opus: multichannel support

11 files changed, 415 insertions, 116 deletions

diff --git a/ext/opus/Makefile.am b/ext/opus/Makefile.am
index 88845a3ca..cb0a9b338 100644
--- a/ext/opus/Makefile.am
+++ b/ext/opus/Makefile.am
@@ -1,6 +1,6 @@
 plugin_LTLIBRARIES = libgstopus.la
 
-libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c
+libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c
 libgstopus_la_CFLAGS = \
         -DGST_USE_UNSTABLE_API \
 	$(GST_PLUGINS_BASE_CFLAGS) \
@@ -15,4 +15,4 @@ libgstopus_la_LIBADD = \
 libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
 libgstopus_la_LIBTOOLFLAGS = --tag=disable-static
 
-noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h
+noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h
diff --git a/ext/opus/gstopuscommon.c b/ext/opus/gstopuscommon.c
new file mode 100644
index 000000000..fc3e0376b
--- /dev/null
+++ b/ext/opus/gstopuscommon.c
@@ -0,0 +1,72 @@
+/* GStreamer
+ * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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.
+ */
+
+#include "gstopuscommon.h"
+
+/* http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9 */
+/* copy of the same structure in the vorbis plugin */
+const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
+  {                             /* Mono */
+      GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
+  {                             /* Stereo */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+  {                             /* Stereo + Centre */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+  {                             /* Quadraphonic */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+      },
+  {                             /* Stereo + Centre + rear stereo */
+        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,
+      },
+  {                             /* Full 5.1 Surround */
+        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,
+      },
+  {                             /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
+      GST_AUDIO_CHANNEL_POSITION_LFE},
+  {                             /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+      GST_AUDIO_CHANNEL_POSITION_LFE},
+};
diff --git a/ext/opus/gstopuscommon.h b/ext/opus/gstopuscommon.h
new file mode 100644
index 000000000..96a303e30
--- /dev/null
+++ b/ext/opus/gstopuscommon.h
@@ -0,0 +1,33 @@
+/* GStreamer Opus Encoder
+ * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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_OPUS_COMMON_H__
+#define __GST_OPUS_COMMON_H__
+
+#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
+G_BEGIN_DECLS
+
+extern const GstAudioChannelPosition gst_opus_channel_positions[][8];
+
+G_END_DECLS
+
+#endif /* __GST_OPUS_COMMON_H__ */
diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c
index f8b39ba08..58283973a 100644
--- a/ext/opus/gstopusdec.c
+++ b/ext/opus/gstopusdec.c
@@ -45,6 +45,7 @@
 #include <string.h>
 #include <gst/tag/tag.h>
 #include "gstopusheader.h"
+#include "gstopuscommon.h"
 #include "gstopusdec.h"
 
 GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
@@ -56,7 +57,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
         "rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
-        "channels = (int) [ 1, 2 ], "
+        "channels = (int) [ 1, 8 ], "
         "endianness = (int) BYTE_ORDER, "
         "signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
     );
@@ -217,26 +218,91 @@ gst_opus_dec_get_r128_volume (gint16 r128_gain)
 static GstFlowReturn
 gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
 {
-  g_return_val_if_fail (gst_opus_header_is_header (buf, "OpusHead", 8),
-      GST_FLOW_ERROR);
-  g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 19, GST_FLOW_ERROR);
+  const guint8 *data = GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  GstStructure *s;
+  const GstAudioChannelPosition *pos = NULL;
 
-  dec->pre_skip = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 10);
-  dec->r128_gain = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 14);
+  g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
+  g_return_val_if_fail (dec->n_channels != data[9], GST_FLOW_ERROR);
+
+  dec->n_channels = data[9];
+  dec->pre_skip = GST_READ_UINT16_LE (data + 10);
+  dec->r128_gain = GST_READ_UINT16_LE (data + 14);
   dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
   GST_INFO_OBJECT (dec,
       "Found pre-skip of %u samples, R128 gain %d (volume %f)",
       dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
 
-  dec->channel_mapping_family = GST_BUFFER_DATA (buf)[18];
-  if (dec->channel_mapping_family != 0) {
-    GST_ELEMENT_ERROR (dec, STREAM, DECODE,
-        ("Decoding error: unsupported channel nmapping family %d",
-            dec->channel_mapping_family), (NULL));
-    return GST_FLOW_ERROR;
+  dec->channel_mapping_family = data[18];
+  if (dec->channel_mapping_family == 0) {
+    /* implicit mapping */
+    GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
+    dec->n_streams = dec->n_stereo_streams = 1;
+    dec->channel_mapping[0] = 0;
+    dec->channel_mapping[1] = 1;
+  } else {
+    dec->n_streams = data[19];
+    dec->n_stereo_streams = data[20];
+    memcpy (dec->channel_mapping, data + 21, dec->n_channels);
+
+    if (dec->channel_mapping_family == 1) {
+      GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
+      switch (dec->n_channels) {
+        case 1:
+        case 2:
+          /* nothing */
+          break;
+        case 3:
+        case 4:
+        case 5:
+        case 6:
+        case 7:
+        case 8:
+          pos = gst_opus_channel_positions[dec->n_channels - 1];
+          break;
+        default:{
+          gint i;
+          GstAudioChannelPosition *posn =
+              g_new (GstAudioChannelPosition, dec->n_channels);
+
+          GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
+              (NULL), ("Using NONE channel layout for more than 8 channels"));
+
+          for (i = 0; i < dec->n_channels; i++)
+            posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+
+          pos = posn;
+        }
+      }
+    } else {
+      GST_INFO_OBJECT (dec, "Channel mapping family %d",
+          dec->channel_mapping_family);
+    }
+  }
+
+  /* negotiate width with downstream */
+  caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
+  s = gst_caps_get_structure (caps, 0);
+  gst_structure_fixate_field_nearest_int (s, "rate", 48000);
+  gst_structure_get_int (s, "rate", &dec->sample_rate);
+  gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
+  gst_structure_get_int (s, "channels", &dec->n_channels);
+
+  GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
+      dec->sample_rate);
+
+  if (pos) {
+    gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
+  }
+
+  if (dec->n_channels > 8) {
+    g_free ((GstAudioChannelPosition *) pos);
   }
-  dec->channel_mapping[0] = 0;
-  dec->channel_mapping[1] = 1;
+
+  GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
+  gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
+  gst_caps_unref (caps);
 
   return GST_FLOW_OK;
 }
@@ -248,48 +314,6 @@ gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
   return GST_FLOW_OK;
 }
 
-static void
-gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
-{
-  GstPad *srcpad, *peer;
-  GstStructure *s;
-  GstCaps *caps;
-  const GstCaps *template_caps;
-  const GstCaps *peer_caps;
-
-  srcpad = GST_AUDIO_DECODER_SRC_PAD (dec);
-  peer = gst_pad_get_peer (srcpad);
-
-  if (peer) {
-    template_caps = gst_pad_get_pad_template_caps (srcpad);
-    peer_caps = gst_pad_get_caps (peer);
-    GST_DEBUG_OBJECT (dec, "Peer caps: %" GST_PTR_FORMAT, peer_caps);
-    caps = gst_caps_intersect (template_caps, peer_caps);
-    gst_pad_fixate_caps (peer, caps);
-    GST_DEBUG_OBJECT (dec, "Fixated caps: %" GST_PTR_FORMAT, caps);
-
-    s = gst_caps_get_structure (caps, 0);
-    if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
-      dec->n_channels = 2;
-      GST_WARNING_OBJECT (dec, "Failed to get channels, using default %d",
-          dec->n_channels);
-    } else {
-      GST_DEBUG_OBJECT (dec, "Got channels %d", dec->n_channels);
-    }
-    if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
-      dec->sample_rate = 48000;
-      GST_WARNING_OBJECT (dec, "Failed to get rate, using default %d",
-          dec->sample_rate);
-    } else {
-      GST_DEBUG_OBJECT (dec, "Got sample rate %d", dec->sample_rate);
-    }
-
-    gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
-  } else {
-    GST_WARNING_OBJECT (dec, "Failed to get src pad peer");
-  }
-}
-
 static GstFlowReturn
 opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
 {
@@ -304,12 +328,11 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
   GstBuffer *buf;
 
   if (dec->state == NULL) {
-    gst_opus_dec_setup_from_peer_caps (dec);
-
     GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
         dec->n_channels, dec->sample_rate);
     dec->state = opus_multistream_decoder_create (dec->sample_rate,
-        dec->n_channels, 1, 1, dec->channel_mapping, &err);
+        dec->n_channels, dec->n_streams, dec->n_stereo_streams,
+        dec->channel_mapping, &err);
     if (!dec->state || err != OPUS_OK)
       goto creation_failed;
   }
diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h
index aa04b814d..3ccfa2696 100644
--- a/ext/opus/gstopusdec.h
+++ b/ext/opus/gstopusdec.h
@@ -55,6 +55,9 @@ struct _GstOpusDec {
   int n_channels;
   guint32 pre_skip;
   gint16 r128_gain;
+
+  guint8 n_streams;
+  guint8 n_stereo_streams;
   guint8 channel_mapping_family;
   guint8 channel_mapping[256];
 
diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c
index 93e00aa38..6dad53115 100644
--- a/ext/opus/gstopusenc.c
+++ b/ext/opus/gstopusenc.c
@@ -49,6 +49,7 @@
 #include <gst/gsttagsetter.h>
 #include <gst/audio/audio.h>
 #include "gstopusheader.h"
+#include "gstopuscommon.h"
 #include "gstopusenc.h"
 
 GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
@@ -116,8 +117,8 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
-        "rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
-        "channels = (int) [ 1, 2 ], "
+        "rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
+        "channels = (int) [ 1, 8 ], "
         "endianness = (int) BYTE_ORDER, "
         "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
     );
@@ -419,6 +420,82 @@ gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
   return frame_samples;
 }
 
+static void
+gst_opus_enc_setup_channel_mapping (GstOpusEnc * enc, const GstAudioInfo * info)
+{
+#define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
+
+  int n;
+
+  GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
+      enc->n_channels);
+
+  /* Start by setting up a default trivial mapping */
+  for (n = 0; n < 255; ++n)
+    enc->channel_mapping[n] = n;
+
+  /* For one channel, use the basic RTP mapping */
+  if (enc->n_channels == 1) {
+    GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
+    enc->channel_mapping_family = 0;
+    enc->channel_mapping[0] = 0;
+    return;
+  }
+
+  /* For two channels, use the basic RTP mapping if the channels are
+     mapped as left/right. */
+  if (enc->n_channels == 2) {
+    if (MAPS (0, FRONT_LEFT) && MAPS (1, FRONT_RIGHT)) {
+      GST_INFO_OBJECT (enc, "Stereo, canonical mapping");
+      enc->channel_mapping_family = 0;
+      /* The channel mapping is implicit for family 0, that's why we do not
+         attempt to create one for right/left - this will be mapped to the
+         Vorbis mapping below. */
+    } else {
+      GST_DEBUG_OBJECT (enc, "Stereo, but not canonical mapping, continuing");
+    }
+  }
+
+  /* For channels between 1 and 8, we use the Vorbis mapping if we can
+     find a permutation that matches it. Mono will have been taken care
+     of earlier, but this code also handles it. */
+  if (enc->n_channels >= 1 && enc->n_channels <= 8) {
+    GST_DEBUG_OBJECT (enc,
+        "In range for the Vorbis mapping, checking channel positions");
+    for (n = 0; n < enc->n_channels; ++n) {
+      GstAudioChannelPosition pos = GST_AUDIO_INFO_POSITION (info, n);
+      int c;
+
+      GST_DEBUG_OBJECT (enc, "Channel %d has position %d", n, pos);
+      for (c = 0; c < enc->n_channels; ++c) {
+        if (gst_opus_channel_positions[enc->n_channels - 1][c] == pos) {
+          GST_DEBUG_OBJECT (enc, "Found in Vorbis mapping as channel %d", c);
+          break;
+        }
+      }
+      if (c == enc->n_channels) {
+        /* We did not find that position, so use undefined */
+        GST_WARNING_OBJECT (enc,
+            "Position %d not found in Vorbis mapping, using unknown mapping",
+            pos);
+        enc->channel_mapping_family = 255;
+        return;
+      }
+      GST_DEBUG_OBJECT (enc, "Mapping output channel %d to %d", c, n);
+      enc->channel_mapping[c] = n;
+    }
+    GST_INFO_OBJECT (enc, "Permutation found, using Vorbis mapping");
+    enc->channel_mapping_family = 1;
+    return;
+  }
+
+  /* For other cases, we use undefined, with the default trivial mapping */
+  GST_WARNING_OBJECT (enc, "Unknown mapping");
+  enc->channel_mapping_family = 255;
+
+#undef MAPS
+}
+
 static gboolean
 gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
 {
@@ -430,6 +507,7 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
 
   enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
   enc->sample_rate = GST_AUDIO_INFO_RATE (info);
+  gst_opus_enc_setup_channel_mapping (enc, info);
   GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
       enc->sample_rate);
 
@@ -455,17 +533,12 @@ static gboolean
 gst_opus_enc_setup (GstOpusEnc * enc)
 {
   int error = OPUS_OK;
-  unsigned char mapping[256];
-  int n;
 
   GST_DEBUG_OBJECT (enc, "setup");
 
-  for (n = 0; n < enc->n_channels; ++n)
-    mapping[n] = n;
-
   enc->state =
       opus_multistream_encoder_create (enc->sample_rate, enc->n_channels,
-      (enc->n_channels + 1) / 2, enc->n_channels / 2, mapping,
+      (enc->n_channels + 1) / 2, enc->n_channels / 2, enc->channel_mapping,
       enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP,
       &error);
   if (!enc->state || error != OPUS_OK)
@@ -557,18 +630,19 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
     GstBuffer *outbuf;
 
     ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
-        GST_BUFFER_OFFSET_NONE, enc->max_payload_size,
+        GST_BUFFER_OFFSET_NONE, enc->max_payload_size * enc->n_channels,
         GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
 
     if (GST_FLOW_OK != ret)
       goto done;
 
     GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
-        enc->frame_samples);
+        enc->frame_samples, (int) bytes);
 
     outsize =
         opus_multistream_encode (enc->state, (const gint16 *) data,
-        enc->frame_samples, GST_BUFFER_DATA (outbuf), enc->max_payload_size);
+        enc->frame_samples, GST_BUFFER_DATA (outbuf),
+        enc->max_payload_size * enc->n_channels);
 
     if (outsize < 0) {
       GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
@@ -582,6 +656,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
       goto done;
     }
 
+    GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
     GST_BUFFER_SIZE (outbuf) = outsize;
 
     ret =
@@ -621,7 +696,8 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
     enc->headers = NULL;
 
     gst_opus_header_create_caps (&caps, &enc->headers, enc->n_channels,
-        enc->sample_rate, gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
+        enc->sample_rate, enc->channel_mapping_family, enc->channel_mapping,
+        gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
 
 
     /* negotiate with these caps */
diff --git a/ext/opus/gstopusenc.h b/ext/opus/gstopusenc.h
index 8304e8255..8c2c3c6e8 100644
--- a/ext/opus/gstopusenc.h
+++ b/ext/opus/gstopusenc.h
@@ -77,6 +77,9 @@ struct _GstOpusEnc {
   GSList                *headers;
 
   GstTagList            *tags;
+
+  guint8                channel_mapping_family;
+  guint8                channel_mapping[256];
 };
 
 struct _GstOpusEncClass {
diff --git a/ext/opus/gstopusheader.c b/ext/opus/gstopusheader.c
index 355105584..0379c909f 100644
--- a/ext/opus/gstopusheader.c
+++ b/ext/opus/gstopusheader.c
@@ -27,7 +27,8 @@
 #include "gstopusheader.h"
 
 static GstBuffer *
-gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate)
+gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate,
+    guint8 channel_mapping_family, const guint8 * channel_mapping)
 {
   GstBuffer *buffer;
   GstByteWriter bw;
@@ -41,7 +42,12 @@ gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate)
   gst_byte_writer_put_uint16_le (&bw, 0);       /* pre-skip *//* TODO: endianness ? */
   gst_byte_writer_put_uint32_le (&bw, sample_rate);
   gst_byte_writer_put_uint16_le (&bw, 0);       /* output gain *//* TODO: endianness ? */
-  gst_byte_writer_put_uint8 (&bw, 0);   /* channel mapping *//* TODO: what is this ? */
+  gst_byte_writer_put_uint8 (&bw, channel_mapping_family);
+  if (channel_mapping_family > 0) {
+    gst_byte_writer_put_uint8 (&bw, (nchannels + 1) / 2);
+    gst_byte_writer_put_uint8 (&bw, nchannels / 2);
+    gst_byte_writer_put_data (&bw, channel_mapping, nchannels);
+  }
 
   buffer = gst_byte_writer_reset_and_get_buffer (&bw);
 
@@ -136,8 +142,24 @@ _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
 }
 
 void
+gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
+    GstBuffer * buf1, GstBuffer * buf2)
+{
+  g_return_if_fail (caps);
+  g_return_if_fail (headers && !*headers);
+
+  /* mark and put on caps */
+  *caps = gst_caps_from_string ("audio/x-opus");
+  *caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
+
+  *headers = g_slist_prepend (*headers, buf2);
+  *headers = g_slist_prepend (*headers, buf1);
+}
+
+void
 gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
-    gint sample_rate, const GstTagList * tags)
+    gint sample_rate, guint8 channel_mapping_family,
+    const guint8 * channel_mapping, const GstTagList * tags)
 {
   GstBuffer *buf1, *buf2;
 
@@ -145,21 +167,19 @@ gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
   g_return_if_fail (headers && !*headers);
   g_return_if_fail (nchannels > 0);
   g_return_if_fail (sample_rate >= 0);  /* 0 -> unset */
+  g_return_if_fail (channel_mapping_family == 0 || channel_mapping);
 
   /* Opus streams in Ogg begin with two headers; the initial header (with
      most of the codec setup parameters) which is mandated by the Ogg
      bitstream spec.  The second header holds any comment fields. */
 
   /* create header buffers */
-  buf1 = gst_opus_enc_create_id_buffer (nchannels, sample_rate);
+  buf1 =
+      gst_opus_enc_create_id_buffer (nchannels, sample_rate,
+      channel_mapping_family, channel_mapping);
   buf2 = gst_opus_enc_create_metadata_buffer (tags);
 
-  /* mark and put on caps */
-  *caps = gst_caps_from_string ("audio/x-opus");
-  *caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
-
-  *headers = g_slist_prepend (*headers, buf2);
-  *headers = g_slist_prepend (*headers, buf1);
+  gst_opus_header_create_caps_from_headers (caps, headers, buf1, buf2);
 }
 
 gboolean
@@ -168,3 +188,43 @@ gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size)
   return (GST_BUFFER_SIZE (buf) >= magic_size
       && !memcmp (magic, GST_BUFFER_DATA (buf), magic_size));
 }
+
+gboolean
+gst_opus_header_is_id_header (GstBuffer * buf)
+{
+  gsize size = GST_BUFFER_SIZE (buf);
+  const guint8 *data = GST_BUFFER_DATA (buf);
+  guint8 channels, channel_mapping_family, n_streams, n_stereo_streams;
+
+  if (size < 19)
+    return FALSE;
+  if (!gst_opus_header_is_header (buf, "OpusHead", 8))
+    return FALSE;
+  channels = data[9];
+  if (channels == 0)
+    return FALSE;
+  channel_mapping_family = data[18];
+  if (channel_mapping_family == 0) {
+    if (channels > 2)
+      return FALSE;
+  } else {
+    channels = data[9];
+    if (size < 21 + channels)
+      return FALSE;
+    n_streams = data[19];
+    n_stereo_streams = data[20];
+    if (n_streams == 0)
+      return FALSE;
+    if (n_stereo_streams > n_streams)
+      return FALSE;
+    if (n_streams + n_stereo_streams > 255)
+      return FALSE;
+  }
+  return TRUE;
+}
+
+gboolean
+gst_opus_header_is_comment_header (GstBuffer * buf)
+{
+  return gst_opus_header_is_header (buf, "OpusTags", 8);
+}
diff --git a/ext/opus/gstopusheader.h b/ext/opus/gstopusheader.h
index 4594264cc..3b2cfc265 100644
--- a/ext/opus/gstopusheader.h
+++ b/ext/opus/gstopusheader.h
@@ -25,8 +25,16 @@
 
 G_BEGIN_DECLS
 
-extern void gst_opus_header_create_caps (GstCaps **caps, GSList **headers, gint nchannels, gint sample_rate, const GstTagList *tags);
-extern gboolean gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size);
+extern void gst_opus_header_create_caps_from_headers (GstCaps **caps, GSList **headers,
+    GstBuffer *id_header, GstBuffer *comment_header);
+extern void gst_opus_header_create_caps (GstCaps **caps, GSList **headers,
+    gint nchannels, gint sample_rate,
+    guint8 channel_mapping_family, const guint8 *channel_mapping,
+    const GstTagList *tags);
+extern gboolean gst_opus_header_is_header (GstBuffer * buf,
+    const char *magic, guint magic_size);
+extern gboolean gst_opus_header_is_id_header (GstBuffer * buf);
+extern gboolean gst_opus_header_is_comment_header (GstBuffer * buf);
 
 
 G_END_DECLS
diff --git a/ext/opus/gstopusparse.c b/ext/opus/gstopusparse.c
index 278cead43..8627f9302 100644
--- a/ext/opus/gstopusparse.c
+++ b/ext/opus/gstopusparse.c
@@ -37,6 +37,7 @@
 #  include "config.h"
 #endif
 
+#include <string.h>
 #include <opus/opus.h>
 #include "gstopusheader.h"
 #include "gstopusparse.h"
@@ -136,7 +137,6 @@ gst_opus_parse_check_valid_frame (GstBaseParse * base,
   gsize size;
   guint32 packet_size;
   int ret = FALSE;
-  int channels;
   const unsigned char *frames[48];
   unsigned char toc;
   short frame_sizes[48];
@@ -152,8 +152,8 @@ gst_opus_parse_check_valid_frame (GstBaseParse * base,
   GST_DEBUG_OBJECT (parse, "Checking for frame, %u bytes in buffer", size);
 
   /* check for headers */
-  is_idheader = gst_opus_header_is_header (frame->buffer, "OpusHead", 8);
-  is_commentheader = gst_opus_header_is_header (frame->buffer, "OpusTags", 8);
+  is_idheader = gst_opus_header_is_id_header (frame->buffer);
+  is_commentheader = gst_opus_header_is_comment_header (frame->buffer);
   is_header = is_idheader || is_commentheader;
 
   if (!is_header) {
@@ -193,27 +193,6 @@ gst_opus_parse_check_valid_frame (GstBaseParse * base,
     data += packet_offset;
   }
 
-  if (!parse->header_sent) {
-    GstCaps *caps;
-
-    /* Opus streams can decode to 1 or 2 channels, so use the header
-       value if we have one, or 2 otherwise */
-    if (is_idheader) {
-      channels = data[9];
-    } else {
-      channels = 2;
-    }
-
-    g_slist_foreach (parse->headers, (GFunc) gst_buffer_unref, NULL);
-    parse->headers = NULL;
-
-    gst_opus_header_create_caps (&caps, &parse->headers, channels, 0, NULL);
-
-    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
-
-    parse->header_sent = TRUE;
-  }
-
   if (is_header) {
     *skip = 0;
     *frame_size = size;
@@ -291,16 +270,56 @@ gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
 {
   guint64 duration;
   GstOpusParse *parse;
+  gboolean is_idheader, is_commentheader;
 
   parse = GST_OPUS_PARSE (base);
 
-  if (gst_opus_header_is_header (frame->buffer, "OpusHead", 8)
-      || gst_opus_header_is_header (frame->buffer, "OpusTags", 8)) {
-    GST_BUFFER_TIMESTAMP (frame->buffer) = 0;
-    GST_BUFFER_DURATION (frame->buffer) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_OFFSET_END (frame->buffer) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_OFFSET (frame->buffer) = GST_CLOCK_TIME_NONE;
-    return GST_FLOW_OK;
+  is_idheader = gst_opus_header_is_id_header (frame->buffer);
+  is_commentheader = gst_opus_header_is_comment_header (frame->buffer);
+
+  if (!parse->header_sent) {
+    GstCaps *caps;
+    guint8 channels, channel_mapping_family, channel_mapping[256];
+    const guint8 *data = GST_BUFFER_DATA (frame->buffer);
+
+    /* Opus streams can decode to 1 or 2 channels, so use the header
+       value if we have one, or 2 otherwise */
+    if (is_idheader) {
+      channels = data[9];
+      channel_mapping_family = data[18];
+      /* header probing will already have done the size check */
+      memcpy (channel_mapping, GST_BUFFER_DATA (frame->buffer) + 21, channels);
+      gst_buffer_replace (&parse->id_header, frame->buffer);
+      GST_DEBUG_OBJECT (parse, "Found ID header, keeping");
+      return GST_BASE_PARSE_FLOW_DROPPED;
+    } else if (is_commentheader) {
+      gst_buffer_replace (&parse->comment_header, frame->buffer);
+      GST_DEBUG_OBJECT (parse, "Found comment header, keeping");
+      return GST_BASE_PARSE_FLOW_DROPPED;
+    }
+
+    g_slist_foreach (parse->headers, (GFunc) gst_buffer_unref, NULL);
+    parse->headers = NULL;
+
+    if (parse->id_header && parse->comment_header) {
+      gst_opus_header_create_caps_from_headers (&caps, &parse->headers,
+          parse->id_header, parse->comment_header);
+    } else {
+      GST_INFO_OBJECT (parse,
+          "No headers, blindly setting up canonical stereo");
+      channels = 2;
+      channel_mapping_family = 0;
+      channel_mapping[0] = 0;
+      channel_mapping[1] = 1;
+      gst_opus_header_create_caps (&caps, &parse->headers, channels, 0,
+          channel_mapping_family, channel_mapping, NULL);
+    }
+
+    gst_buffer_replace (&parse->id_header, NULL);
+    gst_buffer_replace (&parse->comment_header, NULL);
+
+    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
+    parse->header_sent = TRUE;
   }
 
   GST_BUFFER_TIMESTAMP (frame->buffer) = parse->next_ts;
diff --git a/ext/opus/gstopusparse.h b/ext/opus/gstopusparse.h
index 60ea5c536..f0fd24194 100644
--- a/ext/opus/gstopusparse.h
+++ b/ext/opus/gstopusparse.h
@@ -46,6 +46,8 @@ struct _GstOpusParse {
   gboolean header_sent;
   GSList *headers;
   GstClockTime next_ts;
+  GstBuffer *id_header;
+  GstBuffer *comment_header;
 };
 
 struct _GstOpusParseClass {