opus: remove Opus encoder/decoder, moved to -base

https://bugzilla.gnome.org/show_bug.cgi?id=756282

14 files changed, 2 insertions, 2915 deletions

diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
index 57dac6418..710ced52a 100644
--- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
@@ -73,9 +73,6 @@
     <xi:include href="xml/element-ofa.xml" />
     <xi:include href="xml/element-openalsrc.xml" />
     <xi:include href="xml/element-openalsink.xml" />
-    <xi:include href="xml/element-opusdec.xml" />
-    <xi:include href="xml/element-opusenc.xml" />
-    <xi:include href="xml/element-opusparse.xml" />
     <xi:include href="xml/element-pcapparse.xml" />
     <xi:include href="xml/element-pinch.xml" />
     <xi:include href="xml/element-pyramidsegment.xml" />
@@ -141,7 +138,6 @@
     <xi:include href="xml/plugin-ofa.xml" />
     <xi:include href="xml/plugin-openal.xml" />
     <xi:include href="xml/plugin-opencv.xml" />
-    <xi:include href="xml/plugin-opus.xml" />
     <xi:include href="xml/plugin-pcapparse.xml" />
     <xi:include href="xml/plugin-rawparse.xml" />
     <xi:include href="xml/plugin-rfbsrc.xml" />
diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt
index 3b47c167f..2c1365b1d 100644
--- a/docs/plugins/gst-plugins-bad-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt
@@ -1102,45 +1102,6 @@ gst_opencv_text_overlay_get_type
 gst_opencv_text_overlay_plugin_init
 </SECTION>
 
-<FILE>element-opusdec</FILE>
-<TITLE>opusdec</TITLE>
-GstOpusDec
-<SUBSECTION Standard>
-GstOpusDecClass
-gst_opus_dec_get_type
-GST_TYPE_OPUS_DEC
-GST_OPUS_DEC
-GST_OPUS_DEC_CLASS
-GST_IS_OPUS_DEC
-GST_IS_OPUS_DEC_CLASS
-</SECTION>
-
-<FILE>element-opusenc</FILE>
-<TITLE>opusenc</TITLE>
-GstOpusEnc
-<SUBSECTION Standard>
-GstOpusEncClass
-gst_opus_enc_get_type
-GST_TYPE_OPUS_ENC
-GST_OPUS_ENC
-GST_OPUS_ENC_CLASS
-GST_IS_OPUS_ENC
-GST_IS_OPUS_ENC_CLASS
-</SECTION>
-
-<FILE>element-opusparse</FILE>
-<TITLE>opusparse</TITLE>
-GstOpusParse
-<SUBSECTION Standard>
-GstOpusParseClass
-gst_opus_parse_get_type
-GST_TYPE_OPUS_PARSE
-GST_OPUS_PARSE
-GST_OPUS_PARSE_CLASS
-GST_IS_OPUS_PARSE
-GST_IS_OPUS_PARSE_CLASS
-</SECTION>
-
 <SECTION>
 <FILE>element-pcapparse</FILE>
 <TITLE>pcapparse</TITLE>
diff --git a/docs/plugins/inspect/plugin-opus.xml b/docs/plugins/inspect/plugin-opus.xml
deleted file mode 100644
index 7f9053a87..000000000
--- a/docs/plugins/inspect/plugin-opus.xml
+++ /dev/null
@@ -1,76 +0,0 @@
-<plugin>
-  <name>opus</name>
-  <description>OPUS plugin library</description>
-  <filename>../../ext/opus/.libs/libgstopus.so</filename>
-  <basename>libgstopus.so</basename>
-  <version>1.7.2.1</version>
-  <license>LGPL</license>
-  <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins git</package>
-  <origin>Unknown package origin</origin>
-  <elements>
-    <element>
-      <name>opusdec</name>
-      <longname>Opus audio decoder</longname>
-      <class>Codec/Decoder/Audio</class>
-      <description>decode opus streams to audio</description>
-      <author>Vincent Penquerc&apos;h &lt;vincent.penquerch@collabora.co.uk&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-opus, channel-mapping-family=(int)0; audio/x-opus, channel-mapping-family=(int)[ 1, 255 ], channels=(int)[ 1, 255 ], stream-count=(int)[ 1, 255 ], coupled-count=(int)[ 0, 255 ]</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 24000, 16000, 12000, 8000 }, channels=(int)[ 1, 8 ]</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>opusenc</name>
-      <longname>Opus audio encoder</longname>
-      <class>Codec/Encoder/Audio</class>
-      <description>Encodes audio in Opus format</description>
-      <author>Vincent Penquerc&apos;h &lt;vincent.penquerch@collabora.co.uk&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)48000, channels=(int)[ 1, 8 ]; audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 8000, 12000, 16000, 24000 }, channels=(int)[ 1, 8 ]</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-opus</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>opusparse</name>
-      <longname>Opus audio parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>parses opus audio streams</description>
-      <author>Vincent Penquerc&apos;h &lt;vincent.penquerch@collabora.co.uk&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-opus</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-opus</details>
-        </caps>
-      </pads>
-    </element>
-  </elements>
-</plugin>
\ No newline at end of file
diff --git a/ext/opus/Makefile.am b/ext/opus/Makefile.am
index aa0d4388e..4630715ba 100644
--- a/ext/opus/Makefile.am
+++ b/ext/opus/Makefile.am
@@ -1,6 +1,6 @@
 plugin_LTLIBRARIES = libgstopus.la
 
-libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c
+libgstopus_la_SOURCES = gstopus.c gstopusparse.c gstopusheader.c
 libgstopus_la_CFLAGS = \
         -DGST_USE_UNSTABLE_API \
 	$(GST_PLUGINS_BAD_CFLAGS) \
@@ -17,4 +17,4 @@ libgstopus_la_LIBADD = \
 libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
 libgstopus_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
 
-noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h
+noinst_HEADERS = gstopusparse.h gstopusheader.h
diff --git a/ext/opus/gstopus.c b/ext/opus/gstopus.c
index 63d50ef3e..0f9cc379e 100644
--- a/ext/opus/gstopus.c
+++ b/ext/opus/gstopus.c
@@ -21,8 +21,6 @@
 #include <config.h>
 #endif
 
-#include "gstopusdec.h"
-#include "gstopusenc.h"
 #include "gstopusparse.h"
 
 #include <gst/tag/tag.h>
@@ -30,15 +28,6 @@
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-
-  if (!gst_element_register (plugin, "opusenc", GST_RANK_PRIMARY,
-          GST_TYPE_OPUS_ENC))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "opusdec", GST_RANK_PRIMARY,
-          GST_TYPE_OPUS_DEC))
-    return FALSE;
-
   if (!gst_element_register (plugin, "opusparse", GST_RANK_NONE,
           GST_TYPE_OPUS_PARSE))
     return FALSE;
diff --git a/ext/opus/gstopuscommon.c b/ext/opus/gstopuscommon.c
deleted file mode 100644
index febccd85f..000000000
--- a/ext/opus/gstopuscommon.c
+++ /dev/null
@@ -1,111 +0,0 @@
-/* GStreamer
- * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include "gstopuscommon.h"
-
-/* http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9 */
-/* copy of the same structure in the vorbis plugin */
-const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
-  {                             /* Mono */
-      GST_AUDIO_CHANNEL_POSITION_MONO},
-  {                             /* Stereo */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
-  {                             /* Stereo + Centre */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
-  {                             /* Quadraphonic */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
-      },
-  {                             /* Stereo + Centre + rear stereo */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
-      },
-  {                             /* Full 5.1 Surround */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_LFE1,
-      },
-  {                             /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
-      GST_AUDIO_CHANNEL_POSITION_LFE1},
-  {                             /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
-        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
-      GST_AUDIO_CHANNEL_POSITION_LFE1},
-};
-
-const char *gst_opus_channel_names[] = {
-  "mono",
-  "front left",
-  "front right",
-  "rear center",
-  "rear left",
-  "rear right",
-  "lfe",
-  "front center",
-  "front left of center",
-  "front right of center",
-  "side left",
-  "side right",
-  "none"
-};
-
-void
-gst_opus_common_log_channel_mapping_table (GstElement * element,
-    GstDebugCategory * category, const char *msg, int n_channels,
-    const guint8 * table)
-{
-  int n;
-  GString *s;
-
-  if (gst_debug_category_get_threshold (category) < GST_LEVEL_INFO)
-    return;
-
-  s = g_string_new ("[ ");
-  for (n = 0; n < n_channels; ++n) {
-    g_string_append_printf (s, "%d ", table[n]);
-  }
-  g_string_append (s, "]");
-
-  GST_CAT_LEVEL_LOG (category, GST_LEVEL_INFO, element, "%s: %s", msg, s->str);
-  g_string_free (s, TRUE);
-}
diff --git a/ext/opus/gstopuscommon.h b/ext/opus/gstopuscommon.h
deleted file mode 100644
index 71771ae74..000000000
--- a/ext/opus/gstopuscommon.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/* GStreamer Opus Encoder
- * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-
-#ifndef __GST_OPUS_COMMON_H__
-#define __GST_OPUS_COMMON_H__
-
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-
-G_BEGIN_DECLS
-
-extern const GstAudioChannelPosition gst_opus_channel_positions[][8];
-extern const char *gst_opus_channel_names[];
-extern void gst_opus_common_log_channel_mapping_table (GstElement *element,
-    GstDebugCategory * category, const char *msg,
-    int n_channels, const guint8 *table);
-
-G_END_DECLS
-
-#endif /* __GST_OPUS_COMMON_H__ */
diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c
deleted file mode 100644
index 1470ea321..000000000
--- a/ext/opus/gstopusdec.c
+++ /dev/null
@@ -1,819 +0,0 @@
-/* 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>
- * Copyright (C) 2011-2012 Vincent Penquerc'h <vincent.penquerch@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., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, 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-1.0 -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 <math.h>
-#include <string.h>
-#include "gstopusheader.h"
-#include "gstopuscommon.h"
-#include "gstopusdec.h"
-#include <gst/pbutils/pbutils.h>
-
-GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
-#define GST_CAT_DEFAULT opusdec_debug
-
-static GstStaticPadTemplate opus_dec_src_factory =
-GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " GST_AUDIO_NE (S16) ", "
-        "layout = (string) interleaved, "
-        "rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
-        "channels = (int) [ 1, 8 ] ")
-    );
-
-static GstStaticPadTemplate opus_dec_sink_factory =
-    GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-opus, "
-        "channel-mapping-family = (int) 0; "
-        "audio/x-opus, "
-        "channel-mapping-family = (int) [1, 255], "
-        "channels = (int) [1, 255], "
-        "stream-count = (int) [1, 255], " "coupled-count = (int) [0, 255]")
-    );
-
-G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER);
-
-#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
-
-#define DEFAULT_USE_INBAND_FEC FALSE
-#define DEFAULT_APPLY_GAIN TRUE
-
-enum
-{
-  PROP_0,
-  PROP_USE_INBAND_FEC,
-  PROP_APPLY_GAIN
-};
-
-
-static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec,
-    GstBuffer * buf);
-static gboolean gst_opus_dec_start (GstAudioDecoder * dec);
-static gboolean gst_opus_dec_stop (GstAudioDecoder * dec);
-static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,
-    GstBuffer * buffer);
-static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,
-    GstCaps * caps);
-static void gst_opus_dec_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-static void gst_opus_dec_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-
-
-static void
-gst_opus_dec_class_init (GstOpusDecClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstAudioDecoderClass *adclass;
-  GstElementClass *element_class;
-
-  gobject_class = (GObjectClass *) klass;
-  adclass = (GstAudioDecoderClass *) klass;
-  element_class = (GstElementClass *) klass;
-
-  gobject_class->set_property = gst_opus_dec_set_property;
-  gobject_class->get_property = gst_opus_dec_get_property;
-
-  adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start);
-  adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop);
-  adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame);
-  adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format);
-
-  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_static_metadata (element_class, "Opus audio decoder",
-      "Codec/Decoder/Audio",
-      "decode opus streams to audio",
-      "Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");
-  g_object_class_install_property (gobject_class, PROP_USE_INBAND_FEC,
-      g_param_spec_boolean ("use-inband-fec", "Use in-band FEC",
-          "Use forward error correction if available (needs PLC enabled)",
-          DEFAULT_USE_INBAND_FEC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_APPLY_GAIN,
-      g_param_spec_boolean ("apply-gain", "Apply gain",
-          "Apply gain if any is specified in the header", DEFAULT_APPLY_GAIN,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
-      "opus decoding element");
-}
-
-static void
-gst_opus_dec_reset (GstOpusDec * dec)
-{
-  dec->packetno = 0;
-  if (dec->state) {
-    opus_multistream_decoder_destroy (dec->state);
-    dec->state = NULL;
-  }
-
-  gst_buffer_replace (&dec->streamheader, NULL);
-  gst_buffer_replace (&dec->vorbiscomment, NULL);
-  gst_buffer_replace (&dec->last_buffer, NULL);
-  dec->primed = FALSE;
-
-  dec->pre_skip = 0;
-  dec->r128_gain = 0;
-  dec->sample_rate = 0;
-  dec->n_channels = 0;
-  dec->leftover_plc_duration = 0;
-}
-
-static void
-gst_opus_dec_init (GstOpusDec * dec)
-{
-  dec->use_inband_fec = FALSE;
-  dec->apply_gain = DEFAULT_APPLY_GAIN;
-
-  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
-  gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
-      (dec), TRUE);
-  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
-
-  gst_opus_dec_reset (dec);
-}
-
-static gboolean
-gst_opus_dec_start (GstAudioDecoder * dec)
-{
-  GstOpusDec *odec = GST_OPUS_DEC (dec);
-
-  gst_opus_dec_reset (odec);
-
-  /* we know about concealment */
-  gst_audio_decoder_set_plc_aware (dec, TRUE);
-
-  if (odec->use_inband_fec) {
-    /* opusdec outputs samples directly from an input buffer, except if
-     * FEC is on, in which case it buffers one buffer in case one buffer
-     * goes missing.
-     */
-    gst_audio_decoder_set_latency (dec, 120 * GST_MSECOND, 120 * GST_MSECOND);
-  }
-
-  return TRUE;
-}
-
-static gboolean
-gst_opus_dec_stop (GstAudioDecoder * dec)
-{
-  GstOpusDec *odec = GST_OPUS_DEC (dec);
-
-  gst_opus_dec_reset (odec);
-
-  return TRUE;
-}
-
-static double
-gst_opus_dec_get_r128_gain (gint16 r128_gain)
-{
-  return r128_gain / (double) (1 << 8);
-}
-
-static double
-gst_opus_dec_get_r128_volume (gint16 r128_gain)
-{
-  return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));
-}
-
-static void
-gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
-{
-  GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
-  GstStructure *s;
-  GstAudioInfo info;
-
-  if (caps) {
-    gint rate, channels;
-
-    caps = gst_caps_truncate (caps);
-    caps = gst_caps_make_writable (caps);
-    s = gst_caps_get_structure (caps, 0);
-
-    if (gst_structure_has_field (s, "rate"))
-      gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);
-    else
-      gst_structure_set (s, "rate", G_TYPE_INT, dec->sample_rate, NULL);
-    gst_structure_get_int (s, "rate", &rate);
-    dec->sample_rate = rate;
-
-    if (gst_structure_has_field (s, "channels"))
-      gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
-    else
-      gst_structure_set (s, "channels", G_TYPE_INT, dec->n_channels, NULL);
-    gst_structure_get_int (s, "channels", &channels);
-    dec->n_channels = channels;
-
-    gst_caps_unref (caps);
-  }
-
-  if (dec->n_channels == 0) {
-    GST_DEBUG_OBJECT (dec, "Using a default of 2 channels");
-    dec->n_channels = 2;
-    pos = NULL;
-  }
-
-  if (dec->sample_rate == 0) {
-    GST_DEBUG_OBJECT (dec, "Using a default of 48kHz sample rate");
-    dec->sample_rate = 48000;
-  }
-
-  GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
-      dec->sample_rate);
-
-  /* pass valid order to audio info */
-  if (pos) {
-    memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
-    gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
-  }
-
-  /* set up source format */
-  gst_audio_info_init (&info);
-  gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
-      dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
-  gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
-
-  /* but we still need the opus order for later reordering */
-  if (pos) {
-    memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
-  } else {
-    dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
-  }
-
-  dec->info = info;
-}
-
-static GstFlowReturn
-gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
-{
-  GstAudioChannelPosition pos[64];
-  const GstAudioChannelPosition *posn = NULL;
-
-  if (!gst_opus_header_is_id_header (buf)) {
-    GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
-    return GST_FLOW_ERROR;
-  }
-
-  if (!gst_codec_utils_opus_parse_header (buf,
-          &dec->sample_rate,
-          &dec->n_channels,
-          &dec->channel_mapping_family,
-          &dec->n_streams,
-          &dec->n_stereo_streams,
-          dec->channel_mapping, &dec->pre_skip, &dec->r128_gain)) {
-    GST_ERROR_OBJECT (dec, "Failed to parse Opus ID header");
-    return GST_FLOW_ERROR;
-  }
-  dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
-
-  GST_INFO_OBJECT (dec,
-      "Found pre-skip of %u samples, R128 gain %d (volume %f)",
-      dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
-
-  if (dec->channel_mapping_family == 1) {
-    GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
-    switch (dec->n_channels) {
-      case 1:
-      case 2:
-        /* nothing */
-        break;
-      case 3:
-      case 4:
-      case 5:
-      case 6:
-      case 7:
-      case 8:
-        posn = gst_opus_channel_positions[dec->n_channels - 1];
-        break;
-      default:{
-        gint i;
-
-        GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
-            (NULL), ("Using NONE channel layout for more than 8 channels"));
-
-        for (i = 0; i < dec->n_channels; i++)
-          pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
-
-        posn = pos;
-      }
-    }
-  } else {
-    GST_INFO_OBJECT (dec, "Channel mapping family %d",
-        dec->channel_mapping_family);
-  }
-
-  gst_opus_dec_negotiate (dec, posn);
-
-  return GST_FLOW_OK;
-}
-
-
-static GstFlowReturn
-gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
-{
-  return GST_FLOW_OK;
-}
-
-static GstFlowReturn
-opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
-{
-  GstFlowReturn res = GST_FLOW_OK;
-  gsize size;
-  guint8 *data;
-  GstBuffer *outbuf, *bufd;
-  gint16 *out_data;
-  int n, err;
-  int samples;
-  unsigned int packet_size;
-  GstBuffer *buf;
-  GstMapInfo map, omap;
-  GstAudioClippingMeta *cmeta = NULL;
-
-  if (dec->state == NULL) {
-    /* If we did not get any headers, default to 2 channels */
-    if (dec->n_channels == 0) {
-      GST_INFO_OBJECT (dec, "No header, assuming single stream");
-      dec->n_channels = 2;
-      dec->sample_rate = 48000;
-      /* default stereo mapping */
-      dec->channel_mapping_family = 0;
-      dec->channel_mapping[0] = 0;
-      dec->channel_mapping[1] = 1;
-      dec->n_streams = 1;
-      dec->n_stereo_streams = 1;
-
-      gst_opus_dec_negotiate (dec, NULL);
-    }
-
-    GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
-        dec->n_channels, dec->sample_rate);
-#ifndef GST_DISABLE_GST_DEBUG
-    gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
-        "Mapping table", dec->n_channels, dec->channel_mapping);
-#endif
-
-    GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
-        dec->n_stereo_streams);
-    dec->state =
-        opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
-        dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
-    if (!dec->state || err != OPUS_OK)
-      goto creation_failed;
-  }
-
-  if (buffer) {
-    GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
-        gst_buffer_get_size (buffer));
-  } else {
-    GST_DEBUG_OBJECT (dec, "Received missing buffer");
-  }
-
-  /* if using in-band FEC, we introdude one extra frame's delay as we need
-     to potentially wait for next buffer to decode a missing buffer */
-  if (dec->use_inband_fec && !dec->primed) {
-    GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
-    gst_buffer_replace (&dec->last_buffer, buffer);
-    dec->primed = TRUE;
-    goto done;
-  }
-
-  /* That's the buffer we'll be sending to the opus decoder. */
-  buf = (dec->use_inband_fec
-      && gst_buffer_get_size (dec->last_buffer) >
-      0) ? dec->last_buffer : buffer;
-
-  /* That's the buffer we get duration from */
-  bufd = dec->use_inband_fec ? dec->last_buffer : buffer;
-
-  if (buf && gst_buffer_get_size (buf) > 0) {
-    gst_buffer_map (buf, &map, GST_MAP_READ);
-    data = map.data;
-    size = map.size;
-    GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
-  } else {
-    /* concealment data, pass NULL as the bits parameters */
-    GST_DEBUG_OBJECT (dec, "Using NULL buffer");
-    data = NULL;
-    size = 0;
-  }
-
-  if (gst_buffer_get_size (bufd) == 0) {
-    GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
-    GstClockTime aligned_missing_duration;
-    GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);
-
-    GST_DEBUG_OBJECT (dec,
-        "missing buffer, doing PLC duration %" GST_TIME_FORMAT
-        " plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
-        GST_TIME_ARGS (dec->leftover_plc_duration));
-
-    /* add the leftover PLC duration to that of the buffer */
-    missing_duration += dec->leftover_plc_duration;
-
-    /* align the combined buffer and leftover PLC duration to multiples
-     * of 2.5ms, rounding to nearest, and store excess duration for later */
-    aligned_missing_duration =
-        ((missing_duration +
-            opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;
-    dec->leftover_plc_duration = missing_duration - aligned_missing_duration;
-
-    /* Opus' PLC cannot operate with less than 2.5ms; skip PLC
-     * and accumulate the missing duration in the leftover_plc_duration
-     * for the next PLC attempt */
-    if (aligned_missing_duration < opus_plc_alignment) {
-      GST_DEBUG_OBJECT (dec,
-          "current duration %" GST_TIME_FORMAT
-          " of missing data not enough for PLC (minimum needed: %"
-          GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
-          GST_TIME_ARGS (opus_plc_alignment));
-      goto done;
-    }
-
-    /* convert the duration (in nanoseconds) to sample count */
-    samples =
-        gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
-        GST_SECOND);
-
-    GST_DEBUG_OBJECT (dec,
-        "calculated PLC frame length: %" GST_TIME_FORMAT
-        " num frame samples: %d new leftover: %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (aligned_missing_duration), samples,
-        GST_TIME_ARGS (dec->leftover_plc_duration));
-  } else {
-    /* use maximum size (120 ms) as the number of returned samples is
-       not constant over the stream. */
-    samples = 120 * dec->sample_rate / 1000;
-  }
-
-  packet_size = samples * dec->n_channels * 2;
-
-  outbuf =
-      gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
-      packet_size);
-  if (!outbuf) {
-    goto buffer_failed;
-  }
-
-  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
-  out_data = (gint16 *) omap.data;
-
-  if (dec->use_inband_fec) {
-    if (gst_buffer_get_size (dec->last_buffer) > 0) {
-      /* normal delayed decode */
-      GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
-      n = opus_multistream_decode (dec->state, data, size, out_data, samples,
-          0);
-    } else {
-      /* FEC reconstruction decode */
-      GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
-      n = opus_multistream_decode (dec->state, data, size, out_data, samples,
-          1);
-    }
-  } else {
-    /* normal decode */
-    GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
-    n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
-  }
-  gst_buffer_unmap (outbuf, &omap);
-  if (data != NULL)
-    gst_buffer_unmap (buf, &map);
-
-  if (n < 0) {
-    GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
-    gst_buffer_unref (outbuf);
-    return GST_FLOW_ERROR;
-  }
-  GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
-  gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
-
-  cmeta = gst_buffer_get_audio_clipping_meta (buf);
-
-  g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
-
-  /* Skip any samples that need skipping */
-  if (cmeta && cmeta->start) {
-    guint pre_skip = cmeta->start;
-    guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;
-    guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
-    guint scaled_skip = skip * 48000 / dec->sample_rate;
-
-    gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
-
-    GST_INFO_OBJECT (dec,
-        "Skipping %u samples at the beginning (%u at 48000 Hz)",
-        skip, scaled_skip);
-  }
-
-  if (cmeta && cmeta->end) {
-    guint post_skip = cmeta->end;
-    guint scaled_post_skip = post_skip * dec->sample_rate / 48000;
-    guint skip = scaled_post_skip > n ? n : scaled_post_skip;
-    guint scaled_skip = skip * 48000 / dec->sample_rate;
-    guint outsize = gst_buffer_get_size (outbuf);
-    guint skip_bytes = skip * 2 * dec->n_channels;
-
-    if (outsize > skip_bytes)
-      outsize -= skip_bytes;
-    else
-      outsize = 0;
-
-    gst_buffer_resize (outbuf, 0, outsize);
-
-    GST_INFO_OBJECT (dec,
-        "Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);
-  }
-
-  if (gst_buffer_get_size (outbuf) == 0) {
-    gst_buffer_unref (outbuf);
-    outbuf = NULL;
-  } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
-    gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
-        dec->n_channels, dec->opus_pos, dec->info.position);
-  }
-
-  /* Apply gain */
-  /* Would be better off leaving this to a volume element, as this is
-     a naive conversion that does too many int/float conversions.
-     However, we don't have control over the pipeline...
-     So make it optional if the user program wants to use a volume,
-     but do it by default so the correct volume goes out by default */
-  if (dec->apply_gain && outbuf && dec->r128_gain) {
-    gsize rsize;
-    unsigned int i, nsamples;
-    double volume = dec->r128_gain_volume;
-    gint16 *samples;
-
-    gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
-    samples = (gint16 *) omap.data;
-    rsize = omap.size;
-    GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
-    nsamples = rsize / 2;
-    for (i = 0; i < nsamples; ++i) {
-      int sample = (int) (samples[i] * volume + 0.5);
-      samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
-    }
-    gst_buffer_unmap (outbuf, &omap);
-  }
-
-  if (dec->use_inband_fec) {
-    gst_buffer_replace (&dec->last_buffer, buffer);
-  }
-
-  res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
-
-  if (res != GST_FLOW_OK)
-    GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
-
-done:
-  return res;
-
-creation_failed:
-  GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
-  return GST_FLOW_ERROR;
-
-buffer_failed:
-  GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size);
-  return GST_FLOW_ERROR;
-}
-
-static gboolean
-gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
-{
-  GstOpusDec *dec = GST_OPUS_DEC (bdec);
-  gboolean ret = TRUE;
-  GstStructure *s;
-  const GValue *streamheader;
-  GstCaps *old_caps;
-
-  GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
-
-  if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
-    if (gst_caps_is_equal (caps, old_caps)) {
-      gst_caps_unref (old_caps);
-      GST_DEBUG_OBJECT (dec, "caps didn't change");
-      goto done;
-    }
-
-    GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
-    gst_opus_dec_reset (dec);
-    gst_caps_unref (old_caps);
-  }
-
-  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, *vorbiscomment;
-    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 = gst_opus_dec_parse_header (dec, buf);
-      if (res != GST_FLOW_OK) {
-        ret = FALSE;
-        goto done;
-      }
-      gst_buffer_replace (&dec->streamheader, buf);
-    }
-
-    vorbiscomment = gst_value_array_get_value (streamheader, 1);
-    if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
-      buf = gst_value_get_buffer (vorbiscomment);
-      res = gst_opus_dec_parse_comments (dec, buf);
-      if (res != GST_FLOW_OK) {
-        ret = FALSE;
-        goto done;
-      }
-      gst_buffer_replace (&dec->vorbiscomment, buf);
-    }
-  } else {
-    const GstAudioChannelPosition *posn = NULL;
-
-    if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,
-            &dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,
-            &dec->n_stereo_streams, dec->channel_mapping)) {
-      ret = FALSE;
-      goto done;
-    }
-
-    if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)
-      posn = gst_opus_channel_positions[dec->n_channels - 1];
-
-    gst_opus_dec_negotiate (dec, posn);
-  }
-
-done:
-  return ret;
-}
-
-static gboolean
-memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
-{
-  gsize size1, size2;
-  gboolean res;
-  GstMapInfo map;
-
-  size1 = gst_buffer_get_size (buf1);
-  size2 = gst_buffer_get_size (buf2);
-
-  if (size1 != size2)
-    return FALSE;
-
-  gst_buffer_map (buf1, &map, GST_MAP_READ);
-  res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;
-  gst_buffer_unmap (buf1, &map);
-
-  return res;
-}
-
-static GstFlowReturn
-gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
-{
-  GstFlowReturn res;
-  GstOpusDec *dec;
-
-  /* no fancy draining */
-  if (G_UNLIKELY (!buf))
-    return GST_FLOW_OK;
-
-  dec = GST_OPUS_DEC (adec);
-  GST_LOG_OBJECT (dec,
-      "Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
-      GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-
-  /* If we have the streamheader and vorbiscomment from the caps already
-   * ignore them here */
-  if (dec->streamheader && dec->vorbiscomment) {
-    if (memcmp_buffers (dec->streamheader, buf)) {
-      GST_DEBUG_OBJECT (dec, "found streamheader");
-      gst_audio_decoder_finish_frame (adec, NULL, 1);
-      res = GST_FLOW_OK;
-    } else if (memcmp_buffers (dec->vorbiscomment, buf)) {
-      GST_DEBUG_OBJECT (dec, "found vorbiscomments");
-      gst_audio_decoder_finish_frame (adec, NULL, 1);
-      res = GST_FLOW_OK;
-    } else {
-      res = opus_dec_chain_parse_data (dec, buf);
-    }
-  } else {
-    /* Otherwise fall back to packet counting and assume that the
-     * first two packets might be the headers, checking magic. */
-    switch (dec->packetno) {
-      case 0:
-        if (gst_opus_header_is_header (buf, "OpusHead", 8)) {
-          GST_DEBUG_OBJECT (dec, "found streamheader");
-          res = gst_opus_dec_parse_header (dec, buf);
-          gst_audio_decoder_finish_frame (adec, NULL, 1);
-        } else {
-          res = opus_dec_chain_parse_data (dec, buf);
-        }
-        break;
-      case 1:
-        if (gst_opus_header_is_header (buf, "OpusTags", 8)) {
-          GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
-          res = gst_opus_dec_parse_comments (dec, buf);
-          gst_audio_decoder_finish_frame (adec, NULL, 1);
-        } else {
-          res = opus_dec_chain_parse_data (dec, buf);
-        }
-        break;
-      default:
-      {
-        res = opus_dec_chain_parse_data (dec, buf);
-        break;
-      }
-    }
-  }
-
-  dec->packetno++;
-
-  return res;
-}
-
-static void
-gst_opus_dec_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
-{
-  GstOpusDec *dec = GST_OPUS_DEC (object);
-
-  switch (prop_id) {
-    case PROP_USE_INBAND_FEC:
-      g_value_set_boolean (value, dec->use_inband_fec);
-      break;
-    case PROP_APPLY_GAIN:
-      g_value_set_boolean (value, dec->apply_gain);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_opus_dec_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstOpusDec *dec = GST_OPUS_DEC (object);
-
-  switch (prop_id) {
-    case PROP_USE_INBAND_FEC:
-      dec->use_inband_fec = g_value_get_boolean (value);
-      break;
-    case PROP_APPLY_GAIN:
-      dec->apply_gain = g_value_get_boolean (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h
deleted file mode 100644
index df52cfb6f..000000000
--- a/ext/opus/gstopusdec.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Copyright (C) <2011-2012> Vincent Penquerc'h <vincent.penquerch@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., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __GST_OPUS_DEC_H__
-#define __GST_OPUS_DEC_H__
-
-#include <gst/gst.h>
-#include <gst/audio/gstaudiodecoder.h>
-#include <opus_multistream.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 {
-  GstAudioDecoder       element;
-
-  OpusMSDecoder        *state;
-
-  guint64               packetno;
-
-  GstBuffer            *streamheader;
-  GstBuffer            *vorbiscomment;
-
-  guint32 sample_rate;
-  guint8 n_channels;
-  guint16 pre_skip;
-  gint16 r128_gain;
-
-  GstAudioChannelPosition opus_pos[64];
-  GstAudioInfo info;
-
-  guint8 n_streams;
-  guint8 n_stereo_streams;
-  guint8 channel_mapping_family;
-  guint8 channel_mapping[256];
-
-  gboolean apply_gain;
-  double r128_gain_volume;
-
-  gboolean use_inband_fec;
-  GstBuffer *last_buffer;
-  gboolean primed;
-
-  guint64 leftover_plc_duration;
-};
-
-struct _GstOpusDecClass {
-  GstAudioDecoderClass 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
deleted file mode 100644
index 7737bf575..000000000
--- a/ext/opus/gstopusenc.c
+++ /dev/null
@@ -1,1282 +0,0 @@
-/* GStreamer Opus Encoder
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Copyright (C) <2011> Vincent Penquerc'h <vincent.penquerch@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., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, 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-1.0 -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.h>
-
-#include <gst/gsttagsetter.h>
-#include <gst/audio/audio.h>
-#include <gst/pbutils/pbutils.h>
-#include <gst/tag/tag.h>
-#include <gst/glib-compat-private.h>
-#include "gstopusheader.h"
-#include "gstopuscommon.h"
-#include "gstopusenc.h"
-
-GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
-#define GST_CAT_DEFAULT opusenc_debug
-
-/* Some arbitrary bounds beyond which it really doesn't make sense.
-   The spec mentions 6 kb/s to 510 kb/s, so 4000 and 650000 ought to be
-   safe as property bounds. */
-#define LOWEST_BITRATE 4000
-#define HIGHEST_BITRATE 650000
-
-#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_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;
-}
-
-#define GST_OPUS_ENC_TYPE_FRAME_SIZE (gst_opus_enc_frame_size_get_type())
-static GType
-gst_opus_enc_frame_size_get_type (void)
-{
-  static const GEnumValue values[] = {
-    {2, "2.5", "2.5"},
-    {5, "5", "5"},
-    {10, "10", "10"},
-    {20, "20", "20"},
-    {40, "40", "40"},
-    {60, "60", "60"},
-    {0, NULL, NULL}
-  };
-  static volatile GType id = 0;
-
-  if (g_once_init_enter ((gsize *) & id)) {
-    GType _id;
-
-    _id = g_enum_register_static ("GstOpusEncFrameSize", values);
-
-    g_once_init_leave ((gsize *) & id, _id);
-  }
-
-  return id;
-}
-
-#define GST_OPUS_ENC_TYPE_AUDIO_TYPE (gst_opus_enc_audio_type_get_type())
-static GType
-gst_opus_enc_audio_type_get_type (void)
-{
-  static const GEnumValue values[] = {
-    {OPUS_APPLICATION_AUDIO, "Generic audio", "generic"},
-    {OPUS_APPLICATION_VOIP, "Voice", "voice"},
-    {0, NULL, NULL}
-  };
-  static volatile GType id = 0;
-
-  if (g_once_init_enter ((gsize *) & id)) {
-    GType _id;
-
-    _id = g_enum_register_static ("GstOpusEncAudioType", values);
-
-    g_once_init_leave ((gsize *) & id, _id);
-  }
-
-  return id;
-}
-
-#define GST_OPUS_ENC_TYPE_BITRATE_TYPE (gst_opus_enc_bitrate_type_get_type())
-static GType
-gst_opus_enc_bitrate_type_get_type (void)
-{
-  static const GEnumValue values[] = {
-    {BITRATE_TYPE_CBR, "CBR", "cbr"},
-    {BITRATE_TYPE_VBR, "VBR", "vbr"},
-    {BITRATE_TYPE_CONSTRAINED_VBR, "Constrained VBR", "constrained-vbr"},
-    {0, NULL, NULL}
-  };
-  static volatile GType id = 0;
-
-  if (g_once_init_enter ((gsize *) & id)) {
-    GType _id;
-
-    _id = g_enum_register_static ("GstOpusEncBitrateType", values);
-
-    g_once_init_leave ((gsize *) & id, _id);
-  }
-
-  return id;
-}
-
-#define FORMAT_STR GST_AUDIO_NE(S16)
-static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " FORMAT_STR ", "
-        "layout = (string) interleaved, "
-        "rate = (int) 48000, "
-        "channels = (int) [ 1, 8 ]; "
-        "audio/x-raw, "
-        "format = (string) " FORMAT_STR ", "
-        "layout = (string) interleaved, "
-        "rate = (int) { 8000, 12000, 16000, 24000 }, "
-        "channels = (int) [ 1, 8 ] ")
-    );
-
-static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-opus")
-    );
-
-#define DEFAULT_AUDIO           TRUE
-#define DEFAULT_AUDIO_TYPE      OPUS_APPLICATION_AUDIO
-#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_BITRATE_TYPE    BITRATE_TYPE_CBR
-#define DEFAULT_COMPLEXITY      10
-#define DEFAULT_INBAND_FEC      FALSE
-#define DEFAULT_DTX             FALSE
-#define DEFAULT_PACKET_LOSS_PERCENT 0
-#define DEFAULT_MAX_PAYLOAD_SIZE 4000
-
-enum
-{
-  PROP_0,
-  PROP_AUDIO,
-  PROP_AUDIO_TYPE,
-  PROP_BITRATE,
-  PROP_BANDWIDTH,
-  PROP_FRAME_SIZE,
-  PROP_CBR,
-  PROP_CONSTRAINED_VBR,
-  PROP_BITRATE_TYPE,
-  PROP_COMPLEXITY,
-  PROP_INBAND_FEC,
-  PROP_DTX,
-  PROP_PACKET_LOSS_PERCENT,
-  PROP_MAX_PAYLOAD_SIZE
-};
-
-static void gst_opus_enc_finalize (GObject * object);
-
-static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc,
-    GstEvent * event);
-static GstCaps *gst_opus_enc_sink_getcaps (GstAudioEncoder * benc,
-    GstCaps * filter);
-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 void gst_opus_enc_set_tags (GstOpusEnc * enc);
-static gboolean gst_opus_enc_start (GstAudioEncoder * benc);
-static gboolean gst_opus_enc_stop (GstAudioEncoder * benc);
-static gboolean gst_opus_enc_set_format (GstAudioEncoder * benc,
-    GstAudioInfo * info);
-static GstFlowReturn gst_opus_enc_handle_frame (GstAudioEncoder * benc,
-    GstBuffer * buf);
-static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
-
-static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer);
-
-#define gst_opus_enc_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstOpusEnc, gst_opus_enc, GST_TYPE_AUDIO_ENCODER,
-    G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
-    G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
-
-static void
-gst_opus_enc_set_tags (GstOpusEnc * enc)
-{
-  GstTagList *taglist;
-
-  /* create a taglist and add a bitrate tag to it */
-  taglist = gst_tag_list_new_empty ();
-  gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
-      GST_TAG_BITRATE, enc->bitrate, NULL);
-
-  gst_audio_encoder_merge_tags (GST_AUDIO_ENCODER (enc), taglist,
-      GST_TAG_MERGE_REPLACE);
-
-  gst_tag_list_unref (taglist);
-}
-
-static void
-gst_opus_enc_class_init (GstOpusEncClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstAudioEncoderClass *base_class;
-  GstElementClass *gstelement_class;
-
-  gobject_class = (GObjectClass *) klass;
-  base_class = (GstAudioEncoderClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-
-  gobject_class->set_property = gst_opus_enc_set_property;
-  gobject_class->get_property = gst_opus_enc_get_property;
-
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&src_factory));
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&sink_factory));
-  gst_element_class_set_static_metadata (gstelement_class, "Opus audio encoder",
-      "Codec/Encoder/Audio",
-      "Encodes audio in Opus format",
-      "Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");
-
-  base_class->start = GST_DEBUG_FUNCPTR (gst_opus_enc_start);
-  base_class->stop = GST_DEBUG_FUNCPTR (gst_opus_enc_stop);
-  base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format);
-  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame);
-  base_class->sink_event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event);
-  base_class->getcaps = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps);
-
-  g_object_class_install_property (gobject_class, PROP_AUDIO,
-      g_param_spec_boolean ("audio",
-          "Audio or voice",
-          "Audio or voice (DEPRECATED: use audio-type)", DEFAULT_AUDIO,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
-  g_object_class_install_property (gobject_class, PROP_AUDIO_TYPE,
-      g_param_spec_enum ("audio-type", "What type of audio to optimize for",
-          "What type of audio to optimize for", GST_OPUS_ENC_TYPE_AUDIO_TYPE,
-          DEFAULT_AUDIO_TYPE, 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).", LOWEST_BITRATE,
-          HIGHEST_BITRATE, DEFAULT_BITRATE,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
-          GST_PARAM_MUTABLE_PLAYING));
-  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 |
-          GST_PARAM_MUTABLE_PLAYING));
-  g_object_class_install_property (gobject_class, PROP_FRAME_SIZE,
-      g_param_spec_enum ("frame-size", "Frame Size",
-          "The duration of an audio frame, in ms", GST_OPUS_ENC_TYPE_FRAME_SIZE,
-          DEFAULT_FRAMESIZE,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
-          GST_PARAM_MUTABLE_PLAYING));
-  g_object_class_install_property (gobject_class, PROP_CBR,
-      g_param_spec_boolean ("cbr", "Constant bit rate",
-          "Constant bit rate (DEPRECATED: use bitrate-type)", DEFAULT_CBR,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_PLAYING
-          | G_PARAM_DEPRECATED));
-  g_object_class_install_property (gobject_class, PROP_CONSTRAINED_VBR,
-      g_param_spec_boolean ("constrained-vbr", "Constrained VBR",
-          "Constrained VBR (DEPRECATED: use bitrate-type)",
-          DEFAULT_CONSTRAINED_VBR,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_PLAYING
-          | G_PARAM_DEPRECATED));
-  g_object_class_install_property (gobject_class, PROP_BITRATE_TYPE,
-      g_param_spec_enum ("bitrate-type", "Bitrate type", "Bitrate type",
-          GST_OPUS_ENC_TYPE_BITRATE_TYPE, DEFAULT_BITRATE_TYPE,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
-          GST_PARAM_MUTABLE_PLAYING));
-  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 |
-          GST_PARAM_MUTABLE_PLAYING));
-  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 |
-          GST_PARAM_MUTABLE_PLAYING));
-  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 |
-          GST_PARAM_MUTABLE_PLAYING));
-  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 |
-          GST_PARAM_MUTABLE_PLAYING));
-  g_object_class_install_property (G_OBJECT_CLASS (klass),
-      PROP_MAX_PAYLOAD_SIZE, g_param_spec_uint ("max-payload-size",
-          "Max payload size", "Maximum payload size in bytes", 2, 4000,
-          DEFAULT_MAX_PAYLOAD_SIZE,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
-          GST_PARAM_MUTABLE_PLAYING));
-
-  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize);
-
-  GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder");
-}
-
-static void
-gst_opus_enc_finalize (GObject * object)
-{
-  GstOpusEnc *enc;
-
-  enc = GST_OPUS_ENC (object);
-
-  g_mutex_clear (&enc->property_lock);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_opus_enc_init (GstOpusEnc * enc)
-{
-  GST_DEBUG_OBJECT (enc, "init");
-
-  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
-
-  g_mutex_init (&enc->property_lock);
-
-  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->bitrate_type = DEFAULT_BITRATE_TYPE;
-  enc->complexity = DEFAULT_COMPLEXITY;
-  enc->inband_fec = DEFAULT_INBAND_FEC;
-  enc->dtx = DEFAULT_DTX;
-  enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT;
-  enc->max_payload_size = DEFAULT_MAX_PAYLOAD_SIZE;
-  enc->audio_type = DEFAULT_AUDIO_TYPE;
-}
-
-static gboolean
-gst_opus_enc_start (GstAudioEncoder * benc)
-{
-  GstOpusEnc *enc = GST_OPUS_ENC (benc);
-
-  GST_DEBUG_OBJECT (enc, "start");
-  enc->encoded_samples = 0;
-  enc->consumed_samples = 0;
-
-  return TRUE;
-}
-
-static gboolean
-gst_opus_enc_stop (GstAudioEncoder * benc)
-{
-  GstOpusEnc *enc = GST_OPUS_ENC (benc);
-
-  GST_DEBUG_OBJECT (enc, "stop");
-  if (enc->state) {
-    opus_multistream_encoder_destroy (enc->state);
-    enc->state = NULL;
-  }
-  gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
-
-  return TRUE;
-}
-
-static gint64
-gst_opus_enc_get_latency (GstOpusEnc * enc)
-{
-  gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
-      enc->sample_rate);
-  GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
-  return latency;
-}
-
-static void
-gst_opus_enc_setup_base_class (GstOpusEnc * enc, GstAudioEncoder * benc)
-{
-  gst_audio_encoder_set_latency (benc,
-      gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc));
-  gst_audio_encoder_set_frame_samples_min (benc, enc->frame_samples);
-  gst_audio_encoder_set_frame_samples_max (benc, enc->frame_samples);
-  gst_audio_encoder_set_frame_max (benc, 1);
-}
-
-static gint
-gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
-{
-  gint frame_samples = 0;
-  switch (enc->frame_size) {
-    case 2:
-      frame_samples = enc->sample_rate / 400;
-      break;
-    case 5:
-      frame_samples = enc->sample_rate / 200;
-      break;
-    case 10:
-      frame_samples = enc->sample_rate / 100;
-      break;
-    case 20:
-      frame_samples = enc->sample_rate / 50;
-      break;
-    case 40:
-      frame_samples = enc->sample_rate / 25;
-      break;
-    case 60:
-      frame_samples = 3 * enc->sample_rate / 50;
-      break;
-    default:
-      GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size);
-      frame_samples = 0;
-      break;
-  }
-  return frame_samples;
-}
-
-static void
-gst_opus_enc_setup_trivial_mapping (GstOpusEnc * enc, guint8 mapping[256])
-{
-  int n;
-
-  for (n = 0; n < 255; ++n)
-    mapping[n] = n;
-}
-
-static int
-gst_opus_enc_find_channel_position (GstOpusEnc * enc, const GstAudioInfo * info,
-    GstAudioChannelPosition position)
-{
-  int n;
-  for (n = 0; n < enc->n_channels; ++n) {
-    if (GST_AUDIO_INFO_POSITION (info, n) == position) {
-      return n;
-    }
-  }
-  return -1;
-}
-
-static int
-gst_opus_enc_find_channel_position_in_vorbis_order (GstOpusEnc * enc,
-    GstAudioChannelPosition position)
-{
-  int c;
-
-  for (c = 0; c < enc->n_channels; ++c) {
-    if (gst_opus_channel_positions[enc->n_channels - 1][c] == position) {
-      GST_INFO_OBJECT (enc,
-          "Channel position %s maps to index %d in Vorbis order",
-          gst_opus_channel_names[position], c);
-      return c;
-    }
-  }
-  GST_WARNING_OBJECT (enc,
-      "Channel position %s is not representable in Vorbis order",
-      gst_opus_channel_names[position]);
-  return -1;
-}
-
-static void
-gst_opus_enc_setup_channel_mappings (GstOpusEnc * enc,
-    const GstAudioInfo * info)
-{
-#define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
-
-  int n;
-
-  GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
-      enc->n_channels);
-
-  /* Start by setting up a default trivial mapping */
-  enc->n_stereo_streams = 0;
-  gst_opus_enc_setup_trivial_mapping (enc, enc->encoding_channel_mapping);
-  gst_opus_enc_setup_trivial_mapping (enc, enc->decoding_channel_mapping);
-
-  /* For one channel, use the basic RTP mapping */
-  if (enc->n_channels == 1) {
-    GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
-    enc->channel_mapping_family = 0;
-    /* implicit mapping for family 0 */
-    return;
-  }
-
-  /* For two channels, use the basic RTP mapping if the channels are
-     mapped as left/right. */
-  if (enc->n_channels == 2) {
-    GST_INFO_OBJECT (enc, "Stereo, trivial RTP mapping");
-    enc->channel_mapping_family = 0;
-    enc->n_stereo_streams = 1;
-    /* implicit mapping for family 0 */
-    return;
-  }
-
-  /* For channels between 3 and 8, we use the Vorbis mapping if we can
-     find a permutation that matches it. Mono and stereo will have been taken
-     care of earlier, but this code also handles it. There are two mappings.
-     One maps the input channels to an ordering which has the natural pairs
-     first so they can benefit from the Opus stereo channel coupling, and the
-     other maps this ordering to the Vorbis ordering. */
-  if (enc->n_channels >= 3 && enc->n_channels <= 8) {
-    int c0, c1, c0v, c1v;
-    int mapped;
-    gboolean positions_done[256];
-    static const GstAudioChannelPosition pairs[][2] = {
-      {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-          GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
-      {GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-          GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
-      {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
-          GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
-      {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
-          GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
-      {GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
-          GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
-      {GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-          GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
-    };
-    size_t pair;
-
-    GST_DEBUG_OBJECT (enc,
-        "In range for the Vorbis mapping, building channel mapping tables");
-
-    enc->n_stereo_streams = 0;
-    mapped = 0;
-    for (n = 0; n < 256; ++n)
-      positions_done[n] = FALSE;
-
-    /* First, find any natural pairs, and move them to the front */
-    for (pair = 0; pair < G_N_ELEMENTS (pairs); ++pair) {
-      GstAudioChannelPosition p0 = pairs[pair][0];
-      GstAudioChannelPosition p1 = pairs[pair][1];
-      c0 = gst_opus_enc_find_channel_position (enc, info, p0);
-      c1 = gst_opus_enc_find_channel_position (enc, info, p1);
-      if (c0 >= 0 && c1 >= 0) {
-        /* We found a natural pair */
-        GST_DEBUG_OBJECT (enc, "Natural pair '%s/%s' found at %d %d",
-            gst_opus_channel_names[p0], gst_opus_channel_names[p1], c0, c1);
-        /* Find where they map in Vorbis order */
-        c0v = gst_opus_enc_find_channel_position_in_vorbis_order (enc, p0);
-        c1v = gst_opus_enc_find_channel_position_in_vorbis_order (enc, p1);
-        if (c0v < 0 || c1v < 0) {
-          GST_WARNING_OBJECT (enc,
-              "Cannot map channel positions to Vorbis order, using unknown mapping");
-          enc->channel_mapping_family = 255;
-          enc->n_stereo_streams = 0;
-          return;
-        }
-
-        enc->encoding_channel_mapping[mapped] = c0;
-        enc->encoding_channel_mapping[mapped + 1] = c1;
-        enc->decoding_channel_mapping[c0v] = mapped;
-        enc->decoding_channel_mapping[c1v] = mapped + 1;
-        enc->n_stereo_streams++;
-        mapped += 2;
-        positions_done[p0] = positions_done[p1] = TRUE;
-      }
-    }
-
-    /* Now add all other input channels as mono streams */
-    for (n = 0; n < enc->n_channels; ++n) {
-      GstAudioChannelPosition position = GST_AUDIO_INFO_POSITION (info, n);
-
-      /* if we already mapped it while searching for pairs, nothing else
-         needs to be done */
-      if (!positions_done[position]) {
-        int cv;
-        GST_DEBUG_OBJECT (enc, "Channel position %s is not mapped yet, adding",
-            gst_opus_channel_names[position]);
-        cv = gst_opus_enc_find_channel_position_in_vorbis_order (enc, position);
-        if (cv < 0)
-          g_assert_not_reached ();
-        enc->encoding_channel_mapping[mapped] = n;
-        enc->decoding_channel_mapping[cv] = mapped;
-        mapped++;
-      }
-    }
-
-#ifndef GST_DISABLE_GST_DEBUG
-    GST_INFO_OBJECT (enc,
-        "Mapping tables built: %d channels, %d stereo streams", enc->n_channels,
-        enc->n_stereo_streams);
-    gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
-        "Encoding mapping table", enc->n_channels,
-        enc->encoding_channel_mapping);
-    gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
-        "Decoding mapping table", enc->n_channels,
-        enc->decoding_channel_mapping);
-#endif
-
-    enc->channel_mapping_family = 1;
-    return;
-  }
-
-  /* More than 8 channels, if future mappings are added for those */
-
-  /* For other cases, we use undefined, with the default trivial mapping
-     and all mono streams */
-  GST_WARNING_OBJECT (enc, "Unknown mapping");
-  enc->channel_mapping_family = 255;
-  enc->n_stereo_streams = 0;
-
-#undef MAPS
-}
-
-static gboolean
-gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
-{
-  GstOpusEnc *enc;
-
-  enc = GST_OPUS_ENC (benc);
-
-  g_mutex_lock (&enc->property_lock);
-
-  enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
-  enc->sample_rate = GST_AUDIO_INFO_RATE (info);
-  gst_opus_enc_setup_channel_mappings (enc, info);
-  GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
-      enc->sample_rate);
-
-  /* handle reconfigure */
-  if (enc->state) {
-    opus_multistream_encoder_destroy (enc->state);
-    enc->state = NULL;
-  }
-  if (!gst_opus_enc_setup (enc)) {
-    g_mutex_unlock (&enc->property_lock);
-    return FALSE;
-  }
-
-  /* update the tags */
-  gst_opus_enc_set_tags (enc);
-
-  enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
-
-  /* feedback to base class */
-  gst_opus_enc_setup_base_class (enc, benc);
-
-  g_mutex_unlock (&enc->property_lock);
-
-  return TRUE;
-}
-
-static gboolean
-gst_opus_enc_setup (GstOpusEnc * enc)
-{
-  int error = OPUS_OK;
-  GstCaps *caps;
-  gboolean ret;
-  gint32 lookahead;
-  const GstTagList *tags;
-  GstTagList *empty_tags = NULL;
-  GstBuffer *header, *comments;
-
-#ifndef GST_DISABLE_GST_DEBUG
-  GST_DEBUG_OBJECT (enc,
-      "setup: %d Hz, %d channels, %d stereo streams, family %d",
-      enc->sample_rate, enc->n_channels, enc->n_stereo_streams,
-      enc->channel_mapping_family);
-  GST_INFO_OBJECT (enc, "Mapping tables built: %d channels, %d stereo streams",
-      enc->n_channels, enc->n_stereo_streams);
-  gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
-      "Encoding mapping table", enc->n_channels, enc->encoding_channel_mapping);
-  gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
-      "Decoding mapping table", enc->n_channels, enc->decoding_channel_mapping);
-#endif
-
-  enc->state = opus_multistream_encoder_create (enc->sample_rate,
-      enc->n_channels, enc->n_channels - enc->n_stereo_streams,
-      enc->n_stereo_streams, enc->encoding_channel_mapping,
-      enc->audio_type, &error);
-  if (!enc->state || error != OPUS_OK)
-    goto encoder_creation_failed;
-
-  opus_multistream_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
-  opus_multistream_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth),
-      0);
-  opus_multistream_encoder_ctl (enc->state,
-      OPUS_SET_VBR (enc->bitrate_type != BITRATE_TYPE_CBR), 0);
-  opus_multistream_encoder_ctl (enc->state,
-      OPUS_SET_VBR_CONSTRAINT (enc->bitrate_type ==
-          BITRATE_TYPE_CONSTRAINED_VBR), 0);
-  opus_multistream_encoder_ctl (enc->state,
-      OPUS_SET_COMPLEXITY (enc->complexity), 0);
-  opus_multistream_encoder_ctl (enc->state,
-      OPUS_SET_INBAND_FEC (enc->inband_fec), 0);
-  opus_multistream_encoder_ctl (enc->state, OPUS_SET_DTX (enc->dtx), 0);
-  opus_multistream_encoder_ctl (enc->state,
-      OPUS_SET_PACKET_LOSS_PERC (enc->packet_loss_percentage), 0);
-
-  opus_multistream_encoder_ctl (enc->state, OPUS_GET_LOOKAHEAD (&lookahead), 0);
-
-  GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size,
-      lookahead);
-
-  /* lookahead is samples, the Opus header wants it in 48kHz samples */
-  lookahead = lookahead * 48000 / enc->sample_rate;
-  enc->lookahead = enc->pending_lookahead = lookahead;
-
-  header = gst_codec_utils_opus_create_header (enc->sample_rate,
-      enc->n_channels, enc->channel_mapping_family,
-      enc->n_channels - enc->n_stereo_streams, enc->n_stereo_streams,
-      enc->decoding_channel_mapping, lookahead, 0);
-  tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
-  if (!tags)
-    tags = empty_tags = gst_tag_list_new_empty ();
-  comments =
-      gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags",
-      8, "Encoded with GStreamer opusenc");
-  caps = gst_codec_utils_opus_create_caps_from_header (header, comments);
-  if (empty_tags)
-    gst_tag_list_unref (empty_tags);
-  gst_buffer_unref (header);
-  gst_buffer_unref (comments);
-
-  /* negotiate with these caps */
-  GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
-
-  ret = gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
-  gst_caps_unref (caps);
-
-  return ret;
-
-encoder_creation_failed:
-  GST_ERROR_OBJECT (enc, "Encoder creation failed");
-  return FALSE;
-}
-
-static gboolean
-gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
-{
-  GstOpusEnc *enc;
-
-  enc = GST_OPUS_ENC (benc);
-
-  GST_DEBUG_OBJECT (enc, "sink event: %s", GST_EVENT_TYPE_NAME (event));
-  switch (GST_EVENT_TYPE (event)) {
-    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);
-      break;
-    }
-    case GST_EVENT_SEGMENT:
-      enc->encoded_samples = 0;
-      enc->consumed_samples = 0;
-      break;
-
-    default:
-      break;
-  }
-
-  return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
-}
-
-static GstCaps *
-gst_opus_enc_get_sink_template_caps (void)
-{
-  static volatile gsize init = 0;
-  static GstCaps *caps = NULL;
-
-  if (g_once_init_enter (&init)) {
-    GValue rate_array = G_VALUE_INIT;
-    GValue v = G_VALUE_INIT;
-    GstStructure *s1, *s2, *s;
-    gint i, c;
-
-    caps = gst_caps_new_empty ();
-
-    /* Generate our two template structures */
-    g_value_init (&rate_array, GST_TYPE_LIST);
-    g_value_init (&v, G_TYPE_INT);
-    g_value_set_int (&v, 8000);
-    gst_value_list_append_value (&rate_array, &v);
-    g_value_set_int (&v, 12000);
-    gst_value_list_append_value (&rate_array, &v);
-    g_value_set_int (&v, 16000);
-    gst_value_list_append_value (&rate_array, &v);
-    g_value_set_int (&v, 24000);
-    gst_value_list_append_value (&rate_array, &v);
-
-    s1 = gst_structure_new ("audio/x-raw",
-        "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
-        "layout", G_TYPE_STRING, "interleaved",
-        "rate", G_TYPE_INT, 48000, NULL);
-    s2 = gst_structure_new ("audio/x-raw",
-        "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
-        "layout", G_TYPE_STRING, "interleaved", NULL);
-    gst_structure_set_value (s2, "rate", &rate_array);
-    g_value_unset (&rate_array);
-    g_value_unset (&v);
-
-    /* Mono */
-    s = gst_structure_copy (s1);
-    gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
-    gst_caps_append_structure (caps, s);
-
-    s = gst_structure_copy (s2);
-    gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
-    gst_caps_append_structure (caps, s);
-
-    /* Stereo and further */
-    for (i = 2; i <= 8; i++) {
-      guint64 channel_mask = 0;
-      const GstAudioChannelPosition *pos = gst_opus_channel_positions[i - 1];
-
-      for (c = 0; c < i; c++) {
-        channel_mask |= G_GUINT64_CONSTANT (1) << pos[c];
-      }
-
-      s = gst_structure_copy (s1);
-      gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
-          GST_TYPE_BITMASK, channel_mask, NULL);
-      gst_caps_append_structure (caps, s);
-
-      s = gst_structure_copy (s2);
-      gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
-          GST_TYPE_BITMASK, channel_mask, NULL);
-      gst_caps_append_structure (caps, s);
-    }
-
-    gst_structure_free (s1);
-    gst_structure_free (s2);
-
-    g_once_init_leave (&init, 1);
-  }
-
-  return caps;
-}
-
-static GstCaps *
-gst_opus_enc_sink_getcaps (GstAudioEncoder * benc, GstCaps * filter)
-{
-  GstOpusEnc *enc;
-  GstCaps *caps;
-
-  enc = GST_OPUS_ENC (benc);
-
-  GST_DEBUG_OBJECT (enc, "sink getcaps");
-
-  caps = gst_opus_enc_get_sink_template_caps ();
-  caps = gst_audio_encoder_proxy_getcaps (benc, caps, filter);
-
-  GST_DEBUG_OBJECT (enc, "Returning caps: %" GST_PTR_FORMAT, caps);
-
-  return caps;
-}
-
-static GstFlowReturn
-gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
-{
-  guint8 *bdata = NULL, *data, *mdata = NULL;
-  gsize bsize, size;
-  gsize bytes;
-  gint ret = GST_FLOW_OK;
-  GstMapInfo map;
-  GstMapInfo omap;
-  gint outsize;
-  GstBuffer *outbuf;
-  GstSegment *segment;
-  GstClockTime duration;
-  guint64 trim_start = 0, trim_end = 0;
-
-  guint max_payload_size;
-  gint frame_samples, input_samples, output_samples;
-
-  g_mutex_lock (&enc->property_lock);
-
-  bytes = enc->frame_samples * enc->n_channels * 2;
-  max_payload_size = enc->max_payload_size;
-  frame_samples = input_samples = enc->frame_samples;
-
-  g_mutex_unlock (&enc->property_lock);
-
-  if (G_LIKELY (buf)) {
-    gst_buffer_map (buf, &map, GST_MAP_READ);
-    bdata = map.data;
-    bsize = map.size;
-
-    if (G_UNLIKELY (bsize % bytes)) {
-      gint64 diff;
-
-      GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
-      g_assert (bsize < bytes);
-
-      /* If encoding part of a frame, and we have no set stop time on
-       * the output segment, we update the segment stop time to reflect
-       * the last sample. This will let oggmux set the last page's
-       * granpos to tell a decoder the dummy samples should be clipped.
-       */
-      input_samples = bsize / (enc->n_channels * 2);
-      segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
-      if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
-        GST_DEBUG_OBJECT (enc,
-            "No stop time and partial frame, updating segment");
-        duration =
-            gst_util_uint64_scale_ceil (enc->consumed_samples + input_samples,
-            GST_SECOND, enc->sample_rate);
-        segment->stop = segment->start + duration;
-        GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
-            segment);
-        gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
-            gst_event_new_segment (segment));
-      }
-
-      diff =
-          (enc->encoded_samples + frame_samples) - (enc->consumed_samples +
-          input_samples);
-      if (diff >= 0) {
-        GST_DEBUG_OBJECT (enc,
-            "%" G_GINT64_FORMAT " extra samples of padding in this frame",
-            diff);
-        output_samples = frame_samples - diff;
-        trim_end = diff * 48000 / enc->sample_rate;
-      } else {
-        GST_DEBUG_OBJECT (enc,
-            "Need to add %" G_GINT64_FORMAT " extra samples in the next frame",
-            -diff);
-        output_samples = frame_samples;
-      }
-
-      size = ((bsize / bytes) + 1) * bytes;
-      mdata = g_malloc0 (size);
-      /* FIXME: Instead of silence, use LPC with the last real samples.
-       * Otherwise we will create a discontinuity here, which will distort the
-       * last few encoded samples
-       */
-      memcpy (mdata, bdata, bsize);
-      data = mdata;
-    } else {
-      data = bdata;
-      size = bsize;
-
-      /* Adjust for lookahead here */
-      if (enc->pending_lookahead) {
-        guint scaled_lookahead =
-            enc->pending_lookahead * enc->sample_rate / 48000;
-
-        if (input_samples > scaled_lookahead) {
-          output_samples = input_samples - scaled_lookahead;
-          trim_start = enc->pending_lookahead;
-          enc->pending_lookahead = 0;
-        } else {
-          trim_start = ((guint64) input_samples) * 48000 / enc->sample_rate;
-          enc->pending_lookahead -= trim_start;
-          output_samples = 0;
-        }
-      } else {
-        output_samples = input_samples;
-      }
-    }
-  } else {
-    if (enc->encoded_samples < enc->consumed_samples) {
-      /* FIXME: Instead of silence, use LPC with the last real samples.
-       * Otherwise we will create a discontinuity here, which will distort the
-       * last few encoded samples
-       */
-      data = mdata = g_malloc0 (bytes);
-      size = bytes;
-      output_samples = enc->consumed_samples - enc->encoded_samples;
-      input_samples = 0;
-      GST_DEBUG_OBJECT (enc, "draining %d samples", output_samples);
-      trim_end =
-          ((guint64) frame_samples - output_samples) * 48000 / enc->sample_rate;
-    } else if (enc->encoded_samples == enc->consumed_samples) {
-      GST_DEBUG_OBJECT (enc, "nothing to drain");
-      goto done;
-    } else {
-      g_assert_not_reached ();
-      goto done;
-    }
-  }
-
-  g_assert (size == bytes);
-
-  outbuf =
-      gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER (enc),
-      max_payload_size * enc->n_channels);
-  if (!outbuf)
-    goto done;
-
-  GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
-      frame_samples, (int) bytes);
-
-  if (trim_start || trim_end) {
-    GST_DEBUG_OBJECT (enc,
-        "Adding trim-start %" G_GUINT64_FORMAT " trim-end %" G_GUINT64_FORMAT,
-        trim_start, trim_end);
-    gst_buffer_add_audio_clipping_meta (outbuf, GST_FORMAT_DEFAULT, trim_start,
-        trim_end);
-  }
-
-  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
-
-  outsize =
-      opus_multistream_encode (enc->state, (const gint16 *) data,
-      frame_samples, omap.data, max_payload_size * enc->n_channels);
-
-  gst_buffer_unmap (outbuf, &omap);
-
-  if (outsize < 0) {
-    GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
-    ret = GST_FLOW_ERROR;
-    goto done;
-  } else if (outsize > max_payload_size) {
-    GST_WARNING_OBJECT (enc,
-        "Encoded size %d is higher than max payload size (%d bytes)",
-        outsize, max_payload_size);
-    ret = GST_FLOW_ERROR;
-    goto done;
-  }
-
-  GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
-  gst_buffer_set_size (outbuf, outsize);
-
-
-  ret =
-      gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
-      output_samples);
-  enc->encoded_samples += output_samples;
-  enc->consumed_samples += input_samples;
-
-done:
-
-  if (bdata)
-    gst_buffer_unmap (buf, &map);
-
-  g_free (mdata);
-
-  return ret;
-}
-
-static GstFlowReturn
-gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
-{
-  GstOpusEnc *enc;
-  GstFlowReturn ret = GST_FLOW_OK;
-
-  enc = GST_OPUS_ENC (benc);
-  GST_DEBUG_OBJECT (enc, "handle_frame");
-  GST_DEBUG_OBJECT (enc, "received buffer %p of %" G_GSIZE_FORMAT " bytes", buf,
-      buf ? gst_buffer_get_size (buf) : 0);
-
-  ret = gst_opus_enc_encode (enc, buf);
-
-  return ret;
-}
-
-static void
-gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
-{
-  GstOpusEnc *enc;
-
-  enc = GST_OPUS_ENC (object);
-
-  g_mutex_lock (&enc->property_lock);
-
-  switch (prop_id) {
-    case PROP_AUDIO:
-      g_value_set_boolean (value,
-          enc->audio_type == OPUS_APPLICATION_AUDIO ? TRUE : FALSE);
-      break;
-    case PROP_AUDIO_TYPE:
-      g_value_set_enum (value, enc->audio_type);
-      break;
-    case PROP_BITRATE:
-      g_value_set_int (value, enc->bitrate);
-      break;
-    case PROP_BANDWIDTH:
-      g_value_set_enum (value, enc->bandwidth);
-      break;
-    case PROP_FRAME_SIZE:
-      g_value_set_enum (value, enc->frame_size);
-      break;
-    case PROP_CBR:
-      GST_WARNING_OBJECT (enc,
-          "cbr property is deprecated; use bitrate-type instead");
-      g_value_set_boolean (value, enc->bitrate_type == BITRATE_TYPE_CBR);
-      break;
-    case PROP_CONSTRAINED_VBR:
-      GST_WARNING_OBJECT (enc,
-          "constrained-vbr property is deprecated; use bitrate-type instead");
-      g_value_set_boolean (value,
-          enc->bitrate_type == BITRATE_TYPE_CONSTRAINED_VBR);
-      break;
-    case PROP_BITRATE_TYPE:
-      g_value_set_enum (value, enc->bitrate_type);
-      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;
-    case PROP_MAX_PAYLOAD_SIZE:
-      g_value_set_uint (value, enc->max_payload_size);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-
-  g_mutex_unlock (&enc->property_lock);
-}
-
-static void
-gst_opus_enc_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstOpusEnc *enc;
-
-  enc = GST_OPUS_ENC (object);
-
-#define GST_OPUS_UPDATE_PROPERTY(prop,type,ctl) do { \
-  g_mutex_lock (&enc->property_lock); \
-  enc->prop = g_value_get_##type (value); \
-  if (enc->state) { \
-    opus_multistream_encoder_ctl (enc->state, OPUS_SET_##ctl (enc->prop)); \
-  } \
-  g_mutex_unlock (&enc->property_lock); \
-} while(0)
-
-  switch (prop_id) {
-    case PROP_AUDIO:
-      enc->audio_type =
-          g_value_get_boolean (value) ? OPUS_APPLICATION_AUDIO :
-          OPUS_APPLICATION_VOIP;
-      break;
-    case PROP_AUDIO_TYPE:
-      enc->audio_type = g_value_get_enum (value);
-      break;
-    case PROP_BITRATE:
-      GST_OPUS_UPDATE_PROPERTY (bitrate, int, BITRATE);
-      break;
-    case PROP_BANDWIDTH:
-      GST_OPUS_UPDATE_PROPERTY (bandwidth, enum, BANDWIDTH);
-      break;
-    case PROP_FRAME_SIZE:
-      g_mutex_lock (&enc->property_lock);
-      enc->frame_size = g_value_get_enum (value);
-      enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
-      gst_opus_enc_setup_base_class (enc, GST_AUDIO_ENCODER (enc));
-      g_mutex_unlock (&enc->property_lock);
-      break;
-    case PROP_CBR:
-      GST_WARNING_OBJECT (enc,
-          "cbr property is deprecated; use bitrate-type instead");
-      g_warning ("cbr property is deprecated; use bitrate-type instead");
-      g_mutex_lock (&enc->property_lock);
-      enc->bitrate_type = BITRATE_TYPE_CBR;
-      if (enc->state) {
-        opus_multistream_encoder_ctl (enc->state, OPUS_SET_VBR (FALSE));
-        opus_multistream_encoder_ctl (enc->state,
-            OPUS_SET_VBR_CONSTRAINT (FALSE), 0);
-      }
-      g_mutex_unlock (&enc->property_lock);
-      break;
-    case PROP_CONSTRAINED_VBR:
-      GST_WARNING_OBJECT (enc,
-          "constrained-vbr property is deprecated; use bitrate-type instead");
-      g_warning
-          ("constrained-vbr property is deprecated; use bitrate-type instead");
-      g_mutex_lock (&enc->property_lock);
-      enc->bitrate_type = BITRATE_TYPE_CONSTRAINED_VBR;
-      if (enc->state) {
-        opus_multistream_encoder_ctl (enc->state, OPUS_SET_VBR (TRUE));
-        opus_multistream_encoder_ctl (enc->state,
-            OPUS_SET_VBR_CONSTRAINT (TRUE), 0);
-      }
-      g_mutex_unlock (&enc->property_lock);
-      break;
-    case PROP_BITRATE_TYPE:
-      /* this one has an opposite meaning to the opus ctl... */
-      g_mutex_lock (&enc->property_lock);
-      enc->bitrate_type = g_value_get_enum (value);
-      if (enc->state) {
-        opus_multistream_encoder_ctl (enc->state,
-            OPUS_SET_VBR (enc->bitrate_type != BITRATE_TYPE_CBR));
-        opus_multistream_encoder_ctl (enc->state,
-            OPUS_SET_VBR_CONSTRAINT (enc->bitrate_type ==
-                BITRATE_TYPE_CONSTRAINED_VBR), 0);
-      }
-      g_mutex_unlock (&enc->property_lock);
-      break;
-    case PROP_COMPLEXITY:
-      GST_OPUS_UPDATE_PROPERTY (complexity, int, COMPLEXITY);
-      break;
-    case PROP_INBAND_FEC:
-      GST_OPUS_UPDATE_PROPERTY (inband_fec, boolean, INBAND_FEC);
-      break;
-    case PROP_DTX:
-      GST_OPUS_UPDATE_PROPERTY (dtx, boolean, DTX);
-      break;
-    case PROP_PACKET_LOSS_PERCENT:
-      GST_OPUS_UPDATE_PROPERTY (packet_loss_percentage, int, PACKET_LOSS_PERC);
-      break;
-    case PROP_MAX_PAYLOAD_SIZE:
-      g_mutex_lock (&enc->property_lock);
-      enc->max_payload_size = g_value_get_uint (value);
-      g_mutex_unlock (&enc->property_lock);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-
-#undef GST_OPUS_UPDATE_PROPERTY
-
-}
diff --git a/ext/opus/gstopusenc.h b/ext/opus/gstopusenc.h
deleted file mode 100644
index f447292af..000000000
--- a/ext/opus/gstopusenc.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/* GStreamer Opus Encoder
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Copyright (C) <2011-2012> Vincent Penquerc'h <vincent.penquerch@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., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-
-#ifndef __GST_OPUS_ENC_H__
-#define __GST_OPUS_ENC_H__
-
-
-#include <gst/gst.h>
-#include <gst/audio/gstaudioencoder.h>
-
-#include <opus_multistream.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 enum
-{
-  BITRATE_TYPE_CBR,
-  BITRATE_TYPE_VBR,
-  BITRATE_TYPE_CONSTRAINED_VBR,
-} GstOpusEncBitrateType;
-
-typedef struct _GstOpusEnc GstOpusEnc;
-typedef struct _GstOpusEncClass GstOpusEncClass;
-
-struct _GstOpusEnc {
-  GstAudioEncoder       element;
-
-  OpusMSEncoder        *state;
-
-  /* Locks those properties which may be changed at play time */
-  GMutex                property_lock;
-
-  /* properties */
-  gint                  audio_type;
-  gint                  bitrate;
-  gint                  bandwidth;
-  gint                  frame_size;
-  GstOpusEncBitrateType bitrate_type;
-  gint                  complexity;
-  gboolean              inband_fec;
-  gboolean              dtx;
-  gint                  packet_loss_percentage;
-  guint                 max_payload_size;
-
-  gint                  frame_samples;
-  gint                  n_channels;
-  gint                  sample_rate;
-
-  guint64               encoded_samples, consumed_samples;
-  guint16               lookahead, pending_lookahead;
-
-  guint8                channel_mapping_family;
-  guint8                encoding_channel_mapping[256];
-  guint8                decoding_channel_mapping[256];
-  guint8                n_stereo_streams;
-};
-
-struct _GstOpusEncClass {
-  GstAudioEncoderClass parent_class;
-
-  /* signals */
-  void (*frame_encoded) (GstElement *element);
-};
-
-GType gst_opus_enc_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_OPUS_ENC_H__ */
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 1828cb6f9..3f9ad9d2d 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -156,12 +156,6 @@ else
 check_opencv =
 endif
 
-if USE_OPUS
-check_opus = elements/opus
-else
-check_opus =
-endif
-
 if USE_SSH2
 check_curl_sftp = elements/curlsftpsink
 else
@@ -253,7 +247,6 @@ check_PROGRAMS = \
 	$(check_timidity)  \
 	$(check_kate)  \
 	$(check_opencv) \
-	$(check_opus)  \
 	$(check_curl) \
 	$(check_shm) \
 	elements/aiffparse \
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index 881dcd3d4..5910320a7 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -45,7 +45,6 @@ mxfmux
 neonhttpsrc
 netsim
 ofa
-opus
 pcapparse
 rtponvif
 rganalysis
diff --git a/tests/check/elements/opus.c b/tests/check/elements/opus.c
deleted file mode 100644
index c4b4b84e0..000000000
--- a/tests/check/elements/opus.c
+++ /dev/null
@@ -1,338 +0,0 @@
-/* GStreamer
- *
- * unit test for opus
- *
- * Copyright (C) <2011> Vincent Penquerc'h <vincent.penquerch@collbaora.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., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#include <unistd.h>
-
-#include <gst/check/gstcheck.h>
-
-#if G_BYTE_ORDER == G_BIG_ENDIAN
-#define AFORMAT "S16BE"
-#else
-#define AFORMAT "S16LE"
-#endif
-
-#define AUDIO_CAPS_STRING "audio/x-raw, " \
-                           "format = (string) " AFORMAT ", "\
-                           "layout = (string) interleaved, " \
-                           "rate = (int) 48000, " \
-                           "channels = (int) 1 "
-
-/* A lot of these taken from the vorbisdec test */
-
-/* For ease of programming we use globals to keep refs for our floating
- * src and sink pads we create; otherwise we always have to do get_pad,
- * get_peer, and then remove references in every test function */
-static GstPad *mydecsrcpad, *mydecsinkpad;
-static GstPad *myencsrcpad, *myencsinkpad;
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS_ANY);
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS_ANY);
-
-static GstElement *
-setup_opusdec (void)
-{
-  GstElement *opusdec;
-
-  GST_DEBUG ("setup_opusdec");
-  opusdec = gst_check_setup_element ("opusdec");
-  mydecsrcpad = gst_check_setup_src_pad (opusdec, &srctemplate);
-  mydecsinkpad = gst_check_setup_sink_pad (opusdec, &sinktemplate);
-  gst_pad_set_active (mydecsrcpad, TRUE);
-  gst_pad_set_active (mydecsinkpad, TRUE);
-
-  return opusdec;
-}
-
-static void
-cleanup_opusdec (GstElement * opusdec)
-{
-  GST_DEBUG ("cleanup_opusdec");
-  gst_element_set_state (opusdec, GST_STATE_NULL);
-
-  gst_pad_set_active (mydecsrcpad, FALSE);
-  gst_pad_set_active (mydecsinkpad, FALSE);
-  gst_check_teardown_src_pad (opusdec);
-  gst_check_teardown_sink_pad (opusdec);
-  gst_check_teardown_element (opusdec);
-}
-
-static GstElement *
-setup_opusenc (void)
-{
-  GstElement *opusenc;
-
-  GST_DEBUG ("setup_opusenc");
-  opusenc = gst_check_setup_element ("opusenc");
-  myencsrcpad = gst_check_setup_src_pad (opusenc, &srctemplate);
-  myencsinkpad = gst_check_setup_sink_pad (opusenc, &sinktemplate);
-  gst_pad_set_active (myencsrcpad, TRUE);
-  gst_pad_set_active (myencsinkpad, TRUE);
-
-  return opusenc;
-}
-
-static void
-cleanup_opusenc (GstElement * opusenc)
-{
-  GST_DEBUG ("cleanup_opusenc");
-  gst_element_set_state (opusenc, GST_STATE_NULL);
-
-  gst_pad_set_active (myencsrcpad, FALSE);
-  gst_pad_set_active (myencsinkpad, FALSE);
-  gst_check_teardown_src_pad (opusenc);
-  gst_check_teardown_sink_pad (opusenc);
-  gst_check_teardown_element (opusenc);
-}
-
-static void
-check_buffers (guint expected)
-{
-  GstBuffer *outbuffer;
-  guint i, num_buffers;
-
-  /* check buffers are the type we expect */
-  num_buffers = g_list_length (buffers);
-  fail_unless (num_buffers >= expected);
-  for (i = 0; i < num_buffers; ++i) {
-    outbuffer = GST_BUFFER (buffers->data);
-    fail_if (outbuffer == NULL);
-    fail_if (gst_buffer_get_size (outbuffer) == 0);
-
-    buffers = g_list_remove (buffers, outbuffer);
-
-    ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
-    gst_buffer_unref (outbuffer);
-    outbuffer = NULL;
-  }
-}
-
-GST_START_TEST (test_opus_encode_nothing)
-{
-  GstElement *opusenc;
-
-  opusenc = setup_opusenc ();
-  fail_unless (gst_element_set_state (opusenc,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to playing");
-
-  fail_unless (gst_pad_push_event (myencsrcpad, gst_event_new_eos ()) == TRUE);
-
-  fail_unless (gst_element_set_state (opusenc,
-          GST_STATE_READY) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to ready");
-
-  /* cleanup */
-  cleanup_opusenc (opusenc);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_opus_decode_nothing)
-{
-  GstElement *opusdec;
-
-  opusdec = setup_opusdec ();
-  fail_unless (gst_element_set_state (opusdec,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to playing");
-
-  fail_unless (gst_pad_push_event (mydecsrcpad, gst_event_new_eos ()) == TRUE);
-
-  fail_unless (gst_element_set_state (opusdec,
-          GST_STATE_READY) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to ready");
-
-  /* cleanup */
-  cleanup_opusdec (opusdec);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_opus_encode_samples)
-{
-  const unsigned int nsamples = 4096;
-  GstElement *opusenc;
-  GstBuffer *inbuffer;
-  GstCaps *caps;
-
-  opusenc = setup_opusenc ();
-
-  fail_unless (gst_element_set_state (opusenc,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to playing");
-
-  inbuffer = gst_buffer_new_and_alloc (nsamples * 2);
-  gst_buffer_memset (inbuffer, 0, 0, nsamples * 2);
-
-  GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_OFFSET (inbuffer) = 0;
-  GST_BUFFER_DURATION (inbuffer) = GST_CLOCK_TIME_NONE;
-  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-
-  caps = gst_caps_from_string (AUDIO_CAPS_STRING);
-  fail_unless (caps != NULL);
-  gst_check_setup_events (myencsrcpad, opusenc, caps, GST_FORMAT_TIME);
-  gst_caps_unref (caps);
-  gst_buffer_ref (inbuffer);
-
-  /* pushing gives away my reference ... */
-  fail_unless (gst_pad_push (myencsrcpad, inbuffer) == GST_FLOW_OK);
-  /* ... and nothing ends up on the global buffer list */
-  fail_unless (gst_pad_push_event (myencsrcpad, gst_event_new_eos ()) == TRUE);
-
-  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-  gst_buffer_unref (inbuffer);
-
-  fail_unless (gst_element_set_state (opusenc,
-          GST_STATE_READY) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to ready");
-
-  /* default frame size is 20 ms, at 48000 Hz that's 960 samples */
-  check_buffers ((nsamples + 959) / 960);
-
-  /* cleanup */
-  cleanup_opusenc (opusenc);
-  g_list_free (buffers);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_opus_encode_properties)
-{
-  const unsigned int nsamples = 4096;
-  enum
-  { steps = 20 };
-  GstElement *opusenc;
-  GstBuffer *inbuffer;
-  GstCaps *caps;
-  unsigned int step;
-  static const struct
-  {
-    const char *param;
-    int value;
-  } param_changes[steps] = {
-    {
-    "frame-size", 40}, {
-    "inband-fec", 1}, {
-    "complexity", 5}, {
-    "bandwidth", 1104}, {
-    "frame-size", 2}, {
-    "max-payload-size", 80}, {
-    "frame-size", 60}, {
-    "max-payload-size", 900}, {
-    "complexity", 1}, {
-    "bitrate", 30000}, {
-    "frame-size", 10}, {
-    "bitrate", 300000}, {
-    "inband-fec", 0}, {
-    "frame-size", 5}, {
-    "bandwidth", 1101}, {
-    "frame-size", 10}, {
-    "bitrate", 500000}, {
-    "frame-size", 5}, {
-    "bitrate", 80000}, {
-  "complexity", 8},};
-
-  opusenc = setup_opusenc ();
-
-  fail_unless (gst_element_set_state (opusenc,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to playing");
-
-  caps = gst_caps_from_string (AUDIO_CAPS_STRING);
-  fail_unless (caps != NULL);
-
-  gst_check_setup_events (myencsrcpad, opusenc, caps, GST_FORMAT_TIME);
-
-  for (step = 0; step < steps; ++step) {
-    GstSegment segment;
-
-    gst_segment_init (&segment, GST_FORMAT_TIME);
-    gst_pad_push_event (myencsrcpad, gst_event_new_segment (&segment));
-
-    inbuffer = gst_buffer_new_and_alloc (nsamples * 2);
-    gst_buffer_memset (inbuffer, 0, 0, nsamples * 2);
-
-    GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_OFFSET (inbuffer) = 0;
-    GST_BUFFER_DURATION (inbuffer) = GST_CLOCK_TIME_NONE;
-    ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-
-    gst_buffer_ref (inbuffer);
-
-    /* pushing gives away my reference ... */
-    fail_unless (gst_pad_push (myencsrcpad, inbuffer) == GST_FLOW_OK);
-    /* ... and nothing ends up on the global buffer list */
-    fail_unless (gst_pad_push_event (myencsrcpad,
-            gst_event_new_eos ()) == TRUE);
-
-    ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-    gst_buffer_unref (inbuffer);
-
-    /* change random parameters */
-    g_object_set (opusenc, param_changes[step].param, param_changes[step].value,
-        NULL);
-
-    check_buffers (1);
-
-    fail_unless (gst_pad_push_event (myencsrcpad,
-            gst_event_new_flush_start ()) == TRUE);
-    fail_unless (gst_pad_push_event (myencsrcpad,
-            gst_event_new_flush_stop (TRUE)) == TRUE);
-  }
-
-  gst_caps_unref (caps);
-
-  fail_unless (gst_element_set_state (opusenc,
-          GST_STATE_READY) == GST_STATE_CHANGE_SUCCESS,
-      "could not set to ready");
-
-  /* cleanup */
-  cleanup_opusenc (opusenc);
-  g_list_free (buffers);
-}
-
-GST_END_TEST;
-
-static Suite *
-opus_suite (void)
-{
-  Suite *s = suite_create ("opus");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-
-#define X if (0)
-  tcase_add_test (tc_chain, test_opus_encode_nothing);
-  tcase_add_test (tc_chain, test_opus_decode_nothing);
-  tcase_add_test (tc_chain, test_opus_encode_samples);
-  tcase_add_test (tc_chain, test_opus_encode_properties);
-#undef X
-
-  return s;
-}
-
-GST_CHECK_MAIN (opus);