opus: duplicate from CELT

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

8 files changed, 2334 insertions, 0 deletions

diff --git a/configure.ac b/configure.ac
index 7322eaa93..1aac3360a 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1385,6 +1385,19 @@ AG_GST_CHECK_FEATURE(OPENCV, [opencv plugins], opencv, [
   AC_SUBST(OPENCV_LIBS)
 ])
 
+dnl *** Opus ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_OPUS, true)
+AG_GST_CHECK_FEATURE(OPUS, [opus], opus, [
+  PKG_CHECK_MODULES(OPUS, opus >= 0.9.4, [
+          AC_DEFINE([HAVE_OPUS], 1, [Define if Opus >= 0.9.4 is installed])
+          HAVE_OPUS="yes"
+  ], [
+          HAVE_OPUS="no"
+  ])
+  AC_SUBST(OPUS_CFLAGS)
+  AC_SUBST(OPUS_LIBS)
+])
+
 dnl *** rsvg ***
 translit(dnm, m, l) AM_CONDITIONAL(USE_RSVG, true)
 AG_GST_CHECK_FEATURE(RSVG, [rsvg decoder], rsvg, [
@@ -1750,6 +1763,7 @@ AM_CONDITIONAL(USE_NEON, false)
 AM_CONDITIONAL(USE_OFA, false)
 AM_CONDITIONAL(USE_OPENAL, false)
 AM_CONDITIONAL(USE_OPENCV, false)
+AM_CONDITIONAL(USE_OPUS, false)
 AM_CONDITIONAL(USE_RSVG, false)
 AM_CONDITIONAL(USE_TIMIDITY, false)
 AM_CONDITIONAL(USE_WILDMIDI, false)
@@ -1994,6 +2008,7 @@ ext/neon/Makefile
 ext/ofa/Makefile
 ext/openal/Makefile
 ext/opencv/Makefile
+ext/opus/Makefile
 ext/rsvg/Makefile
 ext/resindvd/Makefile
 ext/rtmp/Makefile
diff --git a/ext/Makefile.am b/ext/Makefile.am
index 70d4c69c3..2a6f8ec76 100644
--- a/ext/Makefile.am
+++ b/ext/Makefile.am
@@ -262,6 +262,12 @@ else
 OPENCV_DIR=
 endif
 
+if USE_OPUS
+OPUS_DIR=opus
+else
+OPUS_DIR=
+endif
+
 if USE_RSVG
 RSVG_DIR=rsvg
 else
@@ -419,6 +425,7 @@ SUBDIRS=\
 	$(OFA_DIR) \
 	$(OPENAL_DIR) \
 	$(OPENCV_DIR) \
+	$(OPUS_DIR) \
 	$(RSVG_DIR) \
 	$(SCHRO_DIR) \
 	$(SDL_DIR) \
@@ -471,6 +478,7 @@ DIST_SUBDIRS = \
 	ofa \
 	openal \
 	opencv \
+	opus \
 	rsvg \
 	resindvd \
 	schroedinger \
diff --git a/ext/opus/Makefile.am b/ext/opus/Makefile.am
new file mode 100644
index 000000000..aa50ba96e
--- /dev/null
+++ b/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 
diff --git a/ext/opus/gstopus.c b/ext/opus/gstopus.c
new file mode 100644
index 000000000..65e9dcdc5
--- /dev/null
+++ b/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)
diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c
new file mode 100644
index 000000000..47c06cec0
--- /dev/null
+++ b/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;
+}
diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h
new file mode 100644
index 000000000..886a90753
--- /dev/null
+++ b/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__ */
diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c
new file mode 100644
index 000000000..db57ff75d
--- /dev/null
+++ b/ext/opus/gstopusenc.c
@@ -0,0 +1,1198 @@
+/* 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.
+ */
+
+/*
+ * Based on the speexenc element
+ */
+
+/**
+ * SECTION:element-opusenc
+ * @see_also: opusdec, oggmux
+ *
+ * This element encodes raw audio to OPUS.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! opusenc ! oggmux ! filesink location=sine.ogg
+ * ]| Encode a test sine signal to Ogg/OPUS.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <math.h>
+#include <opus/opus.h>
+
+#include <gst/gsttagsetter.h>
+#include <gst/tag/tag.h>
+#include <gst/audio/audio.h>
+#include "gstopusenc.h"
+
+GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
+#define GST_CAT_DEFAULT opusenc_debug
+
+#define GST_OPUS_ENC_TYPE_BANDWIDTH (gst_opus_enc_bandwidth_get_type())
+static GType
+gst_opus_enc_bandwidth_get_type (void)
+{
+  static const GEnumValue values[] = {
+    {OPUS_BANDWIDTH_NARROWBAND, "Narrow band", "narrowband"},
+    {OPUS_BANDWIDTH_MEDIUMBAND, "Medium band", "mediumband"},
+    {OPUS_BANDWIDTH_WIDEBAND, "Wide band", "wideband"},
+    {OPUS_BANDWIDTH_SUPERWIDEBAND, "Super wide band", "superwideband"},
+    {OPUS_BANDWIDTH_FULLBAND, "Full band", "fullband"},
+    {OPUS_BANDWIDTH_AUTO, "Auto", "auto"},
+    {0, NULL, NULL}
+  };
+  static volatile GType id = 0;
+
+  if (g_once_init_enter ((gsize *) & id)) {
+    GType _id;
+
+    _id = g_enum_register_static ("GstOpusEncBandwidth", values);
+
+    g_once_init_leave ((gsize *) & id, _id);
+  }
+
+  return id;
+}
+
+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 ], "
+        "endianness = (int) BYTE_ORDER, "
+        "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
+    );
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-opus, "
+        "rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
+        "channels = (int) [ 1, 2 ], " "frame-size = (int) [ 2, 60 ]")
+    );
+
+#define DEFAULT_AUDIO           TRUE
+#define DEFAULT_BITRATE         64000
+#define DEFAULT_BANDWIDTH       OPUS_BANDWIDTH_FULLBAND
+#define DEFAULT_FRAMESIZE       20
+#define DEFAULT_CBR             TRUE
+#define DEFAULT_CONSTRAINED_VBR TRUE
+#define DEFAULT_COMPLEXITY      10
+#define DEFAULT_INBAND_FEC      FALSE
+#define DEFAULT_DTX             FALSE
+#define DEFAULT_PACKET_LOSS_PERCENT 0
+
+enum
+{
+  PROP_0,
+  PROP_AUDIO,
+  PROP_BITRATE,
+  PROP_BANDWIDTH,
+  PROP_FRAME_SIZE,
+  PROP_CBR,
+  PROP_CONSTRAINED_VBR,
+  PROP_COMPLEXITY,
+  PROP_INBAND_FEC,
+  PROP_DTX,
+  PROP_PACKET_LOSS_PERCENT
+};
+
+static void gst_opus_enc_finalize (GObject * object);
+
+static gboolean gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_opus_enc_chain (GstPad * pad, GstBuffer * buf);
+static gboolean gst_opus_enc_setup (GstOpusEnc * enc);
+
+static void gst_opus_enc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_opus_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static GstStateChangeReturn gst_opus_enc_change_state (GstElement * element,
+    GstStateChange transition);
+
+static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush);
+
+static void
+gst_opus_enc_setup_interfaces (GType opusenc_type)
+{
+  static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL };
+  const GInterfaceInfo preset_interface_info = {
+    NULL,                       /* interface_init */
+    NULL,                       /* interface_finalize */
+    NULL                        /* interface_data */
+  };
+
+  g_type_add_interface_static (opusenc_type, GST_TYPE_TAG_SETTER,
+      &tag_setter_info);
+  g_type_add_interface_static (opusenc_type, GST_TYPE_PRESET,
+      &preset_interface_info);
+
+  GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder");
+}
+
+GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstElement, GST_TYPE_ELEMENT,
+    gst_opus_enc_setup_interfaces);
+
+static void
+gst_opus_enc_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 (&src_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_factory));
+  gst_element_class_set_details_simple (element_class, "Opus audio encoder",
+      "Codec/Encoder/Audio",
+      "Encodes audio in Opus format",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_opus_enc_class_init (GstOpusEncClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  gobject_class->set_property = gst_opus_enc_set_property;
+  gobject_class->get_property = gst_opus_enc_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_AUDIO,
+      g_param_spec_boolean ("audio", "Audio or voice",
+          "Audio or voice", DEFAULT_AUDIO,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
+      g_param_spec_int ("bitrate", "Encoding Bit-rate",
+          "Specify an encoding bit-rate (in bps).",
+          1, 320000, DEFAULT_BITRATE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
+      g_param_spec_enum ("bandwidth", "Band Width",
+          "Audio Band Width", GST_OPUS_ENC_TYPE_BANDWIDTH, DEFAULT_BANDWIDTH,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_FRAME_SIZE,
+      g_param_spec_int ("frame-size", "Frame Size",
+          "The duration of an audio frame, in ms", 2, 60, DEFAULT_FRAMESIZE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_CBR,
+      g_param_spec_boolean ("cbr", "Constant bit rate",
+          "Constant bit rate", DEFAULT_CBR,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_CONSTRAINED_VBR,
+      g_param_spec_boolean ("constrained-cbr", "Constrained VBR",
+          "Constrained VBR", DEFAULT_CONSTRAINED_VBR,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_COMPLEXITY,
+      g_param_spec_int ("complexity", "Complexity",
+          "Complexity", 0, 10, DEFAULT_COMPLEXITY,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_INBAND_FEC,
+      g_param_spec_boolean ("inband-fec", "In-band FEC",
+          "Enable forward error correction", DEFAULT_INBAND_FEC,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_DTX,
+      g_param_spec_boolean ("dtx", "DTX",
+          "DTX", DEFAULT_DTX, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (G_OBJECT_CLASS (klass),
+      PROP_PACKET_LOSS_PERCENT, g_param_spec_int ("packet-loss-percentage",
+          "Loss percentage", "Packet loss percentage", 0, 100,
+          DEFAULT_PACKET_LOSS_PERCENT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize);
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_opus_enc_change_state);
+}
+
+static void
+gst_opus_enc_finalize (GObject * object)
+{
+  GstOpusEnc *enc;
+
+  enc = GST_OPUS_ENC (object);
+
+  g_object_unref (enc->adapter);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_opus_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+  GstOpusEnc *enc;
+  GstStructure *structure;
+  GstCaps *otherpadcaps;
+
+  enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+  enc->setup = FALSE;
+  enc->frame_size = DEFAULT_FRAMESIZE;
+  otherpadcaps = gst_pad_get_allowed_caps (pad);
+
+  structure = gst_caps_get_structure (caps, 0);
+  gst_structure_get_int (structure, "channels", &enc->n_channels);
+  gst_structure_get_int (structure, "rate", &enc->sample_rate);
+
+  if (otherpadcaps) {
+    if (!gst_caps_is_empty (otherpadcaps)) {
+      GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0);
+      gst_structure_get_int (ps, "frame-size", &enc->frame_size);
+    }
+    gst_caps_unref (otherpadcaps);
+  }
+
+  GST_ERROR_OBJECT (pad, "channels=%d rate=%d frame-size=%d",
+      enc->n_channels, enc->sample_rate, enc->frame_size);
+  switch (enc->frame_size) {
+    case 2:
+      enc->frame_samples = enc->sample_rate / 400;
+      break;
+    case 5:
+      enc->frame_samples = enc->sample_rate / 200;
+      break;
+    case 10:
+      enc->frame_samples = enc->sample_rate / 100;
+      break;
+    case 20:
+      enc->frame_samples = enc->sample_rate / 50;
+      break;
+    case 40:
+      enc->frame_samples = enc->sample_rate / 20;
+      break;
+    case 60:
+      enc->frame_samples = 3 * enc->sample_rate / 50;
+      break;
+    default:
+      return FALSE;
+      break;
+  }
+  GST_ERROR ("frame_samples %d", enc->frame_samples);
+
+  gst_opus_enc_setup (enc);
+
+  return TRUE;
+}
+
+
+static GstCaps *
+gst_opus_enc_sink_getcaps (GstPad * pad)
+{
+  GstCaps *caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+  GstCaps *peercaps = NULL;
+  GstOpusEnc *enc = GST_OPUS_ENC (gst_pad_get_parent_element (pad));
+
+  peercaps = gst_pad_peer_get_caps (enc->srcpad);
+
+  if (peercaps) {
+    if (!gst_caps_is_empty (peercaps) && !gst_caps_is_any (peercaps)) {
+      GstStructure *ps = gst_caps_get_structure (peercaps, 0);
+      GstStructure *s = gst_caps_get_structure (caps, 0);
+      gint rate, channels;
+
+      if (gst_structure_get_int (ps, "rate", &rate)) {
+        gst_structure_fixate_field_nearest_int (s, "rate", rate);
+      }
+
+      if (gst_structure_get_int (ps, "channels", &channels)) {
+        gst_structure_fixate_field_nearest_int (s, "channels", channels);
+      }
+    }
+    gst_caps_unref (peercaps);
+  }
+
+  gst_object_unref (enc);
+
+  return caps;
+}
+
+
+static gboolean
+gst_opus_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
+    GstFormat * dest_format, gint64 * dest_value)
+{
+  gboolean res = TRUE;
+  GstOpusEnc *enc;
+  gint64 avg;
+
+  enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+
+  if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->sample_rate == 0)
+    return FALSE;
+
+  avg = (enc->bytes_out * enc->sample_rate) / (enc->samples_in);
+
+  switch (src_format) {
+    case GST_FORMAT_BYTES:
+      switch (*dest_format) {
+        case GST_FORMAT_TIME:
+          *dest_value = src_value * GST_SECOND / avg;
+          break;
+        default:
+          res = FALSE;
+      }
+      break;
+    case GST_FORMAT_TIME:
+      switch (*dest_format) {
+        case GST_FORMAT_BYTES:
+          *dest_value = src_value * avg / GST_SECOND;
+          break;
+        default:
+          res = FALSE;
+      }
+      break;
+    default:
+      res = FALSE;
+  }
+  return res;
+}
+
+static gboolean
+gst_opus_enc_convert_sink (GstPad * pad, GstFormat src_format,
+    gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+{
+  gboolean res = TRUE;
+  guint scale = 1;
+  gint bytes_per_sample;
+  GstOpusEnc *enc;
+
+  enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+
+  bytes_per_sample = enc->n_channels * 2;
+
+  switch (src_format) {
+    case GST_FORMAT_BYTES:
+      switch (*dest_format) {
+        case GST_FORMAT_DEFAULT:
+          if (bytes_per_sample == 0)
+            return FALSE;
+          *dest_value = src_value / bytes_per_sample;
+          break;
+        case GST_FORMAT_TIME:
+        {
+          gint byterate = bytes_per_sample * enc->sample_rate;
+
+          if (byterate == 0)
+            return FALSE;
+          *dest_value = src_value * GST_SECOND / byterate;
+          break;
+        }
+        default:
+          res = FALSE;
+      }
+      break;
+    case GST_FORMAT_DEFAULT:
+      switch (*dest_format) {
+        case GST_FORMAT_BYTES:
+          *dest_value = src_value * bytes_per_sample;
+          break;
+        case GST_FORMAT_TIME:
+          if (enc->sample_rate == 0)
+            return FALSE;
+          *dest_value = src_value * GST_SECOND / enc->sample_rate;
+          break;
+        default:
+          res = FALSE;
+      }
+      break;
+    case GST_FORMAT_TIME:
+      switch (*dest_format) {
+        case GST_FORMAT_BYTES:
+          scale = bytes_per_sample;
+          /* fallthrough */
+        case GST_FORMAT_DEFAULT:
+          *dest_value = src_value * scale * enc->sample_rate / GST_SECOND;
+          break;
+        default:
+          res = FALSE;
+      }
+      break;
+    default:
+      res = FALSE;
+  }
+  return res;
+}
+
+static gint64
+gst_opus_enc_get_latency (GstOpusEnc * enc)
+{
+  return gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
+      enc->sample_rate);
+}
+
+static const GstQueryType *
+gst_opus_enc_get_query_types (GstPad * pad)
+{
+  static const GstQueryType gst_opus_enc_src_query_types[] = {
+    GST_QUERY_POSITION,
+    GST_QUERY_DURATION,
+    GST_QUERY_CONVERT,
+    GST_QUERY_LATENCY,
+    0
+  };
+
+  return gst_opus_enc_src_query_types;
+}
+
+static gboolean
+gst_opus_enc_src_query (GstPad * pad, GstQuery * query)
+{
+  gboolean res = TRUE;
+  GstOpusEnc *enc;
+
+  enc = GST_OPUS_ENC (gst_pad_get_parent (pad));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_POSITION:
+    {
+      GstFormat fmt, req_fmt;
+      gint64 pos, val;
+
+      gst_query_parse_position (query, &req_fmt, NULL);
+      if ((res = gst_pad_query_peer_position (enc->sinkpad, &req_fmt, &val))) {
+        gst_query_set_position (query, req_fmt, val);
+        break;
+      }
+
+      fmt = GST_FORMAT_TIME;
+      if (!(res = gst_pad_query_peer_position (enc->sinkpad, &fmt, &pos)))
+        break;
+
+      if ((res =
+              gst_pad_query_peer_convert (enc->sinkpad, fmt, pos, &req_fmt,
+                  &val)))
+        gst_query_set_position (query, req_fmt, val);
+
+      break;
+    }
+    case GST_QUERY_DURATION:
+    {
+      GstFormat fmt, req_fmt;
+      gint64 dur, val;
+
+      gst_query_parse_duration (query, &req_fmt, NULL);
+      if ((res = gst_pad_query_peer_duration (enc->sinkpad, &req_fmt, &val))) {
+        gst_query_set_duration (query, req_fmt, val);
+        break;
+      }
+
+      fmt = GST_FORMAT_TIME;
+      if (!(res = gst_pad_query_peer_duration (enc->sinkpad, &fmt, &dur)))
+        break;
+
+      if ((res =
+              gst_pad_query_peer_convert (enc->sinkpad, fmt, dur, &req_fmt,
+                  &val))) {
+        gst_query_set_duration (query, req_fmt, val);
+      }
+      break;
+    }
+    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);
+      if (!(res = gst_opus_enc_convert_src (pad, src_fmt, src_val, &dest_fmt,
+                  &dest_val)))
+        goto error;
+      gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+      break;
+    }
+    case GST_QUERY_LATENCY:
+    {
+      gboolean live;
+      GstClockTime min_latency, max_latency;
+      gint64 latency;
+
+      if ((res = gst_pad_peer_query (pad, query))) {
+        gst_query_parse_latency (query, &live, &min_latency, &max_latency);
+
+        latency = gst_opus_enc_get_latency (enc);
+
+        /* add our latency */
+        min_latency += latency;
+        if (max_latency != -1)
+          max_latency += latency;
+
+        gst_query_set_latency (query, live, min_latency, max_latency);
+      }
+      break;
+    }
+    default:
+      res = gst_pad_peer_query (pad, query);
+      break;
+  }
+
+error:
+
+  gst_object_unref (enc);
+
+  return res;
+}
+
+static gboolean
+gst_opus_enc_sink_query (GstPad * pad, GstQuery * query)
+{
+  gboolean res = TRUE;
+
+  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);
+      if (!(res =
+              gst_opus_enc_convert_sink (pad, src_fmt, src_val, &dest_fmt,
+                  &dest_val)))
+        goto error;
+      gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+      break;
+    }
+    default:
+      res = gst_pad_query_default (pad, query);
+      break;
+  }
+
+error:
+  return res;
+}
+
+static void
+gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass)
+{
+  enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
+  gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
+  gst_pad_set_event_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_sinkevent));
+  gst_pad_set_chain_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_chain));
+  gst_pad_set_setcaps_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_sink_setcaps));
+  gst_pad_set_getcaps_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps));
+  gst_pad_set_query_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_sink_query));
+
+  enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
+  gst_pad_set_query_function (enc->srcpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_src_query));
+  gst_pad_set_query_type_function (enc->srcpad,
+      GST_DEBUG_FUNCPTR (gst_opus_enc_get_query_types));
+  gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+
+  enc->n_channels = -1;
+  enc->sample_rate = -1;
+  enc->frame_samples = 0;
+
+  enc->bitrate = DEFAULT_BITRATE;
+  enc->bandwidth = DEFAULT_BANDWIDTH;
+  enc->frame_size = DEFAULT_FRAMESIZE;
+  enc->cbr = DEFAULT_CBR;
+  enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR;
+  enc->complexity = DEFAULT_COMPLEXITY;
+  enc->inband_fec = DEFAULT_INBAND_FEC;
+  enc->dtx = DEFAULT_DTX;
+  enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT;
+
+  enc->setup = FALSE;
+  enc->header_sent = FALSE;
+
+  enc->adapter = gst_adapter_new ();
+}
+
+#if 0
+static GstBuffer *
+gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc)
+{
+  const GstTagList *tags;
+  GstTagList *empty_tags = NULL;
+  GstBuffer *comments = NULL;
+
+  tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
+
+  GST_DEBUG_OBJECT (enc, "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 ();
+    tags = empty_tags;
+  }
+  comments = gst_tag_list_to_vorbiscomment_buffer (tags, NULL,
+      0, "Encoded with GStreamer Opusenc");
+
+  GST_BUFFER_OFFSET (comments) = enc->bytes_out;
+  GST_BUFFER_OFFSET_END (comments) = 0;
+
+  if (empty_tags)
+    gst_tag_list_free (empty_tags);
+
+  return comments;
+}
+#endif
+
+static gboolean
+gst_opus_enc_setup (GstOpusEnc * enc)
+{
+  //gint error = OPUS_OK;
+
+  enc->setup = FALSE;
+
+#if 0
+#ifdef HAVE_OPUS_0_7
+  enc->mode = opus_mode_create (enc->rate, enc->frame_size, &error);
+#else
+  enc->mode =
+      opus_mode_create (enc->rate, enc->n_channels, enc->frame_size, &error);
+#endif
+  if (!enc->mode)
+    goto mode_initialization_failed;
+
+#ifdef HAVE_OPUS_0_11
+  opus_header_init (&enc->header, enc->mode, enc->frame_size, enc->n_channels);
+#else
+#ifdef HAVE_OPUS_0_7
+  opus_header_init (&enc->header, enc->mode, enc->n_channels);
+#else
+  opus_header_init (&enc->header, enc->mode);
+#endif
+#endif
+  enc->header.nb_channels = enc->n_channels;
+
+#ifdef HAVE_OPUS_0_8
+  enc->frame_size = enc->header.frame_size;
+#else
+  opus_mode_info (enc->mode, OPUS_GET_FRAME_SIZE, &enc->frame_size);
+#endif
+#endif
+
+#if 0
+#ifdef HAVE_OPUS_0_11
+  enc->state = opus_encoder_create_custom (enc->mode, enc->n_channels, &error);
+#else
+#ifdef HAVE_OPUS_0_7
+  enc->state = opus_encoder_create (enc->mode, enc->n_channels, &error);
+#else
+  enc->state = opus_encoder_create (enc->mode);
+#endif
+#endif
+#endif
+  enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels,
+      enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP);
+  if (!enc->state)
+    goto encoder_creation_failed;
+
+  opus_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
+  opus_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth), 0);
+  opus_encoder_ctl (enc->state, OPUS_SET_VBR_FLAG (!enc->cbr), 0);
+  opus_encoder_ctl (enc->state, OPUS_SET_VBR_CONSTRAINT (enc->constrained_vbr),
+      0);
+  opus_encoder_ctl (enc->state, OPUS_SET_COMPLEXITY (enc->complexity), 0);
+  opus_encoder_ctl (enc->state, OPUS_SET_INBAND_FEC_FLAG (enc->inband_fec), 0);
+  opus_encoder_ctl (enc->state, OPUS_SET_DTX_FLAG (enc->dtx), 0);
+  opus_encoder_ctl (enc->state,
+      OPUS_SET_PACKET_LOSS_PERC (enc->packet_loss_percentage), 0);
+
+  GST_LOG_OBJECT (enc, "we have frame size %d", enc->frame_size);
+
+  enc->setup = TRUE;
+
+  return TRUE;
+
+#if 0
+mode_initialization_failed:
+  GST_ERROR_OBJECT (enc, "Mode initialization failed: %d", error);
+  return FALSE;
+#endif
+
+encoder_creation_failed:
+  GST_ERROR_OBJECT (enc, "Encoder creation failed");
+  return FALSE;
+}
+
+
+/* push out the buffer and do internal bookkeeping */
+static GstFlowReturn
+gst_opus_enc_push_buffer (GstOpusEnc * enc, GstBuffer * buffer)
+{
+  guint size;
+
+  size = GST_BUFFER_SIZE (buffer);
+
+  enc->bytes_out += size;
+
+  GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size);
+
+  return gst_pad_push (enc->srcpad, buffer);
+}
+
+#if 0
+static GstCaps *
+gst_opus_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1,
+    GstBuffer * buf2)
+{
+  GstStructure *structure = NULL;
+  GstBuffer *buf;
+  GValue array = { 0 };
+  GValue value = { 0 };
+
+  caps = gst_caps_make_writable (caps);
+  structure = gst_caps_get_structure (caps, 0);
+
+  g_assert (gst_buffer_is_metadata_writable (buf1));
+  g_assert (gst_buffer_is_metadata_writable (buf2));
+
+  /* mark buffers */
+  GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS);
+  GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS);
+
+  /* put buffers in a fixed list */
+  g_value_init (&array, GST_TYPE_ARRAY);
+  g_value_init (&value, GST_TYPE_BUFFER);
+  buf = gst_buffer_copy (buf1);
+  gst_value_set_buffer (&value, buf);
+  gst_buffer_unref (buf);
+  gst_value_array_append_value (&array, &value);
+  g_value_unset (&value);
+  g_value_init (&value, GST_TYPE_BUFFER);
+  buf = gst_buffer_copy (buf2);
+  gst_value_set_buffer (&value, buf);
+  gst_buffer_unref (buf);
+  gst_value_array_append_value (&array, &value);
+  gst_structure_set_value (structure, "streamheader", &array);
+  g_value_unset (&value);
+  g_value_unset (&array);
+
+  return caps;
+}
+#endif
+
+
+static gboolean
+gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event)
+{
+  gboolean res = TRUE;
+  GstOpusEnc *enc;
+
+  enc = GST_OPUS_ENC (gst_pad_get_parent (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+      gst_opus_enc_encode (enc, TRUE);
+      res = gst_pad_event_default (pad, event);
+      break;
+    case GST_EVENT_TAG:
+    {
+      GstTagList *list;
+      GstTagSetter *setter = GST_TAG_SETTER (enc);
+      const GstTagMergeMode mode = gst_tag_setter_get_tag_merge_mode (setter);
+
+      gst_event_parse_tag (event, &list);
+      gst_tag_setter_merge_tags (setter, list, mode);
+      res = gst_pad_event_default (pad, event);
+      break;
+    }
+    default:
+      res = gst_pad_event_default (pad, event);
+      break;
+  }
+
+  gst_object_unref (enc);
+
+  return res;
+}
+
+static GstFlowReturn
+gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush)
+{
+
+  GstFlowReturn ret = GST_FLOW_OK;
+  gint bytes = enc->frame_samples * 2 * enc->n_channels;
+  gint bytes_per_packet;
+
+  bytes_per_packet =
+      (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8;
+
+  if (flush && gst_adapter_available (enc->adapter) % bytes != 0) {
+    guint diff = gst_adapter_available (enc->adapter) % bytes;
+    GstBuffer *buf = gst_buffer_new_and_alloc (diff);
+
+    memset (GST_BUFFER_DATA (buf), 0, diff);
+    gst_adapter_push (enc->adapter, buf);
+  }
+
+
+  while (gst_adapter_available (enc->adapter) >= bytes) {
+    gint16 *data;
+    gint outsize;
+    GstBuffer *outbuf;
+
+    ret = gst_pad_alloc_buffer_and_set_caps (enc->srcpad,
+        GST_BUFFER_OFFSET_NONE, bytes_per_packet, GST_PAD_CAPS (enc->srcpad),
+        &outbuf);
+
+    if (GST_FLOW_OK != ret)
+      goto done;
+
+    data = (gint16 *) gst_adapter_take (enc->adapter, bytes);
+    enc->samples_in += enc->frame_samples;
+
+    GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
+        enc->frame_samples, bytes);
+
+    outsize = opus_encode (enc->state, data, enc->frame_samples,
+        GST_BUFFER_DATA (outbuf), bytes_per_packet);
+
+    g_free (data);
+
+    if (outsize < 0) {
+      GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
+      ret = GST_FLOW_ERROR;
+      goto done;
+    }
+
+    GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts +
+        gst_util_uint64_scale_int (enc->frameno_out * enc->frame_samples,
+        GST_SECOND, enc->sample_rate);
+    GST_BUFFER_DURATION (outbuf) =
+        gst_util_uint64_scale_int (enc->frame_samples, GST_SECOND,
+        enc->sample_rate);
+    GST_BUFFER_OFFSET (outbuf) =
+        gst_util_uint64_scale_int (GST_BUFFER_OFFSET_END (outbuf), GST_SECOND,
+        enc->sample_rate);
+
+    enc->frameno++;
+    enc->frameno_out++;
+
+    ret = gst_opus_enc_push_buffer (enc, outbuf);
+
+    if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret))
+      goto done;
+  }
+
+done:
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_opus_enc_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstOpusEnc *enc;
+  GstFlowReturn ret = GST_FLOW_OK;
+
+  enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+
+  if (!enc->setup)
+    goto not_setup;
+
+#if 0
+  if (!enc->header_sent) {
+    /* Opus streams 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.
+       We merely need to make the headers, then pass them to libopus 
+       one at a time; libopus handles the additional Ogg bitstream 
+       constraints */
+    GstBuffer *buf1, *buf2;
+    GstCaps *caps;
+    guchar data[100];
+
+    /* create header buffer */
+    opus_header_to_packet (&enc->header, data, 100);
+    buf1 = gst_opus_enc_buffer_from_data (enc, data, 100, 0);
+
+    /* create comment buffer */
+    buf2 = gst_opus_enc_create_metadata_buffer (enc);
+
+    /* mark and put on caps */
+    caps = gst_pad_get_caps (enc->srcpad);
+    caps = gst_opus_enc_set_header_on_caps (caps, buf1, buf2);
+
+    gst_caps_set_simple (caps,
+        "rate", G_TYPE_INT, enc->sample_rate,
+        "channels", G_TYPE_INT, enc->n_channels,
+        "frame-size", G_TYPE_INT, enc->frame_size, NULL);
+
+    /* negotiate with these caps */
+    GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
+    GST_LOG_OBJECT (enc, "rate=%d channels=%d frame-size=%d",
+        enc->sample_rate, enc->n_channels, enc->frame_size);
+    gst_pad_set_caps (enc->srcpad, caps);
+
+    gst_buffer_set_caps (buf1, caps);
+    gst_buffer_set_caps (buf2, caps);
+    gst_caps_unref (caps);
+
+    /* push out buffers */
+    ret = gst_opus_enc_push_buffer (enc, buf1);
+
+    if (ret != GST_FLOW_OK) {
+      gst_buffer_unref (buf2);
+      goto done;
+    }
+
+    ret = gst_opus_enc_push_buffer (enc, buf2);
+
+    if (ret != GST_FLOW_OK)
+      goto done;
+
+    enc->header_sent = TRUE;
+  }
+#endif
+
+  GST_DEBUG_OBJECT (enc, "received buffer of %u bytes", GST_BUFFER_SIZE (buf));
+
+  /* Save the timestamp of the first buffer. This will be later
+   * used as offset for all following buffers */
+  if (enc->start_ts == GST_CLOCK_TIME_NONE) {
+    if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+      enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
+    } else {
+      enc->start_ts = 0;
+    }
+  }
+
+
+  /* Check if we have a continous stream, if not drop some samples or the buffer or
+   * insert some silence samples */
+  if (enc->next_ts != GST_CLOCK_TIME_NONE &&
+      GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
+    guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
+    guint64 diff_bytes;
+
+    GST_WARNING_OBJECT (enc, "Buffer is older than previous "
+        "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
+        "), cannot handle. Clipping buffer.",
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (enc->next_ts));
+
+    diff_bytes =
+        GST_CLOCK_TIME_TO_FRAMES (diff, enc->sample_rate) * enc->n_channels * 2;
+    if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
+      gst_buffer_unref (buf);
+      return GST_FLOW_OK;
+    }
+    buf = gst_buffer_make_metadata_writable (buf);
+    GST_BUFFER_DATA (buf) += diff_bytes;
+    GST_BUFFER_SIZE (buf) -= diff_bytes;
+
+    GST_BUFFER_TIMESTAMP (buf) += diff;
+    if (GST_BUFFER_DURATION_IS_VALID (buf))
+      GST_BUFFER_DURATION (buf) -= diff;
+  }
+
+  if (enc->next_ts != GST_CLOCK_TIME_NONE
+      && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+    guint64 max_diff =
+        gst_util_uint64_scale (enc->frame_size, GST_SECOND, enc->sample_rate);
+
+    if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
+        GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > max_diff) {
+      GST_WARNING_OBJECT (enc,
+          "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
+          GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, max_diff);
+
+      gst_opus_enc_encode (enc, TRUE);
+
+      enc->frameno_out = 0;
+      enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
+    }
+  }
+
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
+      && GST_BUFFER_DURATION_IS_VALID (buf))
+    enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+  else
+    enc->next_ts = GST_CLOCK_TIME_NONE;
+
+  /* push buffer to adapter */
+  gst_adapter_push (enc->adapter, buf);
+  buf = NULL;
+
+  ret = gst_opus_enc_encode (enc, FALSE);
+
+done:
+
+  if (buf)
+    gst_buffer_unref (buf);
+
+  return ret;
+
+  /* ERRORS */
+not_setup:
+  {
+    GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
+        ("encoder not initialized (input is not audio?)"));
+    ret = GST_FLOW_NOT_NEGOTIATED;
+    goto done;
+  }
+
+}
+
+
+static void
+gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstOpusEnc *enc;
+
+  enc = GST_OPUS_ENC (object);
+
+  switch (prop_id) {
+    case PROP_AUDIO:
+      g_value_set_boolean (value, enc->audio_or_voip);
+      break;
+    case PROP_BITRATE:
+      g_value_set_int (value, enc->bitrate);
+      break;
+    case PROP_BANDWIDTH:
+      g_value_set_int (value, enc->bandwidth);
+      break;
+    case PROP_FRAME_SIZE:
+      g_value_set_int (value, enc->frame_size);
+      break;
+    case PROP_CBR:
+      g_value_set_boolean (value, enc->cbr);
+      break;
+    case PROP_CONSTRAINED_VBR:
+      g_value_set_boolean (value, enc->constrained_vbr);
+      break;
+    case PROP_COMPLEXITY:
+      g_value_set_int (value, enc->complexity);
+      break;
+    case PROP_INBAND_FEC:
+      g_value_set_boolean (value, enc->inband_fec);
+      break;
+    case PROP_DTX:
+      g_value_set_boolean (value, enc->dtx);
+      break;
+    case PROP_PACKET_LOSS_PERCENT:
+      g_value_set_int (value, enc->packet_loss_percentage);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_opus_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOpusEnc *enc;
+
+  enc = GST_OPUS_ENC (object);
+
+  switch (prop_id) {
+    case PROP_AUDIO:
+      enc->audio_or_voip = g_value_get_boolean (value);
+      break;
+    case PROP_BITRATE:
+      enc->bitrate = g_value_get_int (value);
+      break;
+    case PROP_BANDWIDTH:
+      enc->bandwidth = g_value_get_int (value);
+      break;
+    case PROP_FRAME_SIZE:
+      enc->frame_size = g_value_get_int (value);
+      break;
+    case PROP_CBR:
+      enc->cbr = g_value_get_boolean (value);
+      break;
+    case PROP_CONSTRAINED_VBR:
+      enc->constrained_vbr = g_value_get_boolean (value);
+      break;
+    case PROP_COMPLEXITY:
+      enc->complexity = g_value_get_int (value);
+      break;
+    case PROP_INBAND_FEC:
+      enc->inband_fec = g_value_get_boolean (value);
+      break;
+    case PROP_DTX:
+      enc->dtx = g_value_get_boolean (value);
+      break;
+    case PROP_PACKET_LOSS_PERCENT:
+      enc->packet_loss_percentage = g_value_get_int (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_opus_enc_change_state (GstElement * element, GstStateChange transition)
+{
+  GstOpusEnc *enc = GST_OPUS_ENC (element);
+  GstStateChangeReturn res;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      enc->frameno = 0;
+      enc->samples_in = 0;
+      enc->frameno_out = 0;
+      enc->start_ts = GST_CLOCK_TIME_NONE;
+      enc->next_ts = GST_CLOCK_TIME_NONE;
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      /* fall through */
+    default:
+      break;
+  }
+
+  res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (res == GST_STATE_CHANGE_FAILURE)
+    return res;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      enc->setup = FALSE;
+      enc->header_sent = FALSE;
+      if (enc->state) {
+        opus_encoder_destroy (enc->state);
+        enc->state = NULL;
+      }
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
+    default:
+      break;
+  }
+
+  return res;
+}
diff --git a/ext/opus/gstopusenc.h b/ext/opus/gstopusenc.h
new file mode 100644
index 000000000..5cb54598a
--- /dev/null
+++ b/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__ */