, FALSE, FALSE, FALSE, FALSE, FALSE, FALSE, FALSE };
+
+  if (this->sinkcaps.channels == this->srccaps.channels) {
+    for (i = 0; i < this->sinkcaps.channels; i++) {
+      if (this->sinkcaps.pos[i] != this->srccaps.pos[i])
+        break;
+    }
+    if (i == this->sinkcaps.channels)
+      return buf;
+  }
 
   count = GST_BUFFER_SIZE (buf) / 4 / this->sinkcaps.channels;
   ret = gst_audio_convert_get_buffer (buf, count * 4 * this->srccaps.channels);
+
+  /* conversions from compatible (but not the same) channel schemes. This
+   * goes two ways: if the sink has both pos1[0,1] and src has pos2[0] or
+   * if the src has both pos1[0,1] and sink has pos2[0], then we do the
+   * conversion. We hereby assume that the existance of pos1[0,1] and
+   * pos2[0] are mututally exclusive. There are no checks for that,
+   * unfortunately. This shouldn't lead to issues (like crashes or so),
+   * though. */
+  for (c = 0; conv[c].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; c++) {
+    gint pos1_0 = -1, pos1_1 = -1, pos2_0 = -1, n;
+
+    /* Try to go from the given 2 channels to the given 1 channel */
+    for (n = 0; n < this->sinkcaps.channels; n++) {
+      if (this->sinkcaps.pos[n] == conv[c].pos1[0])
+        pos1_0 = n;
+      else if (this->sinkcaps.pos[n] == conv[c].pos1[1])
+        pos1_1 = n;
+    }
+    for (n = 0; n < this->srccaps.channels; n++) {
+      if (this->srccaps.pos[n] == conv[c].pos2[0])
+        pos2_0 = n;
+    }
+
+    if (pos1_0 != -1 && pos1_1 != -1 && pos2_0 != -1) {
+      src = (gint32 *) GST_BUFFER_DATA (buf);
+      dest = (gint32 *) GST_BUFFER_DATA (ret);
+
+      for (i = 0; i < count; i++) {
+        dest[pos2_0] = (src[pos1_0] >> 1) + (src[pos1_1] >> 1) +
+            ((src[pos1_0] & 1) & (src[pos1_1] & 1));
+        src += this->sinkcaps.channels;
+        dest += this->srccaps.channels;
+      }
+      set[pos2_0] = TRUE;
+    }
+
+    /* Try to go from the given 1 channel to the given 2 channels */
+    pos1_0 = -1; pos1_1 = -1; pos2_0 = -1;
+
+    for (n = 0; n < this->srccaps.channels; n++) {
+      if (this->srccaps.pos[n] == conv[c].pos1[0])
+        pos1_0 = n;
+      else if (this->srccaps.pos[n] == conv[c].pos1[1])
+        pos1_1 = n;
+    }
+    for (n = 0; n < this->sinkcaps.channels; n++) {
+      if (this->sinkcaps.pos[n] == conv[c].pos2[0])
+        pos2_0 = n;
+    }
+                                                                                
+    if (pos1_0 != -1 && pos1_1 != -1 && pos2_0 != -1) {
+      src = (gint32 *) GST_BUFFER_DATA (buf);
+      dest = (gint32 *) GST_BUFFER_DATA (ret);
+
+      for (i = 0; i < count; i++) {
+        dest[pos1_0] = dest[pos1_1] = src[pos2_0];
+        src += this->sinkcaps.channels;
+        dest += this->srccaps.channels;        
+      }
+      set[pos1_0] = set[pos1_1] = TRUE;
+    }
+  }
+
+  /* reset data pointers */
   src = (gint32 *) GST_BUFFER_DATA (buf);
   dest = (gint32 *) GST_BUFFER_DATA (ret);
 
-  if (this->sinkcaps.channels > this->srccaps.channels) {
-    for (i = 0; i < count; i++) {
-      *dest = *src >> 1;
-      src++;
-      *dest += (*src >> 1) + (*src & 1);
-      src++;
-      dest++;
+  /* Apart from the compatible channel assignments, we can also have
+   * same channel assignments. This is much simpler, we simply copy
+   * the value from source to dest! */
+  for (co = 0; co < this->srccaps.channels; co++) {
+    /* find a channel in input with same position */
+    for (ci = 0; ci < this->sinkcaps.channels; ci++) {
+      if (this->sinkcaps.pos[ci] == this->srccaps.pos[co]) {
+        for (i = 0; i < count; i++) {
+          dest[i * this->srccaps.channels + co] =
+              src[i * this->sinkcaps.channels + ci];
+        }
+        set[co] = TRUE;
+        break;
+      }
     }
-  } else {
-    for (i = count - 1; i >= 0; i--) {
-      dest[2 * i] = dest[2 * i + 1] = src[i];
+
+    /* if not found, then silence */
+    if (ci == this->sinkcaps.channels && !set[co]) {
+      for (i = 0; i < count; i++) {
+        dest[i * this->srccaps.channels + co] = 0;
+      }
     }
   }
 
Index: gst-plugins/gst/audioconvert/plugin.c
===================================================================
--- gst-plugins.orig/gst/audioconvert/plugin.c	2004-10-02 17:47:41.000000000 +0200
+++ gst-plugins/gst/audioconvert/plugin.c	2004-11-12 12:37:19.000000000 +0100
@@ -28,6 +28,9 @@
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
+  if (!gst_library_load ("gstaudio"))
+    return FALSE;
+
   if (!gst_element_register (plugin, "audioconvert",
           GST_RANK_PRIMARY, gst_audio_convert_get_type ()) ||
       !gst_element_register (plugin, "buffer-frames-convert",
