opus: Add proper support for multichannel audio

https://bugzilla.gnome.org/show_bug.cgi?id=757152

ext/opus/Makefile.am

@@ -10,6 +10,7 @@ libgstopus_la_CFLAGS = \
 libgstopus_la_LIBADD = \
 	$(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \
 	-lgsttag-$(GST_API_VERSION) -lgstrtp-$(GST_API_VERSION) \
+	-lgstpbutils-$(GST_API_VERSION) \
 	$(GST_BASE_LIBS) \
 	$(GST_LIBS) \
 	$(OPUS_LIBS)

ext/opus/gstopusdec.c

@@ -47,6 +47,7 @@
 #include "gstopusheader.h"
 #include "gstopuscommon.h"
 #include "gstopusdec.h"
+#include <gst/pbutils/pbutils.h>
 
 GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
 #define GST_CAT_DEFAULT opusdec_debug
@@ -225,6 +226,8 @@ gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
   GstAudioInfo info;
 
   if (caps) {
+    gint rate, channels;
+
     caps = gst_caps_truncate (caps);
     caps = gst_caps_make_writable (caps);
     s = gst_caps_get_structure (caps, 0);
@@ -233,13 +236,15 @@ gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
       gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);
     else
       gst_structure_set (s, "rate", G_TYPE_INT, dec->sample_rate, NULL);
-    gst_structure_get_int (s, "rate", &dec->sample_rate);
+    gst_structure_get_int (s, "rate", &rate);
+    dec->sample_rate = rate;
 
     if (gst_structure_has_field (s, "channels"))
       gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
     else
       gst_structure_set (s, "channels", G_TYPE_INT, dec->n_channels, NULL);
-    gst_structure_get_int (s, "channels", &dec->n_channels);
+    gst_structure_get_int (s, "channels", &channels);
+    dec->n_channels = channels;
 
     gst_caps_unref (caps);
   }
@@ -273,7 +278,6 @@ gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
   /* but we still need the opus order for later reordering */
   if (pos) {
     memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
-    gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
   } else {
     dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
   }
@@ -284,79 +288,64 @@ gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
 static GstFlowReturn
 gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
 {
-  const guint8 *data;
   GstAudioChannelPosition pos[64];
   const GstAudioChannelPosition *posn = NULL;
-  GstMapInfo map;
 
   if (!gst_opus_header_is_id_header (buf)) {
     GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
     return GST_FLOW_ERROR;
   }
 
-  gst_buffer_map (buf, &map, GST_MAP_READ);
+  if (!gst_codec_utils_opus_parse_header (buf,
-  data = map.data;
+          &dec->sample_rate,
-
+          &dec->n_channels,
-  dec->n_channels = data[9];
+          &dec->channel_mapping_family,
-  dec->sample_rate = GST_READ_UINT32_LE (data + 12);
+          &dec->n_streams,
-  if (dec->sample_rate == 0)
+          &dec->n_stereo_streams,
-    dec->sample_rate = 48000;
+          dec->channel_mapping, &dec->pre_skip, &dec->r128_gain)) {
-  dec->pre_skip = GST_READ_UINT16_LE (data + 10);
+    GST_ERROR_OBJECT (dec, "Failed to parse Opus ID header");
-  dec->r128_gain = GST_READ_UINT16_LE (data + 16);
+    return GST_FLOW_ERROR;
+  }
   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 = data[18];
+  if (dec->channel_mapping_family == 1) {
-  if (dec->channel_mapping_family == 0) {
+    GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
-    /* implicit mapping */
+    switch (dec->n_channels) {
-    GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
+      case 1:
-    dec->n_streams = dec->n_stereo_streams = 1;
+      case 2:
-    dec->channel_mapping[0] = 0;
+        /* nothing */
-    dec->channel_mapping[1] = 1;
+        break;
-  } else {
+      case 3:
-    dec->n_streams = data[19];
+      case 4:
-    dec->n_stereo_streams = data[20];
+      case 5:
-    memcpy (dec->channel_mapping, data + 21, dec->n_channels);
+      case 6:
+      case 7:
+      case 8:
+        posn = gst_opus_channel_positions[dec->n_channels - 1];
+        break;
+      default:{
+        gint i;
 
-    if (dec->channel_mapping_family == 1) {
+        GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
-      GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
+            (NULL), ("Using NONE channel layout for more than 8 channels"));
-      switch (dec->n_channels) {
-        case 1:
-        case 2:
-          /* nothing */
-          break;
-        case 3:
-        case 4:
-        case 5:
-        case 6:
-        case 7:
-        case 8:
-          posn = gst_opus_channel_positions[dec->n_channels - 1];
-          break;
-        default:{
-          gint i;
 
-          GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
+        for (i = 0; i < dec->n_channels; i++)
-              (NULL), ("Using NONE channel layout for more than 8 channels"));
+          pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
 
-          for (i = 0; i < dec->n_channels; i++)
+        posn = pos;
-            pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
-
-          posn = pos;
-        }
       }
-    } else {
-      GST_INFO_OBJECT (dec, "Channel mapping family %d",
-          dec->channel_mapping_family);
     }
+  } else {
+    GST_INFO_OBJECT (dec, "Channel mapping family %d",
+        dec->channel_mapping_family);
   }
 
   gst_opus_dec_negotiate (dec, posn);
 
-  gst_buffer_unmap (buf, &map);
-
   return GST_FLOW_OK;
 }
 
@@ -660,8 +649,10 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
     if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
       buf = gst_value_get_buffer (header);
       res = gst_opus_dec_parse_header (dec, buf);
-      if (res != GST_FLOW_OK)
+      if (res != GST_FLOW_OK) {
+        ret = FALSE;
         goto done;
+      }
       gst_buffer_replace (&dec->streamheader, buf);
     }
 
@@ -669,26 +660,26 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
     if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
       buf = gst_value_get_buffer (vorbiscomment);
       res = gst_opus_dec_parse_comments (dec, buf);
-      if (res != GST_FLOW_OK)
+      if (res != GST_FLOW_OK) {
+        ret = FALSE;
         goto done;
+      }
       gst_buffer_replace (&dec->vorbiscomment, buf);
     }
   } else {
-    /* defaults if not in the caps */
+    const GstAudioChannelPosition *posn = NULL;
-    dec->n_channels = 2;
-    dec->sample_rate = 48000;
 
-    gst_structure_get_int (s, "channels", &dec->n_channels);
+    if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,
-    gst_structure_get_int (s, "rate", &dec->sample_rate);
+            &dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,
+            &dec->n_stereo_streams, dec->channel_mapping)) {
+      ret = FALSE;
+      goto done;
+    }
 
-    /* default stereo mapping */
+    if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)
-    dec->channel_mapping_family = 0;
+      posn = gst_opus_channel_positions[dec->n_channels - 1];
-    dec->channel_mapping[0] = 0;
-    dec->channel_mapping[1] = 1;
-    dec->n_streams = 1;
-    dec->n_stereo_streams = 1;
 
-    gst_opus_dec_negotiate (dec, NULL);
+    gst_opus_dec_negotiate (dec, posn);
   }
 
 done:

ext/opus/gstopusdec.h

@@ -52,9 +52,9 @@ struct _GstOpusDec {
   GstBuffer            *streamheader;
   GstBuffer            *vorbiscomment;
 
-  int sample_rate;
+  guint32 sample_rate;
-  int n_channels;
+  guint8 n_channels;
-  guint32 pre_skip;
+  guint16 pre_skip;
   gint16 r128_gain;
 
   GstAudioChannelPosition opus_pos[64];

ext/opus/gstopusenc.c

@@ -48,6 +48,8 @@
 
 #include <gst/gsttagsetter.h>
 #include <gst/audio/audio.h>
+#include <gst/pbutils/pbutils.h>
+#include <gst/tag/tag.h>
 #include <gst/glib-compat-private.h>
 #include "gstopusheader.h"
 #include "gstopuscommon.h"
@@ -167,12 +169,12 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
         "format = (string) " FORMAT_STR ", "
         "layout = (string) interleaved, "
         "rate = (int) 48000, "
-        "channels = (int) [ 1, 2 ]; "
+        "channels = (int) [ 1, 8 ]; "
         "audio/x-raw, "
         "format = (string) " FORMAT_STR ", "
         "layout = (string) interleaved, "
         "rate = (int) { 8000, 12000, 16000, 24000 }, "
-        "channels = (int) [ 1, 2 ] ")
+        "channels = (int) [ 1, 8 ] ")
     );
 
 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
@@ -545,27 +547,20 @@ gst_opus_enc_setup_channel_mappings (GstOpusEnc * enc,
   /* 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, trivial RTP mapping");
-      GST_INFO_OBJECT (enc, "Stereo, canonical mapping");
+    enc->channel_mapping_family = 0;
-      enc->channel_mapping_family = 0;
+    enc->n_stereo_streams = 1;
-      enc->n_stereo_streams = 1;
+    /* implicit mapping for family 0 */
-      /* The channel mapping is implicit for family 0, that's why we do not
+    return;
-         attempt to create one for right/left - this will be mapped to the
-         Vorbis mapping below. */
-      return;
-    } 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
+  /* For channels between 3 and 8, we use the Vorbis mapping if we can
-     find a permutation that matches it. Mono will have been taken care
+     find a permutation that matches it. Mono and stereo will have been taken
-     of earlier, but this code also handles it. Same for left/right stereo.
+     care of earlier, but this code also handles it. There are two mappings.
-     There are two mappings. One maps the input channels to an ordering
+     One maps the input channels to an ordering which has the natural pairs
-     which has the natural pairs first so they can benefit from the Opus
+     first so they can benefit from the Opus stereo channel coupling, and the
-     stereo channel coupling, and the other maps this ordering to the
+     other maps this ordering to the Vorbis ordering. */
-     Vorbis ordering. */
+  if (enc->n_channels >= 3 && enc->n_channels <= 8) {
-  if (enc->n_channels >= 1 && enc->n_channels <= 8) {
     int c0, c1, c0v, c1v;
     int mapped;
     gboolean positions_done[256];
@@ -580,6 +575,8 @@ gst_opus_enc_setup_channel_mappings (GstOpusEnc * enc,
           GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
       {GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
           GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
+      {GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+          GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
     };
     size_t pair;
 
@@ -633,13 +630,8 @@ gst_opus_enc_setup_channel_mappings (GstOpusEnc * enc,
         GST_DEBUG_OBJECT (enc, "Channel position %s is not mapped yet, adding",
             gst_opus_channel_names[position]);
         cv = gst_opus_enc_find_channel_position_in_vorbis_order (enc, position);
-        if (cv < 0) {
+        if (cv < 0)
-          GST_WARNING_OBJECT (enc,
+          g_assert_not_reached ();
-              "Cannot map channel positions to Vorbis order, using unknown mapping");
-          enc->channel_mapping_family = 255;
-          enc->n_stereo_streams = 0;
-          return;
-        }
         enc->encoding_channel_mapping[mapped] = n;
         enc->decoding_channel_mapping[cv] = mapped;
         mapped++;
@@ -718,6 +710,8 @@ gst_opus_enc_setup (GstOpusEnc * enc)
   GstCaps *caps;
   gboolean ret;
   gint32 lookahead;
+  const GstTagList *tags;
+  GstBuffer *header, *comments;
 
 #ifndef GST_DISABLE_GST_DEBUG
   GST_DEBUG_OBJECT (enc,
@@ -764,10 +758,17 @@ gst_opus_enc_setup (GstOpusEnc * enc)
   lookahead = lookahead * 48000 / enc->sample_rate;
   enc->lookahead = enc->pending_lookahead = lookahead;
 
-  gst_opus_header_create_caps (&caps, NULL, lookahead, enc->sample_rate,
+  header = gst_codec_utils_opus_create_header (enc->sample_rate,
-      enc->n_channels, enc->n_stereo_streams, enc->channel_mapping_family,
+      enc->n_channels, enc->channel_mapping_family,
-      enc->decoding_channel_mapping,
+      enc->n_channels - enc->n_stereo_streams, enc->n_stereo_streams,
-      gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
+      enc->decoding_channel_mapping, lookahead, 0);
+  tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
+  comments =
+      gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags",
+      8, "Encoded with GStreamer opusenc");
+  caps = gst_codec_utils_opus_create_caps_from_header (header, comments);
+  gst_buffer_unref (header);
+  gst_buffer_unref (comments);
 
   /* negotiate with these caps */
   GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
@@ -813,71 +814,96 @@ gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
   return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
 }
 
+static GstCaps *
+gst_opus_enc_get_sink_template_caps (void)
+{
+  static volatile gsize init = 0;
+  static GstCaps *caps = NULL;
+
+  if (g_once_init_enter (&init)) {
+    GValue rate_array = G_VALUE_INIT;
+    GValue v = G_VALUE_INIT;
+    GstStructure *s1, *s2, *s;
+    gint i, c;
+
+    caps = gst_caps_new_empty ();
+
+    /* Generate our two template structures */
+    g_value_init (&rate_array, GST_TYPE_LIST);
+    g_value_init (&v, G_TYPE_INT);
+    g_value_set_int (&v, 8000);
+    gst_value_list_append_value (&rate_array, &v);
+    g_value_set_int (&v, 12000);
+    gst_value_list_append_value (&rate_array, &v);
+    g_value_set_int (&v, 16000);
+    gst_value_list_append_value (&rate_array, &v);
+    g_value_set_int (&v, 24000);
+    gst_value_list_append_value (&rate_array, &v);
+
+    s1 = gst_structure_new ("audio/x-raw",
+        "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+        "layout", G_TYPE_STRING, "interleaved",
+        "rate", G_TYPE_INT, 48000, NULL);
+    s2 = gst_structure_new ("audio/x-raw",
+        "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+        "layout", G_TYPE_STRING, "interleaved", NULL);
+    gst_structure_set_value (s2, "rate", &rate_array);
+    g_value_unset (&rate_array);
+    g_value_unset (&v);
+
+    /* Mono */
+    s = gst_structure_copy (s1);
+    gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
+    gst_caps_append_structure (caps, s);
+
+    s = gst_structure_copy (s2);
+    gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
+    gst_caps_append_structure (caps, s);
+
+    /* Stereo and further */
+    for (i = 2; i <= 8; i++) {
+      guint64 channel_mask = 0;
+      const GstAudioChannelPosition *pos = gst_opus_channel_positions[i - 1];
+
+      for (c = 0; c < i; c++) {
+        channel_mask |= G_GUINT64_CONSTANT (1) << pos[c];
+      }
+
+      s = gst_structure_copy (s1);
+      gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
+          GST_TYPE_BITMASK, channel_mask, NULL);
+      gst_caps_append_structure (caps, s);
+
+      s = gst_structure_copy (s2);
+      gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
+          GST_TYPE_BITMASK, channel_mask, NULL);
+      gst_caps_append_structure (caps, s);
+    }
+
+    gst_structure_free (s1);
+    gst_structure_free (s2);
+
+    g_once_init_leave (&init, 1);
+  }
+
+  return caps;
+}
+
 static GstCaps *
 gst_opus_enc_sink_getcaps (GstAudioEncoder * benc, GstCaps * filter)
 {
   GstOpusEnc *enc;
   GstCaps *caps;
-  GstCaps *tcaps;
-  GstCaps *peercaps = NULL;
-  GstCaps *intersect = NULL;
-  guint i;
-  gboolean allow_multistream;
 
   enc = GST_OPUS_ENC (benc);
 
   GST_DEBUG_OBJECT (enc, "sink getcaps");
 
-  peercaps = gst_pad_peer_query_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), NULL);
+  caps = gst_opus_enc_get_sink_template_caps ();
-  if (!peercaps) {
+  caps = gst_audio_encoder_proxy_getcaps (benc, caps, filter);
-    GST_DEBUG_OBJECT (benc, "No peercaps, returning template sink caps");
-    return gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SINK_PAD (benc));
-  }
-
-  tcaps = gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (benc));
-  intersect = gst_caps_intersect (peercaps, tcaps);
-  gst_caps_unref (tcaps);
-  gst_caps_unref (peercaps);
-
-  if (gst_caps_is_empty (intersect))
-    return intersect;
-
-  allow_multistream = FALSE;
-  for (i = 0; i < gst_caps_get_size (intersect); i++) {
-    GstStructure *s = gst_caps_get_structure (intersect, i);
-    gboolean multistream;
-    if (gst_structure_get_boolean (s, "multistream", &multistream)) {
-      if (multistream) {
-        allow_multistream = TRUE;
-      }
-    } else {
-      allow_multistream = TRUE;
-    }
-  }
-
-  gst_caps_unref (intersect);
-
-  caps = gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SINK_PAD (benc));
-  caps = gst_caps_make_writable (caps);
-  if (!allow_multistream) {
-    GValue range = { 0 };
-    g_value_init (&range, GST_TYPE_INT_RANGE);
-    gst_value_set_int_range (&range, 1, 2);
-    for (i = 0; i < gst_caps_get_size (caps); i++) {
-      GstStructure *s = gst_caps_get_structure (caps, i);
-      gst_structure_set_value (s, "channels", &range);
-    }
-    g_value_unset (&range);
-  }
-
-  if (filter) {
-    GstCaps *tmp = gst_caps_intersect_full (caps, filter,
-        GST_CAPS_INTERSECT_FIRST);
-    gst_caps_unref (caps);
-    caps = tmp;
-  }
 
   GST_DEBUG_OBJECT (enc, "Returning caps: %" GST_PTR_FORMAT, caps);
+
   return caps;
 }
 

ext/opus/gstopusheader.c

@@ -26,218 +26,6 @@
 #include <gst/base/gstbytewriter.h>
 #include "gstopusheader.h"
 
-static GstBuffer *
-gst_opus_enc_create_id_buffer (guint16 pre_skip, guint sample_rate,
-    guint8 nchannels, guint8 n_stereo_streams, guint8 channel_mapping_family,
-    const guint8 * channel_mapping)
-{
-  GstBuffer *buffer;
-  GstByteWriter bw;
-  gboolean hdl = TRUE;
-
-  g_return_val_if_fail (nchannels > 0, NULL);
-  g_return_val_if_fail (n_stereo_streams <= nchannels - n_stereo_streams, NULL);
-
-  gst_byte_writer_init (&bw);
-
-  /* See http://wiki.xiph.org/OggOpus */
-  hdl &= gst_byte_writer_put_data (&bw, (const guint8 *) "OpusHead", 8);
-  hdl &= gst_byte_writer_put_uint8 (&bw, 0x01); /* version number */
-  hdl &= gst_byte_writer_put_uint8 (&bw, nchannels);
-  hdl &= gst_byte_writer_put_uint16_le (&bw, pre_skip);
-  hdl &= gst_byte_writer_put_uint32_le (&bw, sample_rate);
-  hdl &= gst_byte_writer_put_uint16_le (&bw, 0);        /* output gain */
-  hdl &= gst_byte_writer_put_uint8 (&bw, channel_mapping_family);
-  if (channel_mapping_family > 0) {
-    hdl &= gst_byte_writer_put_uint8 (&bw, nchannels - n_stereo_streams);
-    hdl &= gst_byte_writer_put_uint8 (&bw, n_stereo_streams);
-    hdl &= gst_byte_writer_put_data (&bw, channel_mapping, nchannels);
-  }
-
-  if (!hdl)
-    GST_WARNING ("Error creating header");
-
-  buffer = gst_byte_writer_reset_and_get_buffer (&bw);
-
-  GST_BUFFER_OFFSET (buffer) = 0;
-  GST_BUFFER_OFFSET_END (buffer) = 0;
-
-  return buffer;
-}
-
-static GstBuffer *
-gst_opus_enc_create_metadata_buffer (const GstTagList * tags)
-{
-  GstTagList *empty_tags = NULL;
-  GstBuffer *comments = NULL;
-
-  GST_DEBUG ("tags = %" GST_PTR_FORMAT, tags);
-
-  if (tags == NULL) {
-    /* FIXME: better fix chain of callers to not write metadata at all,
-     * if there is none */
-    empty_tags = gst_tag_list_new_empty ();
-    tags = empty_tags;
-  }
-  comments =
-      gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags",
-      8, "Encoded with GStreamer Opusenc");
-
-  GST_BUFFER_OFFSET (comments) = 0;
-  GST_BUFFER_OFFSET_END (comments) = 0;
-
-  if (empty_tags)
-    gst_tag_list_unref (empty_tags);
-
-  return comments;
-}
-
-/*
- * (really really) FIXME: move into core (dixit tpm)
- */
-/*
- * _gst_caps_set_buffer_array:
- * @caps: (transfer full): a #GstCaps
- * @field: field in caps to set
- * @buf: header buffers
- *
- * Adds given buffers to an array of buffers set as the given @field
- * on the given @caps.  List of buffer arguments must be NULL-terminated.
- *
- * Returns: (transfer full): input caps with a streamheader field added, or NULL
- *     if some error occurred
- */
-static GstCaps *
-_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
-    GstBuffer * buf, ...)
-{
-  GstStructure *structure = NULL;
-  va_list va;
-  GValue array = { 0 };
-  GValue value = { 0 };
-
-  g_return_val_if_fail (caps != NULL, NULL);
-  g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
-  g_return_val_if_fail (field != NULL, NULL);
-
-  caps = gst_caps_make_writable (caps);
-  structure = gst_caps_get_structure (caps, 0);
-
-  g_value_init (&array, GST_TYPE_ARRAY);
-
-  va_start (va, buf);
-  /* put buffers in a fixed list */
-  while (buf) {
-    g_assert (gst_buffer_is_writable (buf));
-
-    /* mark buffer */
-    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
-
-    g_value_init (&value, GST_TYPE_BUFFER);
-    buf = gst_buffer_copy (buf);
-    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
-    gst_value_set_buffer (&value, buf);
-    gst_buffer_unref (buf);
-    gst_value_array_append_value (&array, &value);
-    g_value_unset (&value);
-
-    buf = va_arg (va, GstBuffer *);
-  }
-  va_end (va);
-
-  gst_structure_set_value (structure, field, &array);
-  g_value_unset (&array);
-
-  return caps;
-}
-
-void
-gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
-    GstBuffer * buf1, GstBuffer * buf2)
-{
-  int n_streams, family;
-  gint channels, rate;
-  gboolean multistream;
-  GstMapInfo map;
-  guint8 *data;
-
-  g_return_if_fail (caps);
-  g_return_if_fail (!headers || !*headers);
-  g_return_if_fail (gst_buffer_get_size (buf1) >= 19);
-
-  gst_buffer_map (buf1, &map, GST_MAP_READ);
-  data = map.data;
-
-  channels = data[9];
-  rate = GST_READ_UINT32_LE (data + 12);
-
-  /* work out the number of streams */
-  family = data[18];
-  if (family == 0) {
-    n_streams = 1;
-  } else {
-    /* only included in the header for family > 0 */
-    if (map.size >= 20)
-      n_streams = data[19];
-    else {
-      g_warning ("family > 0 but header buffer size < 20");
-      gst_buffer_unmap (buf1, &map);
-      return;
-    }
-  }
-
-  /* TODO: should probably also put the channel mapping into the caps too once
-   * we actually support multi channel, and pre-skip and other fields */
-
-  gst_buffer_unmap (buf1, &map);
-
-  /* mark and put on caps */
-  multistream = n_streams > 1;
-  *caps = gst_caps_new_simple ("audio/x-opus",
-      "multistream", G_TYPE_BOOLEAN, multistream,
-      "channels", G_TYPE_INT, channels, NULL);
-
-  if (rate > 0) {
-    gst_caps_set_simple (*caps, "rate", G_TYPE_INT, rate, NULL);
-  }
-
-  *caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
-
-  if (headers) {
-    *headers = g_slist_prepend (*headers, gst_buffer_ref (buf2));
-    *headers = g_slist_prepend (*headers, gst_buffer_ref (buf1));
-  }
-}
-
-void
-gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers,
-    guint16 pre_skip, guint sample_rate, guint8 nchannels,
-    guint8 n_stereo_streams, guint8 channel_mapping_family,
-    const guint8 * channel_mapping, const GstTagList * tags)
-{
-  GstBuffer *buf1, *buf2;
-
-  g_return_if_fail (caps);
-  g_return_if_fail (!headers || !*headers);
-  g_return_if_fail (nchannels > 0);
-  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 (pre_skip, sample_rate, nchannels,
-      n_stereo_streams, channel_mapping_family, channel_mapping);
-  buf2 = gst_opus_enc_create_metadata_buffer (tags);
-
-  gst_opus_header_create_caps_from_headers (caps, headers, buf1, buf2);
-
-  gst_buffer_unref (buf2);
-  gst_buffer_unref (buf1);
-}
-
 gboolean
 gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size)
 {

ext/opus/gstopusheader.h

@@ -26,12 +26,6 @@
 
 G_BEGIN_DECLS
 
-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,
-    guint16 pre_skip, guint sample_rate32, guint8 nchannels, guint8 n_stereo_streams,
-    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);