opus: duplicate from CELT

Copy the celt plugin and convert it to Opus.  Mostly works.

ext/opus/Makefile.am

@@ -0,0 +1,16 @@
+plugin_LTLIBRARIES = libgstopus.la
+
+libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c
+libgstopus_la_CFLAGS = \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_CFLAGS) \
+	$(OPUS_CFLAGS)
+libgstopus_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
+	$(GST_BASE_LIBS) \
+	$(GST_LIBS) \
+	$(OPUS_LIBS)
+libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
+libgstopus_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = gstopusenc.h gstopusdec.h 

ext/opus/gstopus.c

@@ -0,0 +1,50 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2008> 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.
+ */
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstopusdec.h"
+#include "gstopusenc.h"
+
+#include <gst/tag/tag.h>
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+
+  if (!gst_element_register (plugin, "opusenc", GST_RANK_NONE,
+          GST_TYPE_OPUS_ENC))
+    return FALSE;
+
+  if (!gst_element_register (plugin, "opusdec", GST_RANK_PRIMARY,
+          GST_TYPE_OPUS_DEC))
+    return FALSE;
+
+  gst_tag_register_musicbrainz_tags ();
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "opus",
+    "OPUS plugin library",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

ext/opus/gstopusdec.c

@@ -0,0 +1,865 @@
+/* GStreamer
+ * Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
+ * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
+ * Copyright (C) 2008 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.
+ */
+
+/*
+ * Based on the speexdec element.
+ */
+
+/**
+ * SECTION:element-opusdec
+ * @see_also: opusenc, oggdemux
+ *
+ * This element decodes a OPUS stream to raw integer audio.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink
+ * ]| Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include "gstopusdec.h"
+#include <string.h>
+#include <gst/tag/tag.h>
+
+GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
+#define GST_CAT_DEFAULT opusdec_debug
+
+#define DEC_MAX_FRAME_SIZE 2000
+
+static GstStaticPadTemplate opus_dec_src_factory =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "rate = (int) [ 32000, 64000 ], "
+        "channels = (int) [ 1, 2 ], "
+        "endianness = (int) BYTE_ORDER, "
+        "signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
+    );
+
+static GstStaticPadTemplate opus_dec_sink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-opus")
+    );
+
+GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstElement, GST_TYPE_ELEMENT);
+
+static gboolean opus_dec_sink_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn opus_dec_chain (GstPad * pad, GstBuffer * buf);
+static gboolean opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps);
+static GstStateChangeReturn opus_dec_change_state (GstElement * element,
+    GstStateChange transition);
+
+static gboolean opus_dec_src_event (GstPad * pad, GstEvent * event);
+static gboolean opus_dec_src_query (GstPad * pad, GstQuery * query);
+static gboolean opus_dec_sink_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *opus_get_src_query_types (GstPad * pad);
+static const GstQueryType *opus_get_sink_query_types (GstPad * pad);
+static gboolean opus_dec_convert (GstPad * pad,
+    GstFormat src_format, gint64 src_value,
+    GstFormat * dest_format, gint64 * dest_value);
+
+static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec,
+    GstBuffer * buf, GstClockTime timestamp, GstClockTime duration);
+static GstFlowReturn opus_dec_chain_parse_header (GstOpusDec * dec,
+    GstBuffer * buf);
+#if 0
+static GstFlowReturn opus_dec_chain_parse_comments (GstOpusDec * dec,
+    GstBuffer * buf);
+#endif
+
+static void
+gst_opus_dec_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&opus_dec_src_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&opus_dec_sink_factory));
+  gst_element_class_set_details_simple (element_class, "Opus audio decoder",
+      "Codec/Decoder/Audio",
+      "decode opus streams to audio",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_opus_dec_class_init (GstOpusDecClass * klass)
+{
+  GstElementClass *gstelement_class;
+
+  gstelement_class = (GstElementClass *) klass;
+
+  gstelement_class->change_state = GST_DEBUG_FUNCPTR (opus_dec_change_state);
+
+  GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
+      "opus decoding element");
+}
+
+static void
+gst_opus_dec_reset (GstOpusDec * dec)
+{
+  gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
+  dec->granulepos = -1;
+  dec->packetno = 0;
+  dec->frame_size = 0;
+  dec->frame_samples = 960;
+  dec->frame_duration = 0;
+  if (dec->state) {
+    opus_decoder_destroy (dec->state);
+    dec->state = NULL;
+  }
+#if 0
+  if (dec->mode) {
+    opus_mode_destroy (dec->mode);
+    dec->mode = NULL;
+  }
+#endif
+
+  gst_buffer_replace (&dec->streamheader, NULL);
+  gst_buffer_replace (&dec->vorbiscomment, NULL);
+  g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (dec->extra_headers);
+  dec->extra_headers = NULL;
+
+#if 0
+  memset (&dec->header, 0, sizeof (dec->header));
+#endif
+}
+
+static void
+gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class)
+{
+  dec->sinkpad =
+      gst_pad_new_from_static_template (&opus_dec_sink_factory, "sink");
+  gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (opus_dec_chain));
+  gst_pad_set_event_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (opus_dec_sink_event));
+  gst_pad_set_query_type_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (opus_get_sink_query_types));
+  gst_pad_set_query_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (opus_dec_sink_query));
+  gst_pad_set_setcaps_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (opus_dec_sink_setcaps));
+  gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
+
+  dec->srcpad = gst_pad_new_from_static_template (&opus_dec_src_factory, "src");
+  gst_pad_use_fixed_caps (dec->srcpad);
+  gst_pad_set_event_function (dec->srcpad,
+      GST_DEBUG_FUNCPTR (opus_dec_src_event));
+  gst_pad_set_query_type_function (dec->srcpad,
+      GST_DEBUG_FUNCPTR (opus_get_src_query_types));
+  gst_pad_set_query_function (dec->srcpad,
+      GST_DEBUG_FUNCPTR (opus_dec_src_query));
+  gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
+
+  dec->sample_rate = 48000;
+  dec->n_channels = 2;
+
+  gst_opus_dec_reset (dec);
+}
+
+static gboolean
+opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+  GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+  gboolean ret = TRUE;
+  GstStructure *s;
+  const GValue *streamheader;
+
+  s = gst_caps_get_structure (caps, 0);
+  if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
+      G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
+      gst_value_array_get_size (streamheader) >= 2) {
+    const GValue *header;
+    GstBuffer *buf;
+    GstFlowReturn res = GST_FLOW_OK;
+
+    header = gst_value_array_get_value (streamheader, 0);
+    if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
+      buf = gst_value_get_buffer (header);
+      res = opus_dec_chain_parse_header (dec, buf);
+      if (res != GST_FLOW_OK)
+        goto done;
+      gst_buffer_replace (&dec->streamheader, buf);
+    }
+#if 0
+    vorbiscomment = gst_value_array_get_value (streamheader, 1);
+    if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
+      buf = gst_value_get_buffer (vorbiscomment);
+      res = opus_dec_chain_parse_comments (dec, buf);
+      if (res != GST_FLOW_OK)
+        goto done;
+      gst_buffer_replace (&dec->vorbiscomment, buf);
+    }
+#endif
+
+    g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
+    g_list_free (dec->extra_headers);
+    dec->extra_headers = NULL;
+
+    if (gst_value_array_get_size (streamheader) > 2) {
+      gint i, n;
+
+      n = gst_value_array_get_size (streamheader);
+      for (i = 2; i < n; i++) {
+        header = gst_value_array_get_value (streamheader, i);
+        buf = gst_value_get_buffer (header);
+        dec->extra_headers =
+            g_list_prepend (dec->extra_headers, gst_buffer_ref (buf));
+      }
+    }
+  }
+
+done:
+  gst_object_unref (dec);
+  return ret;
+}
+
+static gboolean
+opus_dec_convert (GstPad * pad,
+    GstFormat src_format, gint64 src_value,
+    GstFormat * dest_format, gint64 * dest_value)
+{
+  gboolean res = TRUE;
+  GstOpusDec *dec;
+  guint64 scale = 1;
+
+  dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+  if (dec->packetno < 1) {
+    res = FALSE;
+    goto cleanup;
+  }
+
+  if (src_format == *dest_format) {
+    *dest_value = src_value;
+    res = TRUE;
+    goto cleanup;
+  }
+
+  if (pad == dec->sinkpad &&
+      (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) {
+    res = FALSE;
+    goto cleanup;
+  }
+
+  switch (src_format) {
+    case GST_FORMAT_TIME:
+      switch (*dest_format) {
+        case GST_FORMAT_BYTES:
+          scale = sizeof (gint16) * dec->n_channels;
+        case GST_FORMAT_DEFAULT:
+          *dest_value =
+              gst_util_uint64_scale_int (scale * src_value,
+              dec->sample_rate, GST_SECOND);
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    case GST_FORMAT_DEFAULT:
+      switch (*dest_format) {
+        case GST_FORMAT_BYTES:
+          *dest_value = src_value * sizeof (gint16) * dec->n_channels;
+          break;
+        case GST_FORMAT_TIME:
+          *dest_value =
+              gst_util_uint64_scale_int (src_value, GST_SECOND,
+              dec->sample_rate);
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    case GST_FORMAT_BYTES:
+      switch (*dest_format) {
+        case GST_FORMAT_DEFAULT:
+          *dest_value = src_value / (sizeof (gint16) * dec->n_channels);
+          break;
+        case GST_FORMAT_TIME:
+          *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
+              dec->sample_rate * sizeof (gint16) * dec->n_channels);
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    default:
+      res = FALSE;
+      break;
+  }
+
+cleanup:
+  gst_object_unref (dec);
+  return res;
+}
+
+static const GstQueryType *
+opus_get_sink_query_types (GstPad * pad)
+{
+  static const GstQueryType opus_dec_sink_query_types[] = {
+    GST_QUERY_CONVERT,
+    0
+  };
+
+  return opus_dec_sink_query_types;
+}
+
+static gboolean
+opus_dec_sink_query (GstPad * pad, GstQuery * query)
+{
+  GstOpusDec *dec;
+  gboolean res;
+
+  dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_CONVERT:
+    {
+      GstFormat src_fmt, dest_fmt;
+      gint64 src_val, dest_val;
+
+      gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+      res = opus_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val);
+      if (res) {
+        gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+      }
+      break;
+    }
+    default:
+      res = gst_pad_query_default (pad, query);
+      break;
+  }
+
+  gst_object_unref (dec);
+  return res;
+}
+
+static const GstQueryType *
+opus_get_src_query_types (GstPad * pad)
+{
+  static const GstQueryType opus_dec_src_query_types[] = {
+    GST_QUERY_POSITION,
+    GST_QUERY_DURATION,
+    0
+  };
+
+  return opus_dec_src_query_types;
+}
+
+static gboolean
+opus_dec_src_query (GstPad * pad, GstQuery * query)
+{
+  GstOpusDec *dec;
+  gboolean res = FALSE;
+
+  dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_POSITION:{
+      GstSegment segment;
+      GstFormat format;
+      gint64 cur;
+
+      gst_query_parse_position (query, &format, NULL);
+
+      GST_PAD_STREAM_LOCK (dec->sinkpad);
+      segment = dec->segment;
+      GST_PAD_STREAM_UNLOCK (dec->sinkpad);
+
+      if (segment.format != GST_FORMAT_TIME) {
+        GST_DEBUG_OBJECT (dec, "segment not initialised yet");
+        break;
+      }
+
+      if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
+                  segment.last_stop, &format, &cur))) {
+        gst_query_set_position (query, format, cur);
+      }
+      break;
+    }
+    case GST_QUERY_DURATION:{
+      GstFormat format = GST_FORMAT_TIME;
+      gint64 dur;
+
+      /* get duration from demuxer */
+      if (!gst_pad_query_peer_duration (dec->sinkpad, &format, &dur))
+        break;
+
+      gst_query_parse_duration (query, &format, NULL);
+
+      /* and convert it into the requested format */
+      if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
+                  dur, &format, &dur))) {
+        gst_query_set_duration (query, format, dur);
+      }
+      break;
+    }
+    default:
+      res = gst_pad_query_default (pad, query);
+      break;
+  }
+
+  gst_object_unref (dec);
+  return res;
+}
+
+static gboolean
+opus_dec_src_event (GstPad * pad, GstEvent * event)
+{
+  gboolean res = FALSE;
+  GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+  GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_SEEK:{
+      GstFormat format, tformat;
+      gdouble rate;
+      GstEvent *real_seek;
+      GstSeekFlags flags;
+      GstSeekType cur_type, stop_type;
+      gint64 cur, stop;
+      gint64 tcur, tstop;
+
+      gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
+          &stop_type, &stop);
+
+      /* we have to ask our peer to seek to time here as we know
+       * nothing about how to generate a granulepos from the src
+       * formats or anything.
+       *
+       * First bring the requested format to time
+       */
+      tformat = GST_FORMAT_TIME;
+      if (!(res = opus_dec_convert (pad, format, cur, &tformat, &tcur)))
+        break;
+      if (!(res = opus_dec_convert (pad, format, stop, &tformat, &tstop)))
+        break;
+
+      /* then seek with time on the peer */
+      real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
+          flags, cur_type, tcur, stop_type, tstop);
+
+      GST_LOG_OBJECT (dec, "seek to %" GST_TIME_FORMAT, GST_TIME_ARGS (tcur));
+
+      res = gst_pad_push_event (dec->sinkpad, real_seek);
+      gst_event_unref (event);
+      break;
+    }
+    default:
+      res = gst_pad_event_default (pad, event);
+      break;
+  }
+
+  gst_object_unref (dec);
+  return res;
+}
+
+static gboolean
+opus_dec_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstOpusDec *dec;
+  gboolean ret = FALSE;
+
+  dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+  GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_NEWSEGMENT:{
+      GstFormat format;
+      gdouble rate, arate;
+      gint64 start, stop, time;
+      gboolean update;
+
+      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+          &start, &stop, &time);
+
+      if (format != GST_FORMAT_TIME)
+        goto newseg_wrong_format;
+
+      if (rate <= 0.0)
+        goto newseg_wrong_rate;
+
+      if (update) {
+        /* time progressed without data, see if we can fill the gap with
+         * some concealment data */
+        if (dec->segment.last_stop < start) {
+          GstClockTime duration;
+
+          duration = start - dec->segment.last_stop;
+          opus_dec_chain_parse_data (dec, NULL, dec->segment.last_stop,
+              duration);
+        }
+      }
+
+      /* now configure the values */
+      gst_segment_set_newsegment_full (&dec->segment, update,
+          rate, arate, GST_FORMAT_TIME, start, stop, time);
+
+      dec->granulepos = -1;
+
+      GST_DEBUG_OBJECT (dec, "segment now: cur = %" GST_TIME_FORMAT " [%"
+          GST_TIME_FORMAT " - %" GST_TIME_FORMAT "]",
+          GST_TIME_ARGS (dec->segment.last_stop),
+          GST_TIME_ARGS (dec->segment.start),
+          GST_TIME_ARGS (dec->segment.stop));
+
+      ret = gst_pad_push_event (dec->srcpad, event);
+      break;
+    }
+    default:
+      ret = gst_pad_event_default (pad, event);
+      break;
+  }
+
+  gst_object_unref (dec);
+  return ret;
+
+  /* ERRORS */
+newseg_wrong_format:
+  {
+    GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
+    gst_object_unref (dec);
+    return FALSE;
+  }
+newseg_wrong_rate:
+  {
+    GST_DEBUG_OBJECT (dec, "negative rates not supported yet");
+    gst_object_unref (dec);
+    return FALSE;
+  }
+}
+
+static GstFlowReturn
+opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf)
+{
+  GstCaps *caps;
+  //gint error = OPUS_OK;
+
+#if 0
+  dec->samples_per_frame = opus_packet_get_samples_per_frame (
+      (const unsigned char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+#endif
+
+#if 0
+  if (memcmp (dec->header.codec_id, "OPUS    ", 8) != 0)
+    goto invalid_header;
+#endif
+
+#if 0
+#ifdef HAVE_OPUS_0_7
+  dec->mode =
+      opus_mode_create (dec->sample_rate, dec->header.frame_size, &error);
+#else
+  dec->mode =
+      opus_mode_create (dec->sample_rate, dec->header.nb_channels,
+      dec->header.frame_size, &error);
+#endif
+  if (!dec->mode)
+    goto mode_init_failed;
+
+  /* initialize the decoder */
+#ifdef HAVE_OPUS_0_11
+  dec->state =
+      opus_decoder_create_custom (dec->mode, dec->header.nb_channels, &error);
+#else
+#ifdef HAVE_OPUS_0_7
+  dec->state = opus_decoder_create (dec->mode, dec->header.nb_channels, &error);
+#else
+  dec->state = opus_decoder_create (dec->mode);
+#endif
+#endif
+#endif
+  dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels);
+  if (!dec->state)
+    goto init_failed;
+
+#if 0
+#ifdef HAVE_OPUS_0_8
+  dec->frame_size = dec->header.frame_size;
+#else
+  opus_mode_info (dec->mode, OPUS_GET_FRAME_SIZE, &dec->frame_size);
+#endif
+#endif
+
+  dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
+      GST_SECOND, dec->sample_rate);
+
+  /* set caps */
+  caps = gst_caps_new_simple ("audio/x-raw-int",
+      "rate", G_TYPE_INT, dec->sample_rate,
+      "channels", G_TYPE_INT, dec->n_channels,
+      "signed", G_TYPE_BOOLEAN, TRUE,
+      "endianness", G_TYPE_INT, G_BYTE_ORDER,
+      "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
+
+  GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
+      dec->sample_rate, dec->n_channels, dec->frame_size);
+
+  if (!gst_pad_set_caps (dec->srcpad, caps))
+    goto nego_failed;
+
+  gst_caps_unref (caps);
+  return GST_FLOW_OK;
+
+  /* ERRORS */
+#if 0
+invalid_header:
+  {
+    GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+        (NULL), ("Invalid header"));
+    return GST_FLOW_ERROR;
+  }
+mode_init_failed:
+  {
+    GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+        (NULL), ("Mode initialization failed: %d", error));
+    return GST_FLOW_ERROR;
+  }
+#endif
+init_failed:
+  {
+    GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+        (NULL), ("couldn't initialize decoder"));
+    return GST_FLOW_ERROR;
+  }
+nego_failed:
+  {
+    GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+        (NULL), ("couldn't negotiate format"));
+    gst_caps_unref (caps);
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+}
+
+#if 0
+static GstFlowReturn
+opus_dec_chain_parse_comments (GstOpusDec * dec, GstBuffer * buf)
+{
+  GstTagList *list;
+  gchar *encoder = NULL;
+
+  list = gst_tag_list_from_vorbiscomment_buffer (buf, NULL, 0, &encoder);
+
+  if (!list) {
+    GST_WARNING_OBJECT (dec, "couldn't decode comments");
+    list = gst_tag_list_new ();
+  }
+
+  if (encoder) {
+    gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+        GST_TAG_ENCODER, encoder, NULL);
+  }
+
+  gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_AUDIO_CODEC, "Opus", NULL);
+
+  if (dec->header.bytes_per_packet > 0) {
+    gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+        GST_TAG_BITRATE, (guint) dec->header.bytes_per_packet * 8, NULL);
+  }
+
+  GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
+
+  gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, list);
+
+  g_free (encoder);
+  g_free (ver);
+
+  return GST_FLOW_OK;
+}
+#endif
+
+static GstFlowReturn
+opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
+    GstClockTime timestamp, GstClockTime duration)
+{
+  GstFlowReturn res = GST_FLOW_OK;
+  gint size;
+  guint8 *data;
+  GstBuffer *outbuf;
+  gint16 *out_data;
+  int n;
+
+  if (timestamp != -1) {
+    dec->segment.last_stop = timestamp;
+    dec->granulepos = -1;
+  }
+
+  if (dec->state == NULL) {
+    GstCaps *caps;
+
+    dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels);
+
+    /* set caps */
+    caps = gst_caps_new_simple ("audio/x-raw-int",
+        "rate", G_TYPE_INT, dec->sample_rate,
+        "channels", G_TYPE_INT, dec->n_channels,
+        "signed", G_TYPE_BOOLEAN, TRUE,
+        "endianness", G_TYPE_INT, G_BYTE_ORDER,
+        "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
+
+    GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
+        dec->sample_rate, dec->n_channels, dec->frame_size);
+
+    if (!gst_pad_set_caps (dec->srcpad, caps))
+      GST_ERROR ("nego failure");
+
+    gst_caps_unref (caps);
+  }
+
+  if (buf) {
+    data = GST_BUFFER_DATA (buf);
+    size = GST_BUFFER_SIZE (buf);
+
+    GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
+
+    /* copy timestamp */
+  } else {
+    /* concealment data, pass NULL as the bits parameters */
+    GST_DEBUG_OBJECT (dec, "creating concealment data");
+    data = NULL;
+    size = 0;
+  }
+
+  GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data));
+  GST_DEBUG ("samples_per_frame %d", opus_packet_get_samples_per_frame (data,
+          48000));
+  GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data));
+
+  res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
+      GST_BUFFER_OFFSET_NONE, dec->frame_samples * dec->n_channels * 2,
+      GST_PAD_CAPS (dec->srcpad), &outbuf);
+
+  if (res != GST_FLOW_OK) {
+    GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
+    return res;
+  }
+
+  out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
+
+  GST_LOG_OBJECT (dec, "decoding frame");
+
+  n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, TRUE);
+  if (n < 0) {
+    GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
+    return GST_FLOW_ERROR;
+  }
+
+  if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
+    timestamp = gst_util_uint64_scale_int (dec->granulepos - dec->frame_size,
+        GST_SECOND, dec->sample_rate);
+  }
+
+  GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
+
+  GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
+  GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
+  if (dec->discont) {
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+    dec->discont = 0;
+  }
+
+  dec->segment.last_stop += dec->frame_duration;
+
+  GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
+      GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+      GST_TIME_ARGS (dec->frame_duration));
+
+  res = gst_pad_push (dec->srcpad, outbuf);
+
+  if (res != GST_FLOW_OK)
+    GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
+
+  return res;
+}
+
+static GstFlowReturn
+opus_dec_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstFlowReturn res;
+  GstOpusDec *dec;
+
+  dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+  if (GST_BUFFER_IS_DISCONT (buf)) {
+    dec->discont = TRUE;
+  }
+
+  res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+      GST_BUFFER_DURATION (buf));
+
+//done:
+  dec->packetno++;
+
+  gst_buffer_unref (buf);
+  gst_object_unref (dec);
+
+  return res;
+}
+
+static GstStateChangeReturn
+opus_dec_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret;
+  GstOpusDec *dec = GST_OPUS_DEC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+    default:
+      break;
+  }
+
+  ret = parent_class->change_state (element, transition);
+  if (ret != GST_STATE_CHANGE_SUCCESS)
+    return ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_opus_dec_reset (dec);
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}

ext/opus/gstopusdec.h

@@ -0,0 +1,77 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2008> 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_DEC_H__
+#define __GST_OPUS_DEC_H__
+
+#include <gst/gst.h>
+#include <opus/opus.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OPUS_DEC \
+  (gst_opus_dec_get_type())
+#define GST_OPUS_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OPUS_DEC,GstOpusDec))
+#define GST_OPUS_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OPUS_DEC,GstOpusDecClass))
+#define GST_IS_OPUS_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OPUS_DEC))
+#define GST_IS_OPUS_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OPUS_DEC))
+
+typedef struct _GstOpusDec GstOpusDec;
+typedef struct _GstOpusDecClass GstOpusDecClass;
+
+struct _GstOpusDec {
+  GstElement            element;
+
+  /* pads */
+  GstPad                *sinkpad;
+  GstPad                *srcpad;
+
+  OpusDecoder          *state;
+  int frame_samples;
+
+  gint                  frame_size;
+  GstClockTime          frame_duration;
+  guint64               packetno;
+
+  GstSegment            segment;    /* STREAM LOCK */
+  gint64                granulepos; /* -1 = needs to be set from current time */
+  gboolean              discont;
+
+  GstBuffer            *streamheader;
+  GstBuffer            *vorbiscomment;
+  GList                *extra_headers;
+
+  int sample_rate;
+  int n_channels;
+};
+
+struct _GstOpusDecClass {
+  GstElementClass parent_class;
+};
+
+GType gst_opus_dec_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_OPUS_DEC_H__ */

ext/opus/gstopusenc.h

@@ -0,0 +1,105 @@
+/* GStreamer Opus Encoder
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2008> 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_ENC_H__
+#define __GST_OPUS_ENC_H__
+
+
+#include <gst/gst.h>
+#include <gst/base/gstadapter.h>
+
+#include <opus/opus.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OPUS_ENC \
+  (gst_opus_enc_get_type())
+#define GST_OPUS_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OPUS_ENC,GstOpusEnc))
+#define GST_OPUS_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OPUS_ENC,GstOpusEncClass))
+#define GST_IS_OPUS_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OPUS_ENC))
+#define GST_IS_OPUS_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OPUS_ENC))
+
+#define MAX_FRAME_SIZE 2000*2
+#define MAX_FRAME_BYTES 2000
+
+typedef struct _GstOpusEnc GstOpusEnc;
+typedef struct _GstOpusEncClass GstOpusEncClass;
+
+struct _GstOpusEnc {
+  GstElement            element;
+
+  /* pads */
+  GstPad                *sinkpad;
+  GstPad                *srcpad;
+
+  //OpusHeader            header;
+  //OpusMode             *mode;
+  OpusEncoder          *state;
+  GstAdapter           *adapter;
+
+  /* properties */
+  gboolean              audio_or_voip;
+  gint                  bitrate;
+  gint                  bandwidth;
+  gint                  frame_size;
+  gboolean              cbr;
+  gboolean              constrained_vbr;
+  gint                  complexity;
+  gboolean              inband_fec;
+  gboolean              dtx;
+  gint                  packet_loss_percentage;
+
+  int frame_samples;
+
+  gint                  n_channels;
+  gint                  sample_rate;
+
+  gboolean              setup;
+  gboolean              header_sent;
+  gboolean              eos;
+
+  guint64               samples_in;
+  guint64               bytes_out;
+
+  guint64               frameno;
+  guint64               frameno_out;
+
+  GstClockTime     start_ts;
+  GstClockTime     next_ts;
+  guint64          granulepos_offset;
+};
+
+struct _GstOpusEncClass {
+  GstElementClass parent_class;
+
+  /* signals */
+  void (*frame_encoded) (GstElement *element);
+};
+
+GType gst_opus_enc_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_OPUS_ENC_H__ */