audiomixer: remove, moved to -base

https://bugzilla.gnome.org/show_bug.cgi?id=791218

20 files changed, 3 insertions, 7763 deletions

diff --git a/Makefile.am b/Makefile.am
index 994cc3942..08d3ebe08 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -63,6 +63,7 @@ CRUFT_FILES = \
 	$(top_builddir)/ext/qt/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
+	$(top_builddir)/gst/audiomixer/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/audioparsers/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/camerabin2/.libs/libgstcamerabin2.so \
 	$(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \
@@ -105,6 +106,7 @@ CRUFT_DIRS = \
 	$(top_srcdir)/docs/plugins/tmpl \
 	$(top_srcdir)/gst/aacparse \
 	$(top_srcdir)/gst/amrparse \
+	$(top_srcdir)/gst/audiomixer \
 	$(top_srcdir)/gst/camerabin \
 	$(top_srcdir)/gst/dataurisrc \
 	$(top_srcdir)/gst/flacparse \
diff --git a/configure.ac b/configure.ac
index c6b482a66..ceb89af00 100644
--- a/configure.ac
+++ b/configure.ac
@@ -424,7 +424,6 @@ AG_GST_CHECK_PLUGIN(videoframe_audiolevel)
 AG_GST_CHECK_PLUGIN(asfmux)
 AG_GST_CHECK_PLUGIN(audiobuffersplit)
 AG_GST_CHECK_PLUGIN(audiofxbad)
-AG_GST_CHECK_PLUGIN(audiomixer)
 AG_GST_CHECK_PLUGIN(audiomixmatrix)
 AG_GST_CHECK_PLUGIN(compositor)
 AG_GST_CHECK_PLUGIN(audiovisualizers)
@@ -2486,7 +2485,6 @@ gst/videoframe_audiolevel/Makefile
 gst/asfmux/Makefile
 gst/audiobuffersplit/Makefile
 gst/audiofxbad/Makefile
-gst/audiomixer/Makefile
 gst/audiomixmatrix/Makefile
 gst/audiovisualizers/Makefile
 gst/autoconvert/Makefile
diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
index db624a1d2..a590060c1 100644
--- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
@@ -21,8 +21,6 @@
     <xi:include href="xml/element-aiffparse.xml" />
     <xi:include href="xml/element-aiffmux.xml" />
     <xi:include href="xml/element-assrender.xml" />
-    <xi:include href="xml/element-audiointerleave.xml" />
-    <xi:include href="xml/element-audiomixer.xml" />
     <xi:include href="xml/element-audioparse.xml" />
     <xi:include href="xml/element-autoconvert.xml" />
     <xi:include href="xml/element-bs2b.xml" />
@@ -118,7 +116,6 @@
   <chapter>
     <title>gst-plugins-bad Plugins</title>
     <xi:include href="xml/plugin-aiff.xml" />
-    <xi:include href="xml/plugin-audiomixer.xml" />
     <xi:include href="xml/plugin-audiovisualizers.xml" />
     <xi:include href="xml/plugin-autoconvert.xml" />
     <xi:include href="xml/plugin-assrender.xml" />
diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt
index 8e98c9c96..c1bae1d63 100644
--- a/docs/plugins/gst-plugins-bad-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt
@@ -191,38 +191,6 @@ gst_audio_channel_mix_get_type
 </SECTION>
 
 <SECTION>
-<FILE>element-audiointerleave</FILE>
-<TITLE>audiointerleave</TITLE>
-GstAudioInterleave
-<SUBSECTION Standard>
-GstAudioInterleaveClass
-GST_AUDIO_INTERLEAVE
-GST_AUDIO_INTERLEAVE_CAST
-GST_IS_AUDIO_INTERLEAVE
-GST_AUDIO_INTERLEAVE_CLASS
-GST_IS_AUDIO_INTERLEAVE_CLASS
-GST_TYPE_AUDIO_INTERLEAVE
-<SUBSECTION Private>
-gst_audio_interleave_get_type
-</SECTION>
-
-<SECTION>
-<FILE>element-audiomixer</FILE>
-<TITLE>audiomixer</TITLE>
-GstAudioMixer
-<SUBSECTION Standard>
-GstAudioMixerClass
-GST_AUDIO_MIXER
-GST_AUDIO_MIXER_CAST
-GST_IS_AUDIO_MIXER
-GST_AUDIO_MIXER_CLASS
-GST_IS_AUDIO_MIXER_CLASS
-GST_TYPE_AUDIO_MIXER
-<SUBSECTION Private>
-gst_audio_mixer_get_type
-</SECTION>
-
-<SECTION>
 <FILE>element-audiomixmatrix</FILE>
 <TITLE>audiomixmatrix</TITLE>
 GstAudioMixMatrix
diff --git a/docs/plugins/inspect/plugin-audiomixer.xml b/docs/plugins/inspect/plugin-audiomixer.xml
deleted file mode 100644
index 9d0b593af..000000000
--- a/docs/plugins/inspect/plugin-audiomixer.xml
+++ /dev/null
@@ -1,76 +0,0 @@
-<plugin>
-  <name>audiomixer</name>
-  <description>Mixes multiple audio streams</description>
-  <filename>../../gst/audiomixer/.libs/libgstaudiomixer.so</filename>
-  <basename>libgstaudiomixer.so</basename>
-  <version>1.13.0.1</version>
-  <license>LGPL</license>
-  <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins git</package>
-  <origin>Unknown package origin</origin>
-  <elements>
-    <element>
-      <name>audiointerleave</name>
-      <longname>AudioInterleave</longname>
-      <class>Generic/Audio</class>
-      <description>Mixes multiple audio streams</description>
-      <author>Olivier Crete &lt;olivier.crete@collabora.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink_%u</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)1, format=(string){ S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }, layout=(string){ non-interleaved, interleaved }</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], format=(string){ S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }, layout=(string)interleaved</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>audiomixer</name>
-      <longname>AudioMixer</longname>
-      <class>Generic/Audio</class>
-      <description>Mixes multiple audio streams</description>
-      <author>Sebastian Dröge &lt;sebastian@centricular.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink_%u</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>liveadder</name>
-      <longname>AudioMixer</longname>
-      <class>Generic/Audio</class>
-      <description>Mixes multiple audio streams</description>
-      <author>Sebastian Dröge &lt;sebastian@centricular.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink_%u</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }</details>
-        </caps>
-      </pads>
-    </element>
-  </elements>
-</plugin>
\ No newline at end of file
diff --git a/gst/audiomixer/Makefile.am b/gst/audiomixer/Makefile.am
deleted file mode 100644
index f1a4d7395..000000000
--- a/gst/audiomixer/Makefile.am
+++ /dev/null
@@ -1,21 +0,0 @@
-plugin_LTLIBRARIES = libgstaudiomixer.la
-
-ORC_SOURCE=gstaudiomixerorc
-include $(top_srcdir)/common/orc.mak
-
-
-libgstaudiomixer_la_SOURCES = gstaudiomixer.c gstaudiointerleave.c
-nodist_libgstaudiomixer_la_SOURCES = $(ORC_NODIST_SOURCES)
-libgstaudiomixer_la_CFLAGS = \
-	-I$(top_srcdir)/gst-libs \
-	-I$(top_builddir)/gst-libs \
-  $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
-	$(GST_CFLAGS) $(ORC_CFLAGS)
-libgstaudiomixer_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-libgstaudiomixer_la_LIBADD =  \
-		$(top_builddir)/gst-libs/gst/audio/libgstbadaudio-$(GST_API_VERSION).la \
-		$(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_API_VERSION@ \
-		$(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS)
-
-noinst_HEADERS = gstaudiomixer.h gstaudiointerleave.h
-
diff --git a/gst/audiomixer/gstaudiointerleave.c b/gst/audiomixer/gstaudiointerleave.c
deleted file mode 100644
index 90ec363ea..000000000
--- a/gst/audiomixer/gstaudiointerleave.c
+++ /dev/null
@@ -1,902 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2000 Wim Taymans <wtay@chello.be>
- *                    2005 Wim Taymans <wim@fluendo.com>
- *                    2007 Andy Wingo <wingo at pobox.com>
- *                    2008 Sebastian Dröge <slomo@circular-chaos.org>
- *                    2014 Collabora
- *                        Olivier Crete <olivier.crete@collabora.com>
- *
- * gstaudiointerleave.c: audiointerleave element, N in, one out,
- * samples are added
- *
- * 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.
- */
-/**
- * SECTION:element-audiointerleave
- * @title: audiointerleave
- *
- */
-
-/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
- * with newer GLib versions (>= 2.31.0) */
-#define GLIB_DISABLE_DEPRECATION_WARNINGS
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "gstaudiointerleave.h"
-#include <gst/audio/audio.h>
-
-#include <string.h>
-
-#define GST_CAT_DEFAULT gst_audio_interleave_debug
-GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
-
-enum
-{
-  PROP_PAD_0,
-  PROP_PAD_CHANNEL
-};
-
-G_DEFINE_TYPE (GstAudioInterleavePad, gst_audio_interleave_pad,
-    GST_TYPE_AUDIO_AGGREGATOR_PAD);
-
-static void
-gst_audio_interleave_pad_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (object);
-
-  switch (prop_id) {
-    case PROP_PAD_CHANNEL:
-      g_value_set_uint (value, pad->channel);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-
-static void
-gst_audio_interleave_pad_class_init (GstAudioInterleavePadClass * klass)
-{
-  GObjectClass *gobject_class = (GObjectClass *) klass;
-
-  gobject_class->get_property = gst_audio_interleave_pad_get_property;
-
-  g_object_class_install_property (gobject_class,
-      PROP_PAD_CHANNEL,
-      g_param_spec_uint ("channel",
-          "Channel number",
-          "Number of the channel of this pad in the output", 0, G_MAXUINT, 0,
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-}
-
-static void
-gst_audio_interleave_pad_init (GstAudioInterleavePad * pad)
-{
-}
-
-enum
-{
-  PROP_0,
-  PROP_CHANNEL_POSITIONS,
-  PROP_CHANNEL_POSITIONS_FROM_INPUT
-};
-
-/* elementfactory information */
-
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define CAPS \
-  GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
-  ", layout = (string) { interleaved, non-interleaved }"
-#else
-#define CAPS \
-  GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
-  ", layout = (string) { interleaved, non-interleaved }"
-#endif
-
-static GstStaticPadTemplate gst_audio_interleave_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink_%u",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) 1, "
-        "format = (string) " GST_AUDIO_FORMATS_ALL ", "
-        "layout = (string) {non-interleaved, interleaved}")
-    );
-
-static GstStaticPadTemplate gst_audio_interleave_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "format = (string) " GST_AUDIO_FORMATS_ALL ", "
-        "layout = (string) interleaved")
-    );
-
-static void gst_audio_interleave_child_proxy_init (gpointer g_iface,
-    gpointer iface_data);
-
-#define gst_audio_interleave_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstAudioInterleave, gst_audio_interleave,
-    GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
-        gst_audio_interleave_child_proxy_init));
-
-static void gst_audio_interleave_finalize (GObject * object);
-static void gst_audio_interleave_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void gst_audio_interleave_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-static gboolean gst_audio_interleave_setcaps (GstAudioInterleave * self,
-    GstPad * pad, GstCaps * caps);
-static GstPad *gst_audio_interleave_request_new_pad (GstElement * element,
-    GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps);
-static void gst_audio_interleave_release_pad (GstElement * element,
-    GstPad * pad);
-
-static gboolean gst_audio_interleave_stop (GstAggregator * agg);
-
-static gboolean
-gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg,
-    GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
-    GstBuffer * outbuf, guint out_offset, guint num_samples);
-
-
-static void
-__remove_channels (GstCaps * caps)
-{
-  GstStructure *s;
-  gint i, size;
-
-  size = gst_caps_get_size (caps);
-  for (i = 0; i < size; i++) {
-    s = gst_caps_get_structure (caps, i);
-    gst_structure_remove_field (s, "channel-mask");
-    gst_structure_remove_field (s, "channels");
-  }
-}
-
-static void
-__set_channels (GstCaps * caps, gint channels)
-{
-  GstStructure *s;
-  gint i, size;
-
-  size = gst_caps_get_size (caps);
-  for (i = 0; i < size; i++) {
-    s = gst_caps_get_structure (caps, i);
-    if (channels > 0)
-      gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
-    else
-      gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
-  }
-}
-
-/* we can only accept caps that we and downstream can handle.
- * if we have filtercaps set, use those to constrain the target caps.
- */
-static GstCaps *
-gst_audio_interleave_sink_getcaps (GstAggregator * agg, GstPad * pad,
-    GstCaps * filter)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
-  GstCaps *result = NULL, *peercaps, *sinkcaps;
-
-  GST_OBJECT_LOCK (self);
-  /* If we already have caps on one of the sink pads return them */
-  if (self->sinkcaps)
-    result = gst_caps_copy (self->sinkcaps);
-  GST_OBJECT_UNLOCK (self);
-
-  if (result == NULL) {
-    /* get the downstream possible caps */
-    peercaps = gst_pad_peer_query_caps (agg->srcpad, NULL);
-
-    /* get the allowed caps on this sinkpad */
-    sinkcaps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
-    __remove_channels (sinkcaps);
-    if (peercaps) {
-      peercaps = gst_caps_make_writable (peercaps);
-      __remove_channels (peercaps);
-      /* if the peer has caps, intersect */
-      GST_DEBUG_OBJECT (pad, "intersecting peer and template caps");
-      result = gst_caps_intersect (peercaps, sinkcaps);
-      gst_caps_unref (peercaps);
-      gst_caps_unref (sinkcaps);
-    } else {
-      /* the peer has no caps (or there is no peer), just use the allowed caps
-       * of this sinkpad. */
-      GST_DEBUG_OBJECT (pad, "no peer caps, using sinkcaps");
-      result = sinkcaps;
-    }
-    __set_channels (result, 1);
-  }
-
-  if (filter != NULL) {
-    GstCaps *caps = result;
-
-    GST_LOG_OBJECT (pad, "intersecting filter caps %" GST_PTR_FORMAT " with "
-        "preliminary result %" GST_PTR_FORMAT, filter, caps);
-
-    result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
-    gst_caps_unref (caps);
-  }
-
-  GST_DEBUG_OBJECT (pad, "Returning caps %" GST_PTR_FORMAT, result);
-
-  return result;
-}
-
-static gboolean
-gst_audio_interleave_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
-    GstQuery * query)
-{
-  gboolean res = FALSE;
-
-  switch (GST_QUERY_TYPE (query)) {
-    case GST_QUERY_CAPS:
-    {
-      GstCaps *filter, *caps;
-
-      gst_query_parse_caps (query, &filter);
-      caps = gst_audio_interleave_sink_getcaps (agg, GST_PAD (aggpad), filter);
-      gst_query_set_caps_result (query, caps);
-      gst_caps_unref (caps);
-      res = TRUE;
-      break;
-    }
-    default:
-      res =
-          GST_AGGREGATOR_CLASS (parent_class)->sink_query (agg, aggpad, query);
-      break;
-  }
-
-  return res;
-}
-
-static gint
-compare_positions (gconstpointer a, gconstpointer b, gpointer user_data)
-{
-  const gint i = *(const gint *) a;
-  const gint j = *(const gint *) b;
-  const gint *pos = (const gint *) user_data;
-
-  if (pos[i] < pos[j])
-    return -1;
-  else if (pos[i] > pos[j])
-    return 1;
-  else
-    return 0;
-}
-
-static gboolean
-gst_audio_interleave_channel_positions_to_mask (GValueArray * positions,
-    gint default_ordering_map[64], guint64 * mask)
-{
-  gint i;
-  guint channels;
-  GstAudioChannelPosition *pos;
-  gboolean ret;
-
-  channels = positions->n_values;
-  pos = g_new (GstAudioChannelPosition, channels);
-
-  for (i = 0; i < channels; i++) {
-    GValue *val;
-
-    val = g_value_array_get_nth (positions, i);
-    pos[i] = g_value_get_enum (val);
-  }
-
-  /* sort the default ordering map according to the position order */
-  for (i = 0; i < channels; i++) {
-    default_ordering_map[i] = i;
-  }
-  g_qsort_with_data (default_ordering_map, channels,
-      sizeof (*default_ordering_map), compare_positions, pos);
-
-  ret = gst_audio_channel_positions_to_mask (pos, channels, FALSE, mask);
-  g_free (pos);
-
-  return ret;
-}
-
-
-/* Must be called with the object lock held */
-
-static guint64
-gst_audio_interleave_get_channel_mask (GstAudioInterleave * self)
-{
-  guint64 channel_mask = 0;
-
-  if (self->channels <= 64 &&
-      self->channel_positions != NULL &&
-      self->channels == self->channel_positions->n_values) {
-    if (!gst_audio_interleave_channel_positions_to_mask
-        (self->channel_positions, self->default_channels_ordering_map,
-            &channel_mask)) {
-      GST_WARNING_OBJECT (self, "Invalid channel positions, using NONE");
-      channel_mask = 0;
-    }
-  } else if (self->channels <= 64) {
-    GST_WARNING_OBJECT (self, "Using NONE channel positions");
-  }
-
-  return channel_mask;
-}
-
-
-#define MAKE_FUNC(type) \
-static void interleave_##type (guint##type *out, guint##type *in, \
-    guint stride, guint nframes) \
-{ \
-  gint i; \
-  \
-  for (i = 0; i < nframes; i++) { \
-    *out = in[i]; \
-    out += stride; \
-  } \
-}
-
-MAKE_FUNC (8);
-MAKE_FUNC (16);
-MAKE_FUNC (32);
-MAKE_FUNC (64);
-
-static void
-interleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes)
-{
-  gint i;
-
-  for (i = 0; i < nframes; i++) {
-    memcpy (out, in, 3);
-    out += stride * 3;
-    in += 3;
-  }
-}
-
-static void
-gst_audio_interleave_set_process_function (GstAudioInterleave * self,
-    GstAudioInfo * info)
-{
-  switch (GST_AUDIO_INFO_WIDTH (info)) {
-    case 8:
-      self->func = (GstInterleaveFunc) interleave_8;
-      break;
-    case 16:
-      self->func = (GstInterleaveFunc) interleave_16;
-      break;
-    case 24:
-      self->func = (GstInterleaveFunc) interleave_24;
-      break;
-    case 32:
-      self->func = (GstInterleaveFunc) interleave_32;
-      break;
-    case 64:
-      self->func = (GstInterleaveFunc) interleave_64;
-      break;
-    default:
-      g_assert_not_reached ();
-      break;
-  }
-}
-
-
-/* the first caps we receive on any of the sinkpads will define the caps for all
- * the other sinkpads because we can only mix streams with the same caps.
- */
-static gboolean
-gst_audio_interleave_setcaps (GstAudioInterleave * self, GstPad * pad,
-    GstCaps * caps)
-{
-  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self);
-  GstAudioInfo info;
-  GValue *val;
-  guint channel;
-  gboolean new = FALSE;
-
-  if (!gst_audio_info_from_caps (&info, caps))
-    goto invalid_format;
-
-  GST_OBJECT_LOCK (self);
-  if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps))
-    goto cannot_change_caps;
-
-  if (!self->sinkcaps) {
-    GstCaps *sinkcaps = gst_caps_copy (caps);
-    GstStructure *s = gst_caps_get_structure (sinkcaps, 0);
-
-    gst_structure_remove_field (s, "channel-mask");
-
-    GST_DEBUG_OBJECT (self, "setting sinkcaps %" GST_PTR_FORMAT, sinkcaps);
-
-    gst_caps_replace (&self->sinkcaps, sinkcaps);
-    gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (aagg));
-
-    gst_caps_unref (sinkcaps);
-    new = TRUE;
-  }
-
-  if (self->channel_positions_from_input
-      && GST_AUDIO_INFO_CHANNELS (&info) == 1) {
-    channel = GST_AUDIO_INTERLEAVE_PAD (pad)->channel;
-    val = g_value_array_get_nth (self->input_channel_positions, channel);
-    g_value_set_enum (val, GST_AUDIO_INFO_POSITION (&info, 0));
-  }
-  GST_OBJECT_UNLOCK (self);
-
-  gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
-      caps);
-
-  if (!new)
-    return TRUE;
-
-  GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
-
-  return TRUE;
-
-  /* ERRORS */
-invalid_format:
-  {
-    GST_WARNING_OBJECT (self, "invalid format set as caps: %" GST_PTR_FORMAT,
-        caps);
-    return FALSE;
-  }
-cannot_change_caps:
-  {
-    GST_OBJECT_UNLOCK (self);
-    GST_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't "
-        "change", self->sinkcaps);
-    return FALSE;
-  }
-}
-
-static gboolean
-gst_audio_interleave_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
-    GstEvent * event)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
-  gboolean res = TRUE;
-
-  GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad",
-      GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_CAPS:
-    {
-      GstCaps *caps;
-
-      gst_event_parse_caps (event, &caps);
-      res = gst_audio_interleave_setcaps (self, GST_PAD_CAST (aggpad), caps);
-      gst_event_unref (event);
-      event = NULL;
-      break;
-    }
-    default:
-      break;
-  }
-
-  if (event != NULL)
-    return GST_AGGREGATOR_CLASS (parent_class)->sink_event (agg, aggpad, event);
-
-  return res;
-}
-
-static GstFlowReturn
-gst_audio_interleave_update_src_caps (GstAggregator * agg, GstCaps * caps,
-    GstCaps ** ret)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
-  GstStructure *s;
-
-  /* This means that either no caps have been set on the sink pad (if
-   * sinkcaps is NULL) or that there is no sink pad (if channels == 0).
-   */
-  GST_OBJECT_LOCK (self);
-  if (self->sinkcaps == NULL || self->channels == 0) {
-    GST_OBJECT_UNLOCK (self);
-    return GST_FLOW_NOT_NEGOTIATED;
-  }
-
-  *ret = gst_caps_copy (self->sinkcaps);
-  s = gst_caps_get_structure (*ret, 0);
-
-  gst_structure_set (s, "channels", G_TYPE_INT, self->channels, "layout",
-      G_TYPE_STRING, "interleaved", "channel-mask", GST_TYPE_BITMASK,
-      gst_audio_interleave_get_channel_mask (self), NULL);
-
-  GST_OBJECT_UNLOCK (self);
-
-  return GST_FLOW_OK;
-}
-
-static gboolean
-gst_audio_interleave_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
-  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self);
-
-  if (!GST_AGGREGATOR_CLASS (parent_class)->negotiated_src_caps (agg, caps))
-    return FALSE;
-
-  gst_audio_interleave_set_process_function (self, &aagg->info);
-
-  return TRUE;
-}
-
-static void
-gst_audio_interleave_class_init (GstAudioInterleaveClass * klass)
-{
-  GObjectClass *gobject_class = (GObjectClass *) klass;
-  GstElementClass *gstelement_class = (GstElementClass *) klass;
-  GstAggregatorClass *agg_class = (GstAggregatorClass *) klass;
-  GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass;
-
-  GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiointerleave", 0,
-      "audio interleaving element");
-
-  gobject_class->set_property = gst_audio_interleave_set_property;
-  gobject_class->get_property = gst_audio_interleave_get_property;
-  gobject_class->finalize = gst_audio_interleave_finalize;
-
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_audio_interleave_src_template);
-  gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
-      &gst_audio_interleave_sink_template, GST_TYPE_AUDIO_INTERLEAVE_PAD);
-  gst_element_class_set_static_metadata (gstelement_class, "AudioInterleave",
-      "Generic/Audio", "Mixes multiple audio streams",
-      "Olivier Crete <olivier.crete@collabora.com>");
-
-  gstelement_class->request_new_pad =
-      GST_DEBUG_FUNCPTR (gst_audio_interleave_request_new_pad);
-  gstelement_class->release_pad =
-      GST_DEBUG_FUNCPTR (gst_audio_interleave_release_pad);
-
-  agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audio_interleave_sink_query);
-  agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audio_interleave_sink_event);
-  agg_class->stop = gst_audio_interleave_stop;
-  agg_class->update_src_caps = gst_audio_interleave_update_src_caps;
-  agg_class->negotiated_src_caps = gst_audio_interleave_negotiated_src_caps;
-
-  aagg_class->aggregate_one_buffer = gst_audio_interleave_aggregate_one_buffer;
-  aagg_class->convert_buffer = NULL;
-
-  /**
-   * GstInterleave:channel-positions
-   *
-   * Channel positions: This property controls the channel positions
-   * that are used on the src caps. The number of elements should be
-   * the same as the number of sink pads and the array should contain
-   * a valid list of channel positions. The n-th element of the array
-   * is the position of the n-th sink pad.
-   *
-   * These channel positions will only be used if they're valid and the
-   * number of elements is the same as the number of channels. If this
-   * is not given a NONE layout will be used.
-   *
-   */
-  g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS,
-      g_param_spec_value_array ("channel-positions", "Channel positions",
-          "Channel positions used on the output",
-          g_param_spec_enum ("channel-position", "Channel position",
-              "Channel position of the n-th input",
-              GST_TYPE_AUDIO_CHANNEL_POSITION,
-              GST_AUDIO_CHANNEL_POSITION_NONE,
-              G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  /**
-   * GstInterleave:channel-positions-from-input
-   *
-   * Channel positions from input: If this property is set to %TRUE the channel
-   * positions will be taken from the input caps if valid channel positions for
-   * the output can be constructed from them. If this is set to %TRUE setting the
-   * channel-positions property overwrites this property again.
-   *
-   */
-  g_object_class_install_property (gobject_class,
-      PROP_CHANNEL_POSITIONS_FROM_INPUT,
-      g_param_spec_boolean ("channel-positions-from-input",
-          "Channel positions from input",
-          "Take channel positions from the input", TRUE,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-}
-
-static void
-gst_audio_interleave_init (GstAudioInterleave * self)
-{
-  self->input_channel_positions = g_value_array_new (0);
-  self->channel_positions_from_input = TRUE;
-  self->channel_positions = self->input_channel_positions;
-}
-
-static void
-gst_audio_interleave_finalize (GObject * object)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object);
-
-  if (self->channel_positions
-      && self->channel_positions != self->input_channel_positions) {
-    g_value_array_free (self->channel_positions);
-    self->channel_positions = NULL;
-  }
-
-  if (self->input_channel_positions) {
-    g_value_array_free (self->input_channel_positions);
-    self->input_channel_positions = NULL;
-  }
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_audio_interleave_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object);
-
-  switch (prop_id) {
-    case PROP_CHANNEL_POSITIONS:
-      g_return_if_fail (
-          ((GValueArray *) g_value_get_boxed (value))->n_values > 0);
-
-      if (self->channel_positions &&
-          self->channel_positions != self->input_channel_positions)
-        g_value_array_free (self->channel_positions);
-
-      self->channel_positions = g_value_dup_boxed (value);
-      self->channel_positions_from_input = FALSE;
-      break;
-    case PROP_CHANNEL_POSITIONS_FROM_INPUT:
-      self->channel_positions_from_input = g_value_get_boolean (value);
-
-      if (self->channel_positions_from_input) {
-        if (self->channel_positions &&
-            self->channel_positions != self->input_channel_positions)
-          g_value_array_free (self->channel_positions);
-        self->channel_positions = self->input_channel_positions;
-      }
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_audio_interleave_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object);
-
-  switch (prop_id) {
-    case PROP_CHANNEL_POSITIONS:
-      g_value_set_boxed (value, self->channel_positions);
-      break;
-    case PROP_CHANNEL_POSITIONS_FROM_INPUT:
-      g_value_set_boolean (value, self->channel_positions_from_input);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static gboolean
-gst_audio_interleave_stop (GstAggregator * agg)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
-
-  if (!GST_AGGREGATOR_CLASS (parent_class)->stop (agg))
-    return FALSE;
-
-  gst_caps_replace (&self->sinkcaps, NULL);
-
-  return TRUE;
-}
-
-static GstPad *
-gst_audio_interleave_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * req_name, const GstCaps * caps)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (element);
-  GstAudioInterleavePad *newpad;
-  gchar *pad_name;
-  gint channel, padnumber;
-  GValue val = { 0, };
-
-  /* FIXME: We ignore req_name, this is evil! */
-
-  GST_OBJECT_LOCK (self);
-  padnumber = g_atomic_int_add (&self->padcounter, 1);
-  channel = self->channels++;
-  if (!self->channel_positions_from_input)
-    channel = padnumber;
-  GST_OBJECT_UNLOCK (self);
-
-  pad_name = g_strdup_printf ("sink_%u", padnumber);
-  newpad = (GstAudioInterleavePad *)
-      GST_ELEMENT_CLASS (parent_class)->request_new_pad (element,
-      templ, pad_name, caps);
-  g_free (pad_name);
-  if (newpad == NULL)
-    goto could_not_create;
-
-  newpad->channel = channel;
-  gst_pad_use_fixed_caps (GST_PAD (newpad));
-
-  gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (newpad),
-      GST_OBJECT_NAME (newpad));
-
-
-  g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
-  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_NONE);
-  self->input_channel_positions =
-      g_value_array_append (self->input_channel_positions, &val);
-  g_value_unset (&val);
-
-  /* Update the src caps if we already have them */
-  gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (self));
-
-  return GST_PAD_CAST (newpad);
-
-could_not_create:
-  {
-    GST_DEBUG_OBJECT (element, "could not create/add  pad");
-    return NULL;
-  }
-}
-
-static void
-gst_audio_interleave_release_pad (GstElement * element, GstPad * pad)
-{
-  GstAudioInterleave *self;
-  gint position;
-  GList *l;
-
-  self = GST_AUDIO_INTERLEAVE (element);
-
-  /* Take lock to make sure we're not changing this when processing buffers */
-  GST_OBJECT_LOCK (self);
-
-  self->channels--;
-
-  position = GST_AUDIO_INTERLEAVE_PAD (pad)->channel;
-  g_value_array_remove (self->input_channel_positions, position);
-
-  /* Update channel numbers */
-  /* Taken above, GST_OBJECT_LOCK (self); */
-  for (l = GST_ELEMENT_CAST (self)->sinkpads; l != NULL; l = l->next) {
-    GstAudioInterleavePad *ipad = GST_AUDIO_INTERLEAVE_PAD (l->data);
-
-    if (GST_AUDIO_INTERLEAVE_PAD (pad)->channel < ipad->channel)
-      ipad->channel--;
-  }
-
-  gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (self));
-  GST_OBJECT_UNLOCK (self);
-
-
-  GST_DEBUG_OBJECT (self, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
-
-  gst_child_proxy_child_removed (GST_CHILD_PROXY (self), G_OBJECT (pad),
-      GST_OBJECT_NAME (pad));
-
-  GST_ELEMENT_CLASS (parent_class)->release_pad (element, pad);
-}
-
-
-/* Called with object lock and pad object lock held */
-static gboolean
-gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg,
-    GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
-    GstBuffer * outbuf, guint out_offset, guint num_frames)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (aagg);
-  GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (aaggpad);
-  GstMapInfo inmap;
-  GstMapInfo outmap;
-  gint out_width, in_bpf, out_bpf, out_channels, channel;
-  guint8 *outdata;
-
-  GST_OBJECT_LOCK (aagg);
-  GST_OBJECT_LOCK (aaggpad);
-
-  out_width = GST_AUDIO_INFO_WIDTH (&aagg->info) / 8;
-  in_bpf = GST_AUDIO_INFO_BPF (&aaggpad->info);
-  out_bpf = GST_AUDIO_INFO_BPF (&aagg->info);
-  out_channels = GST_AUDIO_INFO_CHANNELS (&aagg->info);
-
-  gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
-  gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
-  GST_LOG_OBJECT (pad, "interleaves %u frames on channel %d/%d at offset %u"
-      " from offset %u", num_frames, pad->channel, out_channels,
-      out_offset * out_bpf, in_offset * in_bpf);
-
-  if (self->channels > 64) {
-    channel = pad->channel;
-  } else {
-    channel = self->default_channels_ordering_map[pad->channel];
-  }
-
-  outdata = outmap.data + (out_offset * out_bpf) + (out_width * channel);
-
-
-  self->func (outdata, inmap.data + (in_offset * in_bpf), out_channels,
-      num_frames);
-
-
-  gst_buffer_unmap (inbuf, &inmap);
-  gst_buffer_unmap (outbuf, &outmap);
-
-  GST_OBJECT_UNLOCK (aaggpad);
-  GST_OBJECT_UNLOCK (aagg);
-
-  return TRUE;
-}
-
-
-/* GstChildProxy implementation */
-static GObject *
-gst_audio_interleave_child_proxy_get_child_by_index (GstChildProxy *
-    child_proxy, guint index)
-{
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (child_proxy);
-  GObject *obj = NULL;
-
-  GST_OBJECT_LOCK (self);
-  obj = g_list_nth_data (GST_ELEMENT_CAST (self)->sinkpads, index);
-  if (obj)
-    gst_object_ref (obj);
-  GST_OBJECT_UNLOCK (self);
-
-  return obj;
-}
-
-static guint
-gst_audio_interleave_child_proxy_get_children_count (GstChildProxy *
-    child_proxy)
-{
-  guint count = 0;
-  GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (child_proxy);
-
-  GST_OBJECT_LOCK (self);
-  count = GST_ELEMENT_CAST (self)->numsinkpads;
-  GST_OBJECT_UNLOCK (self);
-  GST_INFO_OBJECT (self, "Children Count: %d", count);
-
-  return count;
-}
-
-static void
-gst_audio_interleave_child_proxy_init (gpointer g_iface, gpointer iface_data)
-{
-  GstChildProxyInterface *iface = g_iface;
-
-  GST_INFO ("intializing child proxy interface");
-  iface->get_child_by_index =
-      gst_audio_interleave_child_proxy_get_child_by_index;
-  iface->get_children_count =
-      gst_audio_interleave_child_proxy_get_children_count;
-}
diff --git a/gst/audiomixer/gstaudiointerleave.h b/gst/audiomixer/gstaudiointerleave.h
deleted file mode 100644
index bf46f4a50..000000000
--- a/gst/audiomixer/gstaudiointerleave.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2000 Wim Taymans <wtay@chello.be>
- * Copyright (C) 2013      Sebastian Dröge <slomo@circular-chaos.org>
- *
- * gstaudiointerleave.h: Header for audiointerleave element
- *
- * 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_AUDIO_INTERLEAVE_H__
-#define __GST_AUDIO_INTERLEAVE_H__
-
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-
-#include <gst/audio/gstaudioaggregator.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AUDIO_INTERLEAVE            (gst_audio_interleave_get_type())
-#define GST_AUDIO_INTERLEAVE(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_INTERLEAVE,GstAudioInterleave))
-#define GST_IS_AUDIO_INTERLEAVE(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_INTERLEAVE))
-#define GST_AUDIO_INTERLEAVE_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_INTERLEAVE,GstAudioInterleaveClass))
-#define GST_IS_AUDIO_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_INTERLEAVE))
-#define GST_AUDIO_INTERLEAVE_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_INTERLEAVE,GstAudioInterleaveClass))
-
-typedef struct _GstAudioInterleave             GstAudioInterleave;
-typedef struct _GstAudioInterleaveClass        GstAudioInterleaveClass;
-
-typedef struct _GstAudioInterleavePad GstAudioInterleavePad;
-typedef struct _GstAudioInterleavePadClass GstAudioInterleavePadClass;
-
-typedef void (*GstInterleaveFunc) (gpointer out, gpointer in, guint stride,
-    guint nframes);
-
-/**
- * GstAudioInterleave:
- *
- * The GstAudioInterleave object structure.
- */
-struct _GstAudioInterleave {
-  GstAudioAggregator parent;
-
-  gint padcounter;
-  guint channels; /* object lock */
-
-  GstCaps *sinkcaps;
-
-  GValueArray *channel_positions;
-  GValueArray *input_channel_positions;
-  gboolean channel_positions_from_input;
-
-  gint default_channels_ordering_map[64];
-
-  GstInterleaveFunc func;
-};
-
-struct _GstAudioInterleaveClass {
-  GstAudioAggregatorClass parent_class;
-};
-
-GType    gst_audio_interleave_get_type (void);
-
-#define GST_TYPE_AUDIO_INTERLEAVE_PAD            (gst_audio_interleave_pad_get_type())
-#define GST_AUDIO_INTERLEAVE_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_INTERLEAVE_PAD,GstAudioInterleavePad))
-#define GST_IS_AUDIO_INTERLEAVE_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_INTERLEAVE_PAD))
-#define GST_AUDIO_INTERLEAVE_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_INTERLEAVE_PAD,GstAudioInterleavePadClass))
-#define GST_IS_AUDIO_INTERLEAVE_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_INTERLEAVE_PAD))
-#define GST_AUDIO_INTERLEAVE_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_INTERLEAVE_PAD,GstAudioInterleavePadClass))
-
-struct _GstAudioInterleavePad {
-  GstAudioAggregatorPad parent;
-
-  guint channel;
-};
-
-struct _GstAudioInterleavePadClass {
-  GstAudioAggregatorPadClass parent_class;
-};
-
-GType gst_audio_interleave_pad_get_type (void);
-
-G_END_DECLS
-
-
-#endif /* __GST_AUDIO_INTERLEAVE_H__ */
diff --git a/gst/audiomixer/gstaudiomixer.c b/gst/audiomixer/gstaudiomixer.c
deleted file mode 100644
index a0f569010..000000000
--- a/gst/audiomixer/gstaudiomixer.c
+++ /dev/null
@@ -1,577 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2001 Thomas <thomas@apestaart.org>
- *               2005,2006 Wim Taymans <wim@fluendo.com>
- *                    2013 Sebastian Dröge <sebastian@centricular.com>
- *
- * audiomixer.c: AudioMixer element, N in, one out, samples are added
- *
- * 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.
- */
-/**
- * SECTION:element-audiomixer
- * @title: audiomixer
- *
- * The audiomixer allows to mix several streams into one by adding the data.
- * Mixed data is clamped to the min/max values of the data format.
- *
- * Unlike the adder element audiomixer properly synchronises all input streams
- * and also handles live inputs such as capture sources or RTP properly.
- *
- * The audiomixer element can accept any sort of raw audio data, it will
- * be converted to the target format if necessary, with the exception
- * of the sample rate, which has to be identical to either what downstream
- * expects, or the sample rate of the first configured pad. Use a capsfilter
- * after the audiomixer element if you want to precisely control the format
- * that comes out of the audiomixer, which supports changing the format of
- * its output while playing.
- *
- * If you want to control the manner in which incoming data gets converted,
- * see the #GstAudioAggregatorPad:converter-config property, which will let
- * you for example change the way in which channels may get remapped.
- *
- * The input pads are from a GstPad subclass and have additional
- * properties to mute each pad individually and set the volume:
- *
- * * "mute": Whether to mute the pad or not (#gboolean)
- * * "volume": The volume of the pad, between 0.0 and 10.0 (#gdouble)
- *
- * ## Example launch line
- * |[
- * gst-launch-1.0 audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
- * ]| This pipeline produces two sine waves mixed together.
- *
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "gstaudiomixer.h"
-#include <gst/audio/audio.h>
-#include <string.h>             /* strcmp */
-#include "gstaudiomixerorc.h"
-
-#include "gstaudiointerleave.h"
-
-#define GST_CAT_DEFAULT gst_audiomixer_debug
-GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
-
-#define DEFAULT_PAD_VOLUME (1.0)
-#define DEFAULT_PAD_MUTE (FALSE)
-
-/* some defines for audio processing */
-/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
- * we map 1.0 to VOLUME_UNITY_INT*
- */
-#define VOLUME_UNITY_INT8            8  /* internal int for unity 2^(8-5) */
-#define VOLUME_UNITY_INT8_BIT_SHIFT  3  /* number of bits to shift for unity */
-#define VOLUME_UNITY_INT16           2048       /* internal int for unity 2^(16-5) */
-#define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */
-#define VOLUME_UNITY_INT24           524288     /* internal int for unity 2^(24-5) */
-#define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */
-#define VOLUME_UNITY_INT32           134217728  /* internal int for unity 2^(32-5) */
-#define VOLUME_UNITY_INT32_BIT_SHIFT 27
-
-enum
-{
-  PROP_PAD_0,
-  PROP_PAD_VOLUME,
-  PROP_PAD_MUTE
-};
-
-G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad,
-    GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD);
-
-static void
-gst_audiomixer_pad_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object);
-
-  switch (prop_id) {
-    case PROP_PAD_VOLUME:
-      g_value_set_double (value, pad->volume);
-      break;
-    case PROP_PAD_MUTE:
-      g_value_set_boolean (value, pad->mute);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_audiomixer_pad_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object);
-
-  switch (prop_id) {
-    case PROP_PAD_VOLUME:
-      GST_OBJECT_LOCK (pad);
-      pad->volume = g_value_get_double (value);
-      pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8;
-      pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16;
-      pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32;
-      GST_OBJECT_UNLOCK (pad);
-      break;
-    case PROP_PAD_MUTE:
-      GST_OBJECT_LOCK (pad);
-      pad->mute = g_value_get_boolean (value);
-      GST_OBJECT_UNLOCK (pad);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_audiomixer_pad_class_init (GstAudioMixerPadClass * klass)
-{
-  GObjectClass *gobject_class = (GObjectClass *) klass;
-
-  gobject_class->set_property = gst_audiomixer_pad_set_property;
-  gobject_class->get_property = gst_audiomixer_pad_get_property;
-
-  g_object_class_install_property (gobject_class, PROP_PAD_VOLUME,
-      g_param_spec_double ("volume", "Volume", "Volume of this pad",
-          0.0, 10.0, DEFAULT_PAD_VOLUME,
-          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
-  g_object_class_install_property (gobject_class, PROP_PAD_MUTE,
-      g_param_spec_boolean ("mute", "Mute", "Mute this pad",
-          DEFAULT_PAD_MUTE,
-          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
-}
-
-static void
-gst_audiomixer_pad_init (GstAudioMixerPad * pad)
-{
-  pad->volume = DEFAULT_PAD_VOLUME;
-  pad->mute = DEFAULT_PAD_MUTE;
-}
-
-enum
-{
-  PROP_0
-};
-
-/* These are the formats we can mix natively */
-
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define CAPS \
-  GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
-  ", layout = interleaved"
-#else
-#define CAPS \
-  GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
-  ", layout = interleaved"
-#endif
-
-static GstStaticPadTemplate gst_audiomixer_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (CAPS)
-    );
-
-#define SINK_CAPS \
-  GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
-      ", layout=interleaved")
-
-static GstStaticPadTemplate gst_audiomixer_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink_%u",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    SINK_CAPS);
-
-static void gst_audiomixer_child_proxy_init (gpointer g_iface,
-    gpointer iface_data);
-
-#define gst_audiomixer_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer,
-    GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
-        gst_audiomixer_child_proxy_init));
-
-static GstPad *gst_audiomixer_request_new_pad (GstElement * element,
-    GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps);
-static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad);
-
-static gboolean
-gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
-    GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
-    GstBuffer * outbuf, guint out_offset, guint num_samples);
-
-
-static void
-gst_audiomixer_class_init (GstAudioMixerClass * klass)
-{
-  GstElementClass *gstelement_class = (GstElementClass *) klass;
-  GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass;
-
-  gst_element_class_add_static_pad_template (gstelement_class,
-      &gst_audiomixer_src_template);
-  gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
-      &gst_audiomixer_sink_template, GST_TYPE_AUDIO_MIXER_PAD);
-  gst_element_class_set_static_metadata (gstelement_class, "AudioMixer",
-      "Generic/Audio", "Mixes multiple audio streams",
-      "Sebastian Dröge <sebastian@centricular.com>");
-
-  gstelement_class->request_new_pad =
-      GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad);
-  gstelement_class->release_pad =
-      GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad);
-
-  aagg_class->aggregate_one_buffer = gst_audiomixer_aggregate_one_buffer;
-}
-
-static void
-gst_audiomixer_init (GstAudioMixer * audiomixer)
-{
-}
-
-static GstPad *
-gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * templ,
-    const gchar * req_name, const GstCaps * caps)
-{
-  GstAudioMixerPad *newpad;
-
-  newpad = (GstAudioMixerPad *)
-      GST_ELEMENT_CLASS (parent_class)->request_new_pad (element,
-      templ, req_name, caps);
-
-  if (newpad == NULL)
-    goto could_not_create;
-
-  gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (newpad),
-      GST_OBJECT_NAME (newpad));
-
-  return GST_PAD_CAST (newpad);
-
-could_not_create:
-  {
-    GST_DEBUG_OBJECT (element, "could not create/add  pad");
-    return NULL;
-  }
-}
-
-static void
-gst_audiomixer_release_pad (GstElement * element, GstPad * pad)
-{
-  GstAudioMixer *audiomixer;
-
-  audiomixer = GST_AUDIO_MIXER (element);
-
-  GST_DEBUG_OBJECT (audiomixer, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
-
-  gst_child_proxy_child_removed (GST_CHILD_PROXY (audiomixer), G_OBJECT (pad),
-      GST_OBJECT_NAME (pad));
-
-  GST_ELEMENT_CLASS (parent_class)->release_pad (element, pad);
-}
-
-
-static gboolean
-gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
-    GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
-    GstBuffer * outbuf, guint out_offset, guint num_frames)
-{
-  GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aaggpad);
-  GstMapInfo inmap;
-  GstMapInfo outmap;
-  gint bpf;
-
-  GST_OBJECT_LOCK (aagg);
-  GST_OBJECT_LOCK (aaggpad);
-
-  if (pad->mute || pad->volume < G_MINDOUBLE) {
-    GST_DEBUG_OBJECT (pad, "Skipping muted pad");
-    GST_OBJECT_UNLOCK (aaggpad);
-    GST_OBJECT_UNLOCK (aagg);
-    return FALSE;
-  }
-
-  bpf = GST_AUDIO_INFO_BPF (&aagg->info);
-
-  gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
-  gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
-  GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u",
-      num_frames * bpf, out_offset * bpf, in_offset * bpf);
-
-  /* further buffers, need to add them */
-  if (pad->volume == 1.0) {
-    switch (aagg->info.finfo->format) {
-      case GST_AUDIO_FORMAT_U8:
-        audiomixer_orc_add_u8 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_S8:
-        audiomixer_orc_add_s8 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_U16:
-        audiomixer_orc_add_u16 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_S16:
-        audiomixer_orc_add_s16 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_U32:
-        audiomixer_orc_add_u32 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_S32:
-        audiomixer_orc_add_s32 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_F32:
-        audiomixer_orc_add_f32 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_F64:
-        audiomixer_orc_add_f64 ((gpointer) (outmap.data + out_offset * bpf),
-            (gpointer) (inmap.data + in_offset * bpf),
-            num_frames * aagg->info.channels);
-        break;
-      default:
-        g_assert_not_reached ();
-        break;
-    }
-  } else {
-    switch (aagg->info.finfo->format) {
-      case GST_AUDIO_FORMAT_U8:
-        audiomixer_orc_add_volume_u8 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume_i8, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_S8:
-        audiomixer_orc_add_volume_s8 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume_i8, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_U16:
-        audiomixer_orc_add_volume_u16 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume_i16, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_S16:
-        audiomixer_orc_add_volume_s16 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume_i16, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_U32:
-        audiomixer_orc_add_volume_u32 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume_i32, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_S32:
-        audiomixer_orc_add_volume_s32 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume_i32, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_F32:
-        audiomixer_orc_add_volume_f32 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume, num_frames * aagg->info.channels);
-        break;
-      case GST_AUDIO_FORMAT_F64:
-        audiomixer_orc_add_volume_f64 ((gpointer) (outmap.data +
-                out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
-            pad->volume, num_frames * aagg->info.channels);
-        break;
-      default:
-        g_assert_not_reached ();
-        break;
-    }
-  }
-  gst_buffer_unmap (inbuf, &inmap);
-  gst_buffer_unmap (outbuf, &outmap);
-
-  GST_OBJECT_UNLOCK (aaggpad);
-  GST_OBJECT_UNLOCK (aagg);
-
-  return TRUE;
-}
-
-
-/* GstChildProxy implementation */
-static GObject *
-gst_audiomixer_child_proxy_get_child_by_index (GstChildProxy * child_proxy,
-    guint index)
-{
-  GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy);
-  GObject *obj = NULL;
-
-  GST_OBJECT_LOCK (audiomixer);
-  obj = g_list_nth_data (GST_ELEMENT_CAST (audiomixer)->sinkpads, index);
-  if (obj)
-    gst_object_ref (obj);
-  GST_OBJECT_UNLOCK (audiomixer);
-
-  return obj;
-}
-
-static guint
-gst_audiomixer_child_proxy_get_children_count (GstChildProxy * child_proxy)
-{
-  guint count = 0;
-  GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy);
-
-  GST_OBJECT_LOCK (audiomixer);
-  count = GST_ELEMENT_CAST (audiomixer)->numsinkpads;
-  GST_OBJECT_UNLOCK (audiomixer);
-  GST_INFO_OBJECT (audiomixer, "Children Count: %d", count);
-
-  return count;
-}
-
-static void
-gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data)
-{
-  GstChildProxyInterface *iface = g_iface;
-
-  GST_INFO ("intializing child proxy interface");
-  iface->get_child_by_index = gst_audiomixer_child_proxy_get_child_by_index;
-  iface->get_children_count = gst_audiomixer_child_proxy_get_children_count;
-}
-
-/* Empty liveadder alias with non-zero latency */
-
-typedef GstAudioMixer GstLiveAdder;
-typedef GstAudioMixerClass GstLiveAdderClass;
-
-static GType gst_live_adder_get_type (void);
-#define GST_TYPE_LIVE_ADDER gst_live_adder_get_type ()
-
-G_DEFINE_TYPE (GstLiveAdder, gst_live_adder, GST_TYPE_AUDIO_MIXER);
-
-enum
-{
-  LIVEADDER_PROP_LATENCY = 1
-};
-
-static void
-gst_live_adder_init (GstLiveAdder * self)
-{
-}
-
-static void
-gst_live_adder_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  switch (prop_id) {
-    case LIVEADDER_PROP_LATENCY:
-    {
-      GParamSpec *parent_spec =
-          g_object_class_find_property (G_OBJECT_CLASS
-          (gst_live_adder_parent_class), "latency");
-      GObjectClass *pspec_class = g_type_class_peek (parent_spec->owner_type);
-      GValue v = { 0 };
-
-      g_value_init (&v, G_TYPE_UINT64);
-
-      g_value_set_uint64 (&v, g_value_get_uint (value) * GST_MSECOND);
-
-      G_OBJECT_CLASS (pspec_class)->set_property (object,
-          parent_spec->param_id, &v, parent_spec);
-      break;
-    }
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_live_adder_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
-{
-  switch (prop_id) {
-    case LIVEADDER_PROP_LATENCY:
-    {
-      GParamSpec *parent_spec =
-          g_object_class_find_property (G_OBJECT_CLASS
-          (gst_live_adder_parent_class), "latency");
-      GObjectClass *pspec_class = g_type_class_peek (parent_spec->owner_type);
-      GValue v = { 0 };
-
-      g_value_init (&v, G_TYPE_UINT64);
-
-      G_OBJECT_CLASS (pspec_class)->get_property (object,
-          parent_spec->param_id, &v, parent_spec);
-
-      g_value_set_uint (value, g_value_get_uint64 (&v) / GST_MSECOND);
-      break;
-    }
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-
-static void
-gst_live_adder_class_init (GstLiveAdderClass * klass)
-{
-  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
-
-  gobject_class->set_property = gst_live_adder_set_property;
-  gobject_class->get_property = gst_live_adder_get_property;
-
-  g_object_class_install_property (gobject_class, LIVEADDER_PROP_LATENCY,
-      g_param_spec_uint ("latency", "Buffer latency",
-          "Additional latency in live mode to allow upstream "
-          "to take longer to produce buffers for the current "
-          "position (in milliseconds)", 0, G_MAXUINT,
-          30, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT));
-}
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiomixer", 0,
-      "audio mixing element");
-
-  if (!gst_element_register (plugin, "audiomixer", GST_RANK_NONE,
-          GST_TYPE_AUDIO_MIXER))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "liveadder", GST_RANK_NONE,
-          GST_TYPE_LIVE_ADDER))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "audiointerleave", GST_RANK_NONE,
-          GST_TYPE_AUDIO_INTERLEAVE))
-    return FALSE;
-
-  return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    audiomixer,
-    "Mixes multiple audio streams",
-    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/gst/audiomixer/gstaudiomixer.h b/gst/audiomixer/gstaudiomixer.h
deleted file mode 100644
index 67ccb27e6..000000000
--- a/gst/audiomixer/gstaudiomixer.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2000 Wim Taymans <wtay@chello.be>
- * Copyright (C) 2013      Sebastian Dröge <slomo@circular-chaos.org>
- *
- * gstaudiomixer.h: Header for GstAudioMixer element
- *
- * 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_AUDIO_MIXER_H__
-#define __GST_AUDIO_MIXER_H__
-
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-#include <gst/audio/gstaudioaggregator.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AUDIO_MIXER            (gst_audiomixer_get_type())
-#define GST_AUDIO_MIXER(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_MIXER,GstAudioMixer))
-#define GST_IS_AUDIO_MIXER(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_MIXER))
-#define GST_AUDIO_MIXER_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_MIXER,GstAudioMixerClass))
-#define GST_IS_AUDIO_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_MIXER))
-#define GST_AUDIO_MIXER_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER,GstAudioMixerClass))
-
-typedef struct _GstAudioMixer             GstAudioMixer;
-typedef struct _GstAudioMixerClass        GstAudioMixerClass;
-
-typedef struct _GstAudioMixerPad GstAudioMixerPad;
-typedef struct _GstAudioMixerPadClass GstAudioMixerPadClass;
-
-/**
- * GstAudioMixer:
- *
- * The audiomixer object structure.
- */
-struct _GstAudioMixer {
-  GstAudioAggregator element;
-};
-
-struct _GstAudioMixerClass {
-  GstAudioAggregatorClass parent_class;
-};
-
-GType    gst_audiomixer_get_type (void);
-
-#define GST_TYPE_AUDIO_MIXER_PAD            (gst_audiomixer_pad_get_type())
-#define GST_AUDIO_MIXER_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPad))
-#define GST_IS_AUDIO_MIXER_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_MIXER_PAD))
-#define GST_AUDIO_MIXER_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass))
-#define GST_IS_AUDIO_MIXER_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_MIXER_PAD))
-#define GST_AUDIO_MIXER_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass))
-
-struct _GstAudioMixerPad {
-  GstAudioAggregatorConvertPad parent;
-
-  gdouble volume;
-  gint volume_i32;
-  gint volume_i16;
-  gint volume_i8;
-  gboolean mute;
-};
-
-struct _GstAudioMixerPadClass {
-  GstAudioAggregatorConvertPadClass parent_class;
-};
-
-GType gst_audiomixer_pad_get_type (void);
-
-G_END_DECLS
-
-
-#endif /* __GST_AUDIO_MIXER_H__ */
diff --git a/gst/audiomixer/gstaudiomixerorc-dist.c b/gst/audiomixer/gstaudiomixerorc-dist.c
deleted file mode 100644
index be377f705..000000000
--- a/gst/audiomixer/gstaudiomixerorc-dist.c
+++ /dev/null
@@ -1,2605 +0,0 @@
-
-/* autogenerated from gstaudiomixerorc.orc */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-#include <glib.h>
-
-#ifndef _ORC_INTEGER_TYPEDEFS_
-#define _ORC_INTEGER_TYPEDEFS_
-#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
-#include <stdint.h>
-typedef int8_t orc_int8;
-typedef int16_t orc_int16;
-typedef int32_t orc_int32;
-typedef int64_t orc_int64;
-typedef uint8_t orc_uint8;
-typedef uint16_t orc_uint16;
-typedef uint32_t orc_uint32;
-typedef uint64_t orc_uint64;
-#define ORC_UINT64_C(x) UINT64_C(x)
-#elif defined(_MSC_VER)
-typedef signed __int8 orc_int8;
-typedef signed __int16 orc_int16;
-typedef signed __int32 orc_int32;
-typedef signed __int64 orc_int64;
-typedef unsigned __int8 orc_uint8;
-typedef unsigned __int16 orc_uint16;
-typedef unsigned __int32 orc_uint32;
-typedef unsigned __int64 orc_uint64;
-#define ORC_UINT64_C(x) (x##Ui64)
-#define inline __inline
-#else
-#include <limits.h>
-typedef signed char orc_int8;
-typedef short orc_int16;
-typedef int orc_int32;
-typedef unsigned char orc_uint8;
-typedef unsigned short orc_uint16;
-typedef unsigned int orc_uint32;
-#if INT_MAX == LONG_MAX
-typedef long long orc_int64;
-typedef unsigned long long orc_uint64;
-#define ORC_UINT64_C(x) (x##ULL)
-#else
-typedef long orc_int64;
-typedef unsigned long orc_uint64;
-#define ORC_UINT64_C(x) (x##UL)
-#endif
-#endif
-typedef union
-{
-  orc_int16 i;
-  orc_int8 x2[2];
-} orc_union16;
-typedef union
-{
-  orc_int32 i;
-  float f;
-  orc_int16 x2[2];
-  orc_int8 x4[4];
-} orc_union32;
-typedef union
-{
-  orc_int64 i;
-  double f;
-  orc_int32 x2[2];
-  float x2f[2];
-  orc_int16 x4[4];
-} orc_union64;
-#endif
-#ifndef ORC_RESTRICT
-#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
-#define ORC_RESTRICT restrict
-#elif defined(__GNUC__) && __GNUC__ >= 4
-#define ORC_RESTRICT __restrict__
-#else
-#define ORC_RESTRICT
-#endif
-#endif
-
-#ifndef ORC_INTERNAL
-#if defined(__SUNPRO_C) && (__SUNPRO_C >= 0x590)
-#define ORC_INTERNAL __attribute__((visibility("hidden")))
-#elif defined(__SUNPRO_C) && (__SUNPRO_C >= 0x550)
-#define ORC_INTERNAL __hidden
-#elif defined (__GNUC__)
-#define ORC_INTERNAL __attribute__((visibility("hidden")))
-#else
-#define ORC_INTERNAL
-#endif
-#endif
-
-
-#ifndef DISABLE_ORC
-#include <orc/orc.h>
-#endif
-void audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1,
-    const gint32 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1,
-    const gint16 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1,
-    const gint8 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1,
-    const guint32 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1,
-    const guint16 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1,
-    const guint8 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_f32 (float *ORC_RESTRICT d1,
-    const float *ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_f64 (double *ORC_RESTRICT d1,
-    const double *ORC_RESTRICT s1, int n);
-void audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n);
-void audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1,
-    const guint8 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1,
-    const gint8 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1,
-    const guint16 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1,
-    const gint16 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1,
-    const guint32 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1,
-    const gint32 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_f32 (float *ORC_RESTRICT d1,
-    const float *ORC_RESTRICT s1, float p1, int n);
-void audiomixer_orc_add_volume_f64 (double *ORC_RESTRICT d1,
-    const double *ORC_RESTRICT s1, double p1, int n);
-
-
-/* begin Orc C target preamble */
-#define ORC_CLAMP(x,a,b) ((x)<(a) ? (a) : ((x)>(b) ? (b) : (x)))
-#define ORC_ABS(a) ((a)<0 ? -(a) : (a))
-#define ORC_MIN(a,b) ((a)<(b) ? (a) : (b))
-#define ORC_MAX(a,b) ((a)>(b) ? (a) : (b))
-#define ORC_SB_MAX 127
-#define ORC_SB_MIN (-1-ORC_SB_MAX)
-#define ORC_UB_MAX 255
-#define ORC_UB_MIN 0
-#define ORC_SW_MAX 32767
-#define ORC_SW_MIN (-1-ORC_SW_MAX)
-#define ORC_UW_MAX 65535
-#define ORC_UW_MIN 0
-#define ORC_SL_MAX 2147483647
-#define ORC_SL_MIN (-1-ORC_SL_MAX)
-#define ORC_UL_MAX 4294967295U
-#define ORC_UL_MIN 0
-#define ORC_CLAMP_SB(x) ORC_CLAMP(x,ORC_SB_MIN,ORC_SB_MAX)
-#define ORC_CLAMP_UB(x) ORC_CLAMP(x,ORC_UB_MIN,ORC_UB_MAX)
-#define ORC_CLAMP_SW(x) ORC_CLAMP(x,ORC_SW_MIN,ORC_SW_MAX)
-#define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX)
-#define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX)
-#define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX)
-#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8))
-#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24))
-#define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56))
-#define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset)))
-#define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff))
-#define ORC_ISNAN(x) ((((x)&0x7f800000) == 0x7f800000) && (((x)&0x007fffff) != 0))
-#define ORC_DENORMAL_DOUBLE(x) ((x) & ((((x)&ORC_UINT64_C(0x7ff0000000000000)) == 0) ? ORC_UINT64_C(0xfff0000000000000) : ORC_UINT64_C(0xffffffffffffffff)))
-#define ORC_ISNAN_DOUBLE(x) ((((x)&ORC_UINT64_C(0x7ff0000000000000)) == ORC_UINT64_C(0x7ff0000000000000)) && (((x)&ORC_UINT64_C(0x000fffffffffffff)) != 0))
-#ifndef ORC_RESTRICT
-#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
-#define ORC_RESTRICT restrict
-#elif defined(__GNUC__) && __GNUC__ >= 4
-#define ORC_RESTRICT __restrict__
-#else
-#define ORC_RESTRICT
-#endif
-#endif
-/* end Orc C target preamble */
-
-
-
-/* audiomixer_orc_add_s32 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1,
-    const gint32 * ORC_RESTRICT s1, int n)
-{
-  int i;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var32;
-  orc_union32 var33;
-  orc_union32 var34;
-
-  ptr0 = (orc_union32 *) d1;
-  ptr4 = (orc_union32 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var32 = ptr0[i];
-    /* 1: loadl */
-    var33 = ptr4[i];
-    /* 2: addssl */
-    var34.i = ORC_CLAMP_SL ((orc_int64) var32.i + (orc_int64) var33.i);
-    /* 3: storel */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_s32 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var32;
-  orc_union32 var33;
-  orc_union32 var34;
-
-  ptr0 = (orc_union32 *) ex->arrays[0];
-  ptr4 = (orc_union32 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var32 = ptr0[i];
-    /* 1: loadl */
-    var33 = ptr4[i];
-    /* 2: addssl */
-    var34.i = ORC_CLAMP_SL ((orc_int64) var32.i + (orc_int64) var33.i);
-    /* 3: storel */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1,
-    const gint32 * ORC_RESTRICT s1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 115, 51, 50, 11, 4, 4, 12, 4, 4, 104,
-        0, 0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s32);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_s32");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s32);
-      orc_program_add_destination (p, 4, "d1");
-      orc_program_add_source (p, 4, "s1");
-
-      orc_program_append_2 (p, "addssl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_s16 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1,
-    const gint16 * ORC_RESTRICT s1, int n)
-{
-  int i;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var32;
-  orc_union16 var33;
-  orc_union16 var34;
-
-  ptr0 = (orc_union16 *) d1;
-  ptr4 = (orc_union16 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var32 = ptr0[i];
-    /* 1: loadw */
-    var33 = ptr4[i];
-    /* 2: addssw */
-    var34.i = ORC_CLAMP_SW (var32.i + var33.i);
-    /* 3: storew */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_s16 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var32;
-  orc_union16 var33;
-  orc_union16 var34;
-
-  ptr0 = (orc_union16 *) ex->arrays[0];
-  ptr4 = (orc_union16 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var32 = ptr0[i];
-    /* 1: loadw */
-    var33 = ptr4[i];
-    /* 2: addssw */
-    var34.i = ORC_CLAMP_SW (var32.i + var33.i);
-    /* 3: storew */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1,
-    const gint16 * ORC_RESTRICT s1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 115, 49, 54, 11, 2, 2, 12, 2, 2, 71,
-        0, 0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s16);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_s16");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s16);
-      orc_program_add_destination (p, 2, "d1");
-      orc_program_add_source (p, 2, "s1");
-
-      orc_program_append_2 (p, "addssw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_s8 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1,
-    int n)
-{
-  int i;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var32;
-  orc_int8 var33;
-  orc_int8 var34;
-
-  ptr0 = (orc_int8 *) d1;
-  ptr4 = (orc_int8 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var32 = ptr0[i];
-    /* 1: loadb */
-    var33 = ptr4[i];
-    /* 2: addssb */
-    var34 = ORC_CLAMP_SB (var32 + var33);
-    /* 3: storeb */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_s8 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var32;
-  orc_int8 var33;
-  orc_int8 var34;
-
-  ptr0 = (orc_int8 *) ex->arrays[0];
-  ptr4 = (orc_int8 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var32 = ptr0[i];
-    /* 1: loadb */
-    var33 = ptr4[i];
-    /* 2: addssb */
-    var34 = ORC_CLAMP_SB (var32 + var33);
-    /* 3: storeb */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1,
-    int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 21, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 115, 56, 11, 1, 1, 12, 1, 1, 34, 0,
-        0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s8);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_s8");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s8);
-      orc_program_add_destination (p, 1, "d1");
-      orc_program_add_source (p, 1, "s1");
-
-      orc_program_append_2 (p, "addssb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_u32 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1,
-    const guint32 * ORC_RESTRICT s1, int n)
-{
-  int i;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var32;
-  orc_union32 var33;
-  orc_union32 var34;
-
-  ptr0 = (orc_union32 *) d1;
-  ptr4 = (orc_union32 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var32 = ptr0[i];
-    /* 1: loadl */
-    var33 = ptr4[i];
-    /* 2: addusl */
-    var34.i =
-        ORC_CLAMP_UL ((orc_int64) (orc_uint32) var32.i +
-        (orc_int64) (orc_uint32) var33.i);
-    /* 3: storel */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_u32 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var32;
-  orc_union32 var33;
-  orc_union32 var34;
-
-  ptr0 = (orc_union32 *) ex->arrays[0];
-  ptr4 = (orc_union32 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var32 = ptr0[i];
-    /* 1: loadl */
-    var33 = ptr4[i];
-    /* 2: addusl */
-    var34.i =
-        ORC_CLAMP_UL ((orc_int64) (orc_uint32) var32.i +
-        (orc_int64) (orc_uint32) var33.i);
-    /* 3: storel */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1,
-    const guint32 * ORC_RESTRICT s1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 117, 51, 50, 11, 4, 4, 12, 4, 4, 105,
-        0, 0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u32);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_u32");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u32);
-      orc_program_add_destination (p, 4, "d1");
-      orc_program_add_source (p, 4, "s1");
-
-      orc_program_append_2 (p, "addusl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_u16 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1,
-    const guint16 * ORC_RESTRICT s1, int n)
-{
-  int i;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var32;
-  orc_union16 var33;
-  orc_union16 var34;
-
-  ptr0 = (orc_union16 *) d1;
-  ptr4 = (orc_union16 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var32 = ptr0[i];
-    /* 1: loadw */
-    var33 = ptr4[i];
-    /* 2: addusw */
-    var34.i = ORC_CLAMP_UW ((orc_uint16) var32.i + (orc_uint16) var33.i);
-    /* 3: storew */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_u16 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var32;
-  orc_union16 var33;
-  orc_union16 var34;
-
-  ptr0 = (orc_union16 *) ex->arrays[0];
-  ptr4 = (orc_union16 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var32 = ptr0[i];
-    /* 1: loadw */
-    var33 = ptr4[i];
-    /* 2: addusw */
-    var34.i = ORC_CLAMP_UW ((orc_uint16) var32.i + (orc_uint16) var33.i);
-    /* 3: storew */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1,
-    const guint16 * ORC_RESTRICT s1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 117, 49, 54, 11, 2, 2, 12, 2, 2, 72,
-        0, 0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u16);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_u16");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u16);
-      orc_program_add_destination (p, 2, "d1");
-      orc_program_add_source (p, 2, "s1");
-
-      orc_program_append_2 (p, "addusw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_u8 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1,
-    int n)
-{
-  int i;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var32;
-  orc_int8 var33;
-  orc_int8 var34;
-
-  ptr0 = (orc_int8 *) d1;
-  ptr4 = (orc_int8 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var32 = ptr0[i];
-    /* 1: loadb */
-    var33 = ptr4[i];
-    /* 2: addusb */
-    var34 = ORC_CLAMP_UB ((orc_uint8) var32 + (orc_uint8) var33);
-    /* 3: storeb */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_u8 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var32;
-  orc_int8 var33;
-  orc_int8 var34;
-
-  ptr0 = (orc_int8 *) ex->arrays[0];
-  ptr4 = (orc_int8 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var32 = ptr0[i];
-    /* 1: loadb */
-    var33 = ptr4[i];
-    /* 2: addusb */
-    var34 = ORC_CLAMP_UB ((orc_uint8) var32 + (orc_uint8) var33);
-    /* 3: storeb */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1,
-    int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 21, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 117, 56, 11, 1, 1, 12, 1, 1, 35, 0,
-        0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u8);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_u8");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u8);
-      orc_program_add_destination (p, 1, "d1");
-      orc_program_add_source (p, 1, "s1");
-
-      orc_program_append_2 (p, "addusb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_f32 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_f32 (float *ORC_RESTRICT d1, const float *ORC_RESTRICT s1,
-    int n)
-{
-  int i;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var32;
-  orc_union32 var33;
-  orc_union32 var34;
-
-  ptr0 = (orc_union32 *) d1;
-  ptr4 = (orc_union32 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var32 = ptr0[i];
-    /* 1: loadl */
-    var33 = ptr4[i];
-    /* 2: addf */
-    {
-      orc_union32 _src1;
-      orc_union32 _src2;
-      orc_union32 _dest1;
-      _src1.i = ORC_DENORMAL (var32.i);
-      _src2.i = ORC_DENORMAL (var33.i);
-      _dest1.f = _src1.f + _src2.f;
-      var34.i = ORC_DENORMAL (_dest1.i);
-    }
-    /* 3: storel */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_f32 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var32;
-  orc_union32 var33;
-  orc_union32 var34;
-
-  ptr0 = (orc_union32 *) ex->arrays[0];
-  ptr4 = (orc_union32 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var32 = ptr0[i];
-    /* 1: loadl */
-    var33 = ptr4[i];
-    /* 2: addf */
-    {
-      orc_union32 _src1;
-      orc_union32 _src2;
-      orc_union32 _dest1;
-      _src1.i = ORC_DENORMAL (var32.i);
-      _src2.i = ORC_DENORMAL (var33.i);
-      _dest1.f = _src1.f + _src2.f;
-      var34.i = ORC_DENORMAL (_dest1.i);
-    }
-    /* 3: storel */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_f32 (float *ORC_RESTRICT d1, const float *ORC_RESTRICT s1,
-    int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 102, 51, 50, 11, 4, 4, 12, 4, 4, 200,
-        0, 0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f32);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_f32");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f32);
-      orc_program_add_destination (p, 4, "d1");
-      orc_program_add_source (p, 4, "s1");
-
-      orc_program_append_2 (p, "addf", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_f64 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_f64 (double *ORC_RESTRICT d1, const double *ORC_RESTRICT s1,
-    int n)
-{
-  int i;
-  orc_union64 *ORC_RESTRICT ptr0;
-  const orc_union64 *ORC_RESTRICT ptr4;
-  orc_union64 var32;
-  orc_union64 var33;
-  orc_union64 var34;
-
-  ptr0 = (orc_union64 *) d1;
-  ptr4 = (orc_union64 *) s1;
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadq */
-    var32 = ptr0[i];
-    /* 1: loadq */
-    var33 = ptr4[i];
-    /* 2: addd */
-    {
-      orc_union64 _src1;
-      orc_union64 _src2;
-      orc_union64 _dest1;
-      _src1.i = ORC_DENORMAL_DOUBLE (var32.i);
-      _src2.i = ORC_DENORMAL_DOUBLE (var33.i);
-      _dest1.f = _src1.f + _src2.f;
-      var34.i = ORC_DENORMAL_DOUBLE (_dest1.i);
-    }
-    /* 3: storeq */
-    ptr0[i] = var34;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_f64 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union64 *ORC_RESTRICT ptr0;
-  const orc_union64 *ORC_RESTRICT ptr4;
-  orc_union64 var32;
-  orc_union64 var33;
-  orc_union64 var34;
-
-  ptr0 = (orc_union64 *) ex->arrays[0];
-  ptr4 = (orc_union64 *) ex->arrays[4];
-
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadq */
-    var32 = ptr0[i];
-    /* 1: loadq */
-    var33 = ptr4[i];
-    /* 2: addd */
-    {
-      orc_union64 _src1;
-      orc_union64 _src2;
-      orc_union64 _dest1;
-      _src1.i = ORC_DENORMAL_DOUBLE (var32.i);
-      _src2.i = ORC_DENORMAL_DOUBLE (var33.i);
-      _dest1.f = _src1.f + _src2.f;
-      var34.i = ORC_DENORMAL_DOUBLE (_dest1.i);
-    }
-    /* 3: storeq */
-    ptr0[i] = var34;
-  }
-
-}
-
-void
-audiomixer_orc_add_f64 (double *ORC_RESTRICT d1, const double *ORC_RESTRICT s1,
-    int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 102, 54, 52, 11, 8, 8, 12, 8, 8, 212,
-        0, 0, 4, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f64);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_f64");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f64);
-      orc_program_add_destination (p, 8, "d1");
-      orc_program_add_source (p, 8, "s1");
-
-      orc_program_append_2 (p, "addd", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_volume_u8 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n)
-{
-  int i;
-  orc_int8 *ORC_RESTRICT ptr0;
-  orc_int8 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_int8 var35;
-#else
-  orc_int8 var35;
-#endif
-  orc_int8 var36;
-  orc_int8 var37;
-  orc_int8 var38;
-  orc_union16 var39;
-  orc_union16 var40;
-  orc_int8 var41;
-
-  ptr0 = (orc_int8 *) d1;
-
-  /* 1: loadpb */
-  var35 = (int) 0x00000080;     /* 128 or 6.32404e-322f */
-  /* 3: loadpb */
-  var36 = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var34 = ptr0[i];
-    /* 2: xorb */
-    var38 = var34 ^ var35;
-    /* 4: mulsbw */
-    var39.i = var38 * var36;
-    /* 5: shrsw */
-    var40.i = var39.i >> 3;
-    /* 6: convssswb */
-    var41 = ORC_CLAMP_SB (var40.i);
-    /* 7: xorb */
-    var37 = var41 ^ var35;
-    /* 8: storeb */
-    ptr0[i] = var37;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_volume_u8 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_int8 *ORC_RESTRICT ptr0;
-  orc_int8 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_int8 var35;
-#else
-  orc_int8 var35;
-#endif
-  orc_int8 var36;
-  orc_int8 var37;
-  orc_int8 var38;
-  orc_union16 var39;
-  orc_union16 var40;
-  orc_int8 var41;
-
-  ptr0 = (orc_int8 *) ex->arrays[0];
-
-  /* 1: loadpb */
-  var35 = (int) 0x00000080;     /* 128 or 6.32404e-322f */
-  /* 3: loadpb */
-  var36 = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var34 = ptr0[i];
-    /* 2: xorb */
-    var38 = var34 ^ var35;
-    /* 4: mulsbw */
-    var39.i = var38 * var36;
-    /* 5: shrsw */
-    var40.i = var39.i >> 3;
-    /* 6: convssswb */
-    var41 = ORC_CLAMP_SB (var40.i);
-    /* 7: xorb */
-    var37 = var41 ^ var35;
-    /* 8: storeb */
-    ptr0[i] = var37;
-  }
-
-}
-
-void
-audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 24, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 118, 111, 108, 117, 109, 101, 95, 117, 56, 11, 1, 1, 14, 1,
-        128, 0, 0, 0, 14, 2, 3, 0, 0, 0, 16, 1, 20, 2, 20, 1,
-        68, 33, 0, 16, 174, 32, 33, 24, 94, 32, 32, 17, 159, 33, 32, 68,
-        0, 33, 16, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_volume_u8);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_volume_u8");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_volume_u8);
-      orc_program_add_destination (p, 1, "d1");
-      orc_program_add_constant (p, 1, 0x00000080, "c1");
-      orc_program_add_constant (p, 2, 0x00000003, "c2");
-      orc_program_add_parameter (p, 1, "p1");
-      orc_program_add_temporary (p, 2, "t1");
-      orc_program_add_temporary (p, 1, "t2");
-
-      orc_program_append_2 (p, "xorb", 0, ORC_VAR_T2, ORC_VAR_D1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "mulsbw", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsw", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convssswb", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "xorb", 0, ORC_VAR_D1, ORC_VAR_T2, ORC_VAR_C1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_u8 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1,
-    const guint8 * ORC_RESTRICT s1, int p1, int n)
-{
-  int i;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_int8 var35;
-#else
-  orc_int8 var35;
-#endif
-  orc_int8 var36;
-  orc_int8 var37;
-  orc_int8 var38;
-  orc_int8 var39;
-  orc_union16 var40;
-  orc_union16 var41;
-  orc_int8 var42;
-  orc_int8 var43;
-
-  ptr0 = (orc_int8 *) d1;
-  ptr4 = (orc_int8 *) s1;
-
-  /* 1: loadpb */
-  var35 = (int) 0x00000080;     /* 128 or 6.32404e-322f */
-  /* 3: loadpb */
-  var36 = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var34 = ptr4[i];
-    /* 2: xorb */
-    var39 = var34 ^ var35;
-    /* 4: mulsbw */
-    var40.i = var39 * var36;
-    /* 5: shrsw */
-    var41.i = var40.i >> 3;
-    /* 6: convssswb */
-    var42 = ORC_CLAMP_SB (var41.i);
-    /* 7: xorb */
-    var43 = var42 ^ var35;
-    /* 8: loadb */
-    var37 = ptr0[i];
-    /* 9: addusb */
-    var38 = ORC_CLAMP_UB ((orc_uint8) var37 + (orc_uint8) var43);
-    /* 10: storeb */
-    ptr0[i] = var38;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_u8 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_int8 var35;
-#else
-  orc_int8 var35;
-#endif
-  orc_int8 var36;
-  orc_int8 var37;
-  orc_int8 var38;
-  orc_int8 var39;
-  orc_union16 var40;
-  orc_union16 var41;
-  orc_int8 var42;
-  orc_int8 var43;
-
-  ptr0 = (orc_int8 *) ex->arrays[0];
-  ptr4 = (orc_int8 *) ex->arrays[4];
-
-  /* 1: loadpb */
-  var35 = (int) 0x00000080;     /* 128 or 6.32404e-322f */
-  /* 3: loadpb */
-  var36 = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var34 = ptr4[i];
-    /* 2: xorb */
-    var39 = var34 ^ var35;
-    /* 4: mulsbw */
-    var40.i = var39 * var36;
-    /* 5: shrsw */
-    var41.i = var40.i >> 3;
-    /* 6: convssswb */
-    var42 = ORC_CLAMP_SB (var41.i);
-    /* 7: xorb */
-    var43 = var42 ^ var35;
-    /* 8: loadb */
-    var37 = ptr0[i];
-    /* 9: addusb */
-    var38 = ORC_CLAMP_UB ((orc_uint8) var37 + (orc_uint8) var43);
-    /* 10: storeb */
-    ptr0[i] = var38;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1,
-    const guint8 * ORC_RESTRICT s1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 28, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 117, 56, 11,
-        1, 1, 12, 1, 1, 14, 1, 128, 0, 0, 0, 14, 2, 3, 0, 0,
-        0, 16, 1, 20, 2, 20, 1, 68, 33, 4, 16, 174, 32, 33, 24, 94,
-        32, 32, 17, 159, 33, 32, 68, 33, 33, 16, 35, 0, 0, 33, 2, 0,
-
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_u8);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_u8");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_u8);
-      orc_program_add_destination (p, 1, "d1");
-      orc_program_add_source (p, 1, "s1");
-      orc_program_add_constant (p, 1, 0x00000080, "c1");
-      orc_program_add_constant (p, 2, 0x00000003, "c2");
-      orc_program_add_parameter (p, 1, "p1");
-      orc_program_add_temporary (p, 2, "t1");
-      orc_program_add_temporary (p, 1, "t2");
-
-      orc_program_append_2 (p, "xorb", 0, ORC_VAR_T2, ORC_VAR_S1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "mulsbw", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsw", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convssswb", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "xorb", 0, ORC_VAR_T2, ORC_VAR_T2, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "addusb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_s8 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1,
-    const gint8 * ORC_RESTRICT s1, int p1, int n)
-{
-  int i;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var34;
-  orc_int8 var35;
-  orc_int8 var36;
-  orc_int8 var37;
-  orc_union16 var38;
-  orc_union16 var39;
-  orc_int8 var40;
-
-  ptr0 = (orc_int8 *) d1;
-  ptr4 = (orc_int8 *) s1;
-
-  /* 1: loadpb */
-  var35 = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var34 = ptr4[i];
-    /* 2: mulsbw */
-    var38.i = var34 * var35;
-    /* 3: shrsw */
-    var39.i = var38.i >> 3;
-    /* 4: convssswb */
-    var40 = ORC_CLAMP_SB (var39.i);
-    /* 5: loadb */
-    var36 = ptr0[i];
-    /* 6: addssb */
-    var37 = ORC_CLAMP_SB (var36 + var40);
-    /* 7: storeb */
-    ptr0[i] = var37;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_s8 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_int8 *ORC_RESTRICT ptr0;
-  const orc_int8 *ORC_RESTRICT ptr4;
-  orc_int8 var34;
-  orc_int8 var35;
-  orc_int8 var36;
-  orc_int8 var37;
-  orc_union16 var38;
-  orc_union16 var39;
-  orc_int8 var40;
-
-  ptr0 = (orc_int8 *) ex->arrays[0];
-  ptr4 = (orc_int8 *) ex->arrays[4];
-
-  /* 1: loadpb */
-  var35 = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadb */
-    var34 = ptr4[i];
-    /* 2: mulsbw */
-    var38.i = var34 * var35;
-    /* 3: shrsw */
-    var39.i = var38.i >> 3;
-    /* 4: convssswb */
-    var40 = ORC_CLAMP_SB (var39.i);
-    /* 5: loadb */
-    var36 = ptr0[i];
-    /* 6: addssb */
-    var37 = ORC_CLAMP_SB (var36 + var40);
-    /* 7: storeb */
-    ptr0[i] = var37;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1,
-    const gint8 * ORC_RESTRICT s1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 28, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 115, 56, 11,
-        1, 1, 12, 1, 1, 14, 2, 3, 0, 0, 0, 16, 1, 20, 2, 20,
-        1, 174, 32, 4, 24, 94, 32, 32, 16, 159, 33, 32, 34, 0, 0, 33,
-        2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_s8);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_s8");
-      orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_s8);
-      orc_program_add_destination (p, 1, "d1");
-      orc_program_add_source (p, 1, "s1");
-      orc_program_add_constant (p, 2, 0x00000003, "c1");
-      orc_program_add_parameter (p, 1, "p1");
-      orc_program_add_temporary (p, 2, "t1");
-      orc_program_add_temporary (p, 1, "t2");
-
-      orc_program_append_2 (p, "mulsbw", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsw", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convssswb", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "addssb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_u16 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1,
-    const guint16 * ORC_RESTRICT s1, int p1, int n)
-{
-  int i;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_union16 var35;
-#else
-  orc_union16 var35;
-#endif
-  orc_union16 var36;
-  orc_union16 var37;
-  orc_union16 var38;
-  orc_union16 var39;
-  orc_union32 var40;
-  orc_union32 var41;
-  orc_union16 var42;
-  orc_union16 var43;
-
-  ptr0 = (orc_union16 *) d1;
-  ptr4 = (orc_union16 *) s1;
-
-  /* 1: loadpw */
-  var35.i = (int) 0x00008000;   /* 32768 or 1.61895e-319f */
-  /* 3: loadpw */
-  var36.i = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var34 = ptr4[i];
-    /* 2: xorw */
-    var39.i = var34.i ^ var35.i;
-    /* 4: mulswl */
-    var40.i = var39.i * var36.i;
-    /* 5: shrsl */
-    var41.i = var40.i >> 11;
-    /* 6: convssslw */
-    var42.i = ORC_CLAMP_SW (var41.i);
-    /* 7: xorw */
-    var43.i = var42.i ^ var35.i;
-    /* 8: loadw */
-    var37 = ptr0[i];
-    /* 9: addusw */
-    var38.i = ORC_CLAMP_UW ((orc_uint16) var37.i + (orc_uint16) var43.i);
-    /* 10: storew */
-    ptr0[i] = var38;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_u16 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_union16 var35;
-#else
-  orc_union16 var35;
-#endif
-  orc_union16 var36;
-  orc_union16 var37;
-  orc_union16 var38;
-  orc_union16 var39;
-  orc_union32 var40;
-  orc_union32 var41;
-  orc_union16 var42;
-  orc_union16 var43;
-
-  ptr0 = (orc_union16 *) ex->arrays[0];
-  ptr4 = (orc_union16 *) ex->arrays[4];
-
-  /* 1: loadpw */
-  var35.i = (int) 0x00008000;   /* 32768 or 1.61895e-319f */
-  /* 3: loadpw */
-  var36.i = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var34 = ptr4[i];
-    /* 2: xorw */
-    var39.i = var34.i ^ var35.i;
-    /* 4: mulswl */
-    var40.i = var39.i * var36.i;
-    /* 5: shrsl */
-    var41.i = var40.i >> 11;
-    /* 6: convssslw */
-    var42.i = ORC_CLAMP_SW (var41.i);
-    /* 7: xorw */
-    var43.i = var42.i ^ var35.i;
-    /* 8: loadw */
-    var37 = ptr0[i];
-    /* 9: addusw */
-    var38.i = ORC_CLAMP_UW ((orc_uint16) var37.i + (orc_uint16) var43.i);
-    /* 10: storew */
-    ptr0[i] = var38;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1,
-    const guint16 * ORC_RESTRICT s1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 117, 49, 54,
-        11, 2, 2, 12, 2, 2, 14, 2, 0, 128, 0, 0, 14, 4, 11, 0,
-        0, 0, 16, 2, 20, 4, 20, 2, 101, 33, 4, 16, 176, 32, 33, 24,
-        125, 32, 32, 17, 165, 33, 32, 101, 33, 33, 16, 72, 0, 0, 33, 2,
-        0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_u16);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_u16");
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_u16);
-      orc_program_add_destination (p, 2, "d1");
-      orc_program_add_source (p, 2, "s1");
-      orc_program_add_constant (p, 2, 0x00008000, "c1");
-      orc_program_add_constant (p, 4, 0x0000000b, "c2");
-      orc_program_add_parameter (p, 2, "p1");
-      orc_program_add_temporary (p, 4, "t1");
-      orc_program_add_temporary (p, 2, "t2");
-
-      orc_program_append_2 (p, "xorw", 0, ORC_VAR_T2, ORC_VAR_S1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "mulswl", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsl", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convssslw", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "xorw", 0, ORC_VAR_T2, ORC_VAR_T2, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "addusw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_s16 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1,
-    const gint16 * ORC_RESTRICT s1, int p1, int n)
-{
-  int i;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var34;
-  orc_union16 var35;
-  orc_union16 var36;
-  orc_union16 var37;
-  orc_union32 var38;
-  orc_union32 var39;
-  orc_union16 var40;
-
-  ptr0 = (orc_union16 *) d1;
-  ptr4 = (orc_union16 *) s1;
-
-  /* 1: loadpw */
-  var35.i = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var34 = ptr4[i];
-    /* 2: mulswl */
-    var38.i = var34.i * var35.i;
-    /* 3: shrsl */
-    var39.i = var38.i >> 11;
-    /* 4: convssslw */
-    var40.i = ORC_CLAMP_SW (var39.i);
-    /* 5: loadw */
-    var36 = ptr0[i];
-    /* 6: addssw */
-    var37.i = ORC_CLAMP_SW (var36.i + var40.i);
-    /* 7: storew */
-    ptr0[i] = var37;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_s16 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union16 *ORC_RESTRICT ptr0;
-  const orc_union16 *ORC_RESTRICT ptr4;
-  orc_union16 var34;
-  orc_union16 var35;
-  orc_union16 var36;
-  orc_union16 var37;
-  orc_union32 var38;
-  orc_union32 var39;
-  orc_union16 var40;
-
-  ptr0 = (orc_union16 *) ex->arrays[0];
-  ptr4 = (orc_union16 *) ex->arrays[4];
-
-  /* 1: loadpw */
-  var35.i = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadw */
-    var34 = ptr4[i];
-    /* 2: mulswl */
-    var38.i = var34.i * var35.i;
-    /* 3: shrsl */
-    var39.i = var38.i >> 11;
-    /* 4: convssslw */
-    var40.i = ORC_CLAMP_SW (var39.i);
-    /* 5: loadw */
-    var36 = ptr0[i];
-    /* 6: addssw */
-    var37.i = ORC_CLAMP_SW (var36.i + var40.i);
-    /* 7: storew */
-    ptr0[i] = var37;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1,
-    const gint16 * ORC_RESTRICT s1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 115, 49, 54,
-        11, 2, 2, 12, 2, 2, 14, 4, 11, 0, 0, 0, 16, 2, 20, 4,
-        20, 2, 176, 32, 4, 24, 125, 32, 32, 16, 165, 33, 32, 71, 0, 0,
-        33, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_s16);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_s16");
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_s16);
-      orc_program_add_destination (p, 2, "d1");
-      orc_program_add_source (p, 2, "s1");
-      orc_program_add_constant (p, 4, 0x0000000b, "c1");
-      orc_program_add_parameter (p, 2, "p1");
-      orc_program_add_temporary (p, 4, "t1");
-      orc_program_add_temporary (p, 2, "t2");
-
-      orc_program_append_2 (p, "mulswl", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsl", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convssslw", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "addssw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_u32 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1,
-    const guint32 * ORC_RESTRICT s1, int p1, int n)
-{
-  int i;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_union32 var35;
-#else
-  orc_union32 var35;
-#endif
-  orc_union32 var36;
-  orc_union32 var37;
-  orc_union32 var38;
-  orc_union32 var39;
-  orc_union64 var40;
-  orc_union64 var41;
-  orc_union32 var42;
-  orc_union32 var43;
-
-  ptr0 = (orc_union32 *) d1;
-  ptr4 = (orc_union32 *) s1;
-
-  /* 1: loadpl */
-  var35.i = (int) 0x80000000;   /* -2147483648 or 1.061e-314f */
-  /* 3: loadpl */
-  var36.i = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var34 = ptr4[i];
-    /* 2: xorl */
-    var39.i = var34.i ^ var35.i;
-    /* 4: mulslq */
-    var40.i = ((orc_int64) var39.i) * ((orc_int64) var36.i);
-    /* 5: shrsq */
-    var41.i = var40.i >> 27;
-    /* 6: convsssql */
-    var42.i = ORC_CLAMP_SL (var41.i);
-    /* 7: xorl */
-    var43.i = var42.i ^ var35.i;
-    /* 8: loadl */
-    var37 = ptr0[i];
-    /* 9: addusl */
-    var38.i =
-        ORC_CLAMP_UL ((orc_int64) (orc_uint32) var37.i +
-        (orc_int64) (orc_uint32) var43.i);
-    /* 10: storel */
-    ptr0[i] = var38;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_u32 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var34;
-#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__)
-  volatile orc_union32 var35;
-#else
-  orc_union32 var35;
-#endif
-  orc_union32 var36;
-  orc_union32 var37;
-  orc_union32 var38;
-  orc_union32 var39;
-  orc_union64 var40;
-  orc_union64 var41;
-  orc_union32 var42;
-  orc_union32 var43;
-
-  ptr0 = (orc_union32 *) ex->arrays[0];
-  ptr4 = (orc_union32 *) ex->arrays[4];
-
-  /* 1: loadpl */
-  var35.i = (int) 0x80000000;   /* -2147483648 or 1.061e-314f */
-  /* 3: loadpl */
-  var36.i = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var34 = ptr4[i];
-    /* 2: xorl */
-    var39.i = var34.i ^ var35.i;
-    /* 4: mulslq */
-    var40.i = ((orc_int64) var39.i) * ((orc_int64) var36.i);
-    /* 5: shrsq */
-    var41.i = var40.i >> 27;
-    /* 6: convsssql */
-    var42.i = ORC_CLAMP_SL (var41.i);
-    /* 7: xorl */
-    var43.i = var42.i ^ var35.i;
-    /* 8: loadl */
-    var37 = ptr0[i];
-    /* 9: addusl */
-    var38.i =
-        ORC_CLAMP_UL ((orc_int64) (orc_uint32) var37.i +
-        (orc_int64) (orc_uint32) var43.i);
-    /* 10: storel */
-    ptr0[i] = var38;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1,
-    const guint32 * ORC_RESTRICT s1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 117, 51, 50,
-        11, 4, 4, 12, 4, 4, 14, 4, 0, 0, 0, 128, 15, 8, 27, 0,
-        0, 0, 0, 0, 0, 0, 16, 4, 20, 8, 20, 4, 132, 33, 4, 16,
-        178, 32, 33, 24, 147, 32, 32, 17, 170, 33, 32, 132, 33, 33, 16, 105,
-        0, 0, 33, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_u32);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_u32");
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_u32);
-      orc_program_add_destination (p, 4, "d1");
-      orc_program_add_source (p, 4, "s1");
-      orc_program_add_constant (p, 4, 0x80000000, "c1");
-      orc_program_add_constant_int64 (p, 8, 0x000000000000001bULL, "c2");
-      orc_program_add_parameter (p, 4, "p1");
-      orc_program_add_temporary (p, 8, "t1");
-      orc_program_add_temporary (p, 4, "t2");
-
-      orc_program_append_2 (p, "xorl", 0, ORC_VAR_T2, ORC_VAR_S1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "mulslq", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsq", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convsssql", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "xorl", 0, ORC_VAR_T2, ORC_VAR_T2, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "addusl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_s32 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1,
-    const gint32 * ORC_RESTRICT s1, int p1, int n)
-{
-  int i;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var34;
-  orc_union32 var35;
-  orc_union32 var36;
-  orc_union32 var37;
-  orc_union64 var38;
-  orc_union64 var39;
-  orc_union32 var40;
-
-  ptr0 = (orc_union32 *) d1;
-  ptr4 = (orc_union32 *) s1;
-
-  /* 1: loadpl */
-  var35.i = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var34 = ptr4[i];
-    /* 2: mulslq */
-    var38.i = ((orc_int64) var34.i) * ((orc_int64) var35.i);
-    /* 3: shrsq */
-    var39.i = var38.i >> 27;
-    /* 4: convsssql */
-    var40.i = ORC_CLAMP_SL (var39.i);
-    /* 5: loadl */
-    var36 = ptr0[i];
-    /* 6: addssl */
-    var37.i = ORC_CLAMP_SL ((orc_int64) var36.i + (orc_int64) var40.i);
-    /* 7: storel */
-    ptr0[i] = var37;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_s32 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var34;
-  orc_union32 var35;
-  orc_union32 var36;
-  orc_union32 var37;
-  orc_union64 var38;
-  orc_union64 var39;
-  orc_union32 var40;
-
-  ptr0 = (orc_union32 *) ex->arrays[0];
-  ptr4 = (orc_union32 *) ex->arrays[4];
-
-  /* 1: loadpl */
-  var35.i = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var34 = ptr4[i];
-    /* 2: mulslq */
-    var38.i = ((orc_int64) var34.i) * ((orc_int64) var35.i);
-    /* 3: shrsq */
-    var39.i = var38.i >> 27;
-    /* 4: convsssql */
-    var40.i = ORC_CLAMP_SL (var39.i);
-    /* 5: loadl */
-    var36 = ptr0[i];
-    /* 6: addssl */
-    var37.i = ORC_CLAMP_SL ((orc_int64) var36.i + (orc_int64) var40.i);
-    /* 7: storel */
-    ptr0[i] = var37;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1,
-    const gint32 * ORC_RESTRICT s1, int p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 115, 51, 50,
-        11, 4, 4, 12, 4, 4, 15, 8, 27, 0, 0, 0, 0, 0, 0, 0,
-        16, 4, 20, 8, 20, 4, 178, 32, 4, 24, 147, 32, 32, 16, 170, 33,
-        32, 104, 0, 0, 33, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_s32);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_s32");
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_s32);
-      orc_program_add_destination (p, 4, "d1");
-      orc_program_add_source (p, 4, "s1");
-      orc_program_add_constant_int64 (p, 8, 0x000000000000001bULL, "c1");
-      orc_program_add_parameter (p, 4, "p1");
-      orc_program_add_temporary (p, 8, "t1");
-      orc_program_add_temporary (p, 4, "t2");
-
-      orc_program_append_2 (p, "mulslq", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "shrsq", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "convsssql", 0, ORC_VAR_T2, ORC_VAR_T1,
-          ORC_VAR_D1, ORC_VAR_D1);
-      orc_program_append_2 (p, "addssl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  ex->params[ORC_VAR_P1] = p1;
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_f32 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_f32 (float *ORC_RESTRICT d1,
-    const float *ORC_RESTRICT s1, float p1, int n)
-{
-  int i;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var33;
-  orc_union32 var34;
-  orc_union32 var35;
-  orc_union32 var36;
-  orc_union32 var37;
-
-  ptr0 = (orc_union32 *) d1;
-  ptr4 = (orc_union32 *) s1;
-
-  /* 1: loadpl */
-  var34.f = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var33 = ptr4[i];
-    /* 2: mulf */
-    {
-      orc_union32 _src1;
-      orc_union32 _src2;
-      orc_union32 _dest1;
-      _src1.i = ORC_DENORMAL (var33.i);
-      _src2.i = ORC_DENORMAL (var34.i);
-      _dest1.f = _src1.f * _src2.f;
-      var37.i = ORC_DENORMAL (_dest1.i);
-    }
-    /* 3: loadl */
-    var35 = ptr0[i];
-    /* 4: addf */
-    {
-      orc_union32 _src1;
-      orc_union32 _src2;
-      orc_union32 _dest1;
-      _src1.i = ORC_DENORMAL (var35.i);
-      _src2.i = ORC_DENORMAL (var37.i);
-      _dest1.f = _src1.f + _src2.f;
-      var36.i = ORC_DENORMAL (_dest1.i);
-    }
-    /* 5: storel */
-    ptr0[i] = var36;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_f32 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union32 *ORC_RESTRICT ptr0;
-  const orc_union32 *ORC_RESTRICT ptr4;
-  orc_union32 var33;
-  orc_union32 var34;
-  orc_union32 var35;
-  orc_union32 var36;
-  orc_union32 var37;
-
-  ptr0 = (orc_union32 *) ex->arrays[0];
-  ptr4 = (orc_union32 *) ex->arrays[4];
-
-  /* 1: loadpl */
-  var34.i = ex->params[24];
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadl */
-    var33 = ptr4[i];
-    /* 2: mulf */
-    {
-      orc_union32 _src1;
-      orc_union32 _src2;
-      orc_union32 _dest1;
-      _src1.i = ORC_DENORMAL (var33.i);
-      _src2.i = ORC_DENORMAL (var34.i);
-      _dest1.f = _src1.f * _src2.f;
-      var37.i = ORC_DENORMAL (_dest1.i);
-    }
-    /* 3: loadl */
-    var35 = ptr0[i];
-    /* 4: addf */
-    {
-      orc_union32 _src1;
-      orc_union32 _src2;
-      orc_union32 _dest1;
-      _src1.i = ORC_DENORMAL (var35.i);
-      _src2.i = ORC_DENORMAL (var37.i);
-      _dest1.f = _src1.f + _src2.f;
-      var36.i = ORC_DENORMAL (_dest1.i);
-    }
-    /* 5: storel */
-    ptr0[i] = var36;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_f32 (float *ORC_RESTRICT d1,
-    const float *ORC_RESTRICT s1, float p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 102, 51, 50,
-        11, 4, 4, 12, 4, 4, 17, 4, 20, 4, 202, 32, 4, 24, 200, 0,
-        0, 32, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_f32);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_f32");
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_f32);
-      orc_program_add_destination (p, 4, "d1");
-      orc_program_add_source (p, 4, "s1");
-      orc_program_add_parameter_float (p, 4, "p1");
-      orc_program_add_temporary (p, 4, "t1");
-
-      orc_program_append_2 (p, "mulf", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "addf", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  {
-    orc_union32 tmp;
-    tmp.f = p1;
-    ex->params[ORC_VAR_P1] = tmp.i;
-  }
-
-  func = c->exec;
-  func (ex);
-}
-#endif
-
-
-/* audiomixer_orc_add_volume_f64 */
-#ifdef DISABLE_ORC
-void
-audiomixer_orc_add_volume_f64 (double *ORC_RESTRICT d1,
-    const double *ORC_RESTRICT s1, double p1, int n)
-{
-  int i;
-  orc_union64 *ORC_RESTRICT ptr0;
-  const orc_union64 *ORC_RESTRICT ptr4;
-  orc_union64 var33;
-  orc_union64 var34;
-  orc_union64 var35;
-  orc_union64 var36;
-  orc_union64 var37;
-
-  ptr0 = (orc_union64 *) d1;
-  ptr4 = (orc_union64 *) s1;
-
-  /* 1: loadpq */
-  var34.f = p1;
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadq */
-    var33 = ptr4[i];
-    /* 2: muld */
-    {
-      orc_union64 _src1;
-      orc_union64 _src2;
-      orc_union64 _dest1;
-      _src1.i = ORC_DENORMAL_DOUBLE (var33.i);
-      _src2.i = ORC_DENORMAL_DOUBLE (var34.i);
-      _dest1.f = _src1.f * _src2.f;
-      var37.i = ORC_DENORMAL_DOUBLE (_dest1.i);
-    }
-    /* 3: loadq */
-    var35 = ptr0[i];
-    /* 4: addd */
-    {
-      orc_union64 _src1;
-      orc_union64 _src2;
-      orc_union64 _dest1;
-      _src1.i = ORC_DENORMAL_DOUBLE (var35.i);
-      _src2.i = ORC_DENORMAL_DOUBLE (var37.i);
-      _dest1.f = _src1.f + _src2.f;
-      var36.i = ORC_DENORMAL_DOUBLE (_dest1.i);
-    }
-    /* 5: storeq */
-    ptr0[i] = var36;
-  }
-
-}
-
-#else
-static void
-_backup_audiomixer_orc_add_volume_f64 (OrcExecutor * ORC_RESTRICT ex)
-{
-  int i;
-  int n = ex->n;
-  orc_union64 *ORC_RESTRICT ptr0;
-  const orc_union64 *ORC_RESTRICT ptr4;
-  orc_union64 var33;
-  orc_union64 var34;
-  orc_union64 var35;
-  orc_union64 var36;
-  orc_union64 var37;
-
-  ptr0 = (orc_union64 *) ex->arrays[0];
-  ptr4 = (orc_union64 *) ex->arrays[4];
-
-  /* 1: loadpq */
-  var34.i =
-      (ex->params[24] & 0xffffffff) | ((orc_uint64) (ex->params[24 +
-              (ORC_VAR_T1 - ORC_VAR_P1)]) << 32);
-
-  for (i = 0; i < n; i++) {
-    /* 0: loadq */
-    var33 = ptr4[i];
-    /* 2: muld */
-    {
-      orc_union64 _src1;
-      orc_union64 _src2;
-      orc_union64 _dest1;
-      _src1.i = ORC_DENORMAL_DOUBLE (var33.i);
-      _src2.i = ORC_DENORMAL_DOUBLE (var34.i);
-      _dest1.f = _src1.f * _src2.f;
-      var37.i = ORC_DENORMAL_DOUBLE (_dest1.i);
-    }
-    /* 3: loadq */
-    var35 = ptr0[i];
-    /* 4: addd */
-    {
-      orc_union64 _src1;
-      orc_union64 _src2;
-      orc_union64 _dest1;
-      _src1.i = ORC_DENORMAL_DOUBLE (var35.i);
-      _src2.i = ORC_DENORMAL_DOUBLE (var37.i);
-      _dest1.f = _src1.f + _src2.f;
-      var36.i = ORC_DENORMAL_DOUBLE (_dest1.i);
-    }
-    /* 5: storeq */
-    ptr0[i] = var36;
-  }
-
-}
-
-void
-audiomixer_orc_add_volume_f64 (double *ORC_RESTRICT d1,
-    const double *ORC_RESTRICT s1, double p1, int n)
-{
-  OrcExecutor _ex, *ex = &_ex;
-  static volatile int p_inited = 0;
-  static OrcCode *c = 0;
-  void (*func) (OrcExecutor *);
-
-  if (!p_inited) {
-    orc_once_mutex_lock ();
-    if (!p_inited) {
-      OrcProgram *p;
-
-#if 1
-      static const orc_uint8 bc[] = {
-        1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114,
-        99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 102, 54, 52,
-        11, 8, 8, 12, 8, 8, 18, 8, 20, 8, 214, 32, 4, 24, 212, 0,
-        0, 32, 2, 0,
-      };
-      p = orc_program_new_from_static_bytecode (bc);
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_f64);
-#else
-      p = orc_program_new ();
-      orc_program_set_name (p, "audiomixer_orc_add_volume_f64");
-      orc_program_set_backup_function (p,
-          _backup_audiomixer_orc_add_volume_f64);
-      orc_program_add_destination (p, 8, "d1");
-      orc_program_add_source (p, 8, "s1");
-      orc_program_add_parameter_double (p, 8, "p1");
-      orc_program_add_temporary (p, 8, "t1");
-
-      orc_program_append_2 (p, "muld", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1,
-          ORC_VAR_D1);
-      orc_program_append_2 (p, "addd", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T1,
-          ORC_VAR_D1);
-#endif
-
-      orc_program_compile (p);
-      c = orc_program_take_code (p);
-      orc_program_free (p);
-    }
-    p_inited = TRUE;
-    orc_once_mutex_unlock ();
-  }
-  ex->arrays[ORC_VAR_A2] = c;
-  ex->program = 0;
-
-  ex->n = n;
-  ex->arrays[ORC_VAR_D1] = d1;
-  ex->arrays[ORC_VAR_S1] = (void *) s1;
-  {
-    orc_union64 tmp;
-    tmp.f = p1;
-    ex->params[ORC_VAR_P1] = ((orc_uint64) tmp.i) & 0xffffffff;
-    ex->params[ORC_VAR_T1] = ((orc_uint64) tmp.i) >> 32;
-  }
-
-  func = c->exec;
-  func (ex);
-}
-#endif
diff --git a/gst/audiomixer/gstaudiomixerorc-dist.h b/gst/audiomixer/gstaudiomixerorc-dist.h
deleted file mode 100644
index af0de0139..000000000
--- a/gst/audiomixer/gstaudiomixerorc-dist.h
+++ /dev/null
@@ -1,106 +0,0 @@
-
-/* autogenerated from gstaudiomixerorc.orc */
-
-#ifndef _GSTAUDIOMIXERORC_H_
-#define _GSTAUDIOMIXERORC_H_
-
-#include <glib.h>
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-
-#ifndef _ORC_INTEGER_TYPEDEFS_
-#define _ORC_INTEGER_TYPEDEFS_
-#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
-#include <stdint.h>
-typedef int8_t orc_int8;
-typedef int16_t orc_int16;
-typedef int32_t orc_int32;
-typedef int64_t orc_int64;
-typedef uint8_t orc_uint8;
-typedef uint16_t orc_uint16;
-typedef uint32_t orc_uint32;
-typedef uint64_t orc_uint64;
-#define ORC_UINT64_C(x) UINT64_C(x)
-#elif defined(_MSC_VER)
-typedef signed __int8 orc_int8;
-typedef signed __int16 orc_int16;
-typedef signed __int32 orc_int32;
-typedef signed __int64 orc_int64;
-typedef unsigned __int8 orc_uint8;
-typedef unsigned __int16 orc_uint16;
-typedef unsigned __int32 orc_uint32;
-typedef unsigned __int64 orc_uint64;
-#define ORC_UINT64_C(x) (x##Ui64)
-#define inline __inline
-#else
-#include <limits.h>
-typedef signed char orc_int8;
-typedef short orc_int16;
-typedef int orc_int32;
-typedef unsigned char orc_uint8;
-typedef unsigned short orc_uint16;
-typedef unsigned int orc_uint32;
-#if INT_MAX == LONG_MAX
-typedef long long orc_int64;
-typedef unsigned long long orc_uint64;
-#define ORC_UINT64_C(x) (x##ULL)
-#else
-typedef long orc_int64;
-typedef unsigned long orc_uint64;
-#define ORC_UINT64_C(x) (x##UL)
-#endif
-#endif
-typedef union { orc_int16 i; orc_int8 x2[2]; } orc_union16;
-typedef union { orc_int32 i; float f; orc_int16 x2[2]; orc_int8 x4[4]; } orc_union32;
-typedef union { orc_int64 i; double f; orc_int32 x2[2]; float x2f[2]; orc_int16 x4[4]; } orc_union64;
-#endif
-#ifndef ORC_RESTRICT
-#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
-#define ORC_RESTRICT restrict
-#elif defined(__GNUC__) && __GNUC__ >= 4
-#define ORC_RESTRICT __restrict__
-#else
-#define ORC_RESTRICT
-#endif
-#endif
-
-#ifndef ORC_INTERNAL
-#if defined(__SUNPRO_C) && (__SUNPRO_C >= 0x590)
-#define ORC_INTERNAL __attribute__((visibility("hidden")))
-#elif defined(__SUNPRO_C) && (__SUNPRO_C >= 0x550)
-#define ORC_INTERNAL __hidden
-#elif defined (__GNUC__)
-#define ORC_INTERNAL __attribute__((visibility("hidden")))
-#else
-#define ORC_INTERNAL
-#endif
-#endif
-
-void audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1, const gint32 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1, const gint16 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1, const guint32 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1, const guint16 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_f32 (float * ORC_RESTRICT d1, const float * ORC_RESTRICT s1, int n);
-void audiomixer_orc_add_f64 (double * ORC_RESTRICT d1, const double * ORC_RESTRICT s1, int n);
-void audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n);
-void audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1, const guint16 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1, const gint16 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1, const guint32 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1, const gint32 * ORC_RESTRICT s1, int p1, int n);
-void audiomixer_orc_add_volume_f32 (float * ORC_RESTRICT d1, const float * ORC_RESTRICT s1, float p1, int n);
-void audiomixer_orc_add_volume_f64 (double * ORC_RESTRICT d1, const double * ORC_RESTRICT s1, double p1, int n);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
-
diff --git a/gst/audiomixer/gstaudiomixerorc.orc b/gst/audiomixer/gstaudiomixerorc.orc
deleted file mode 100644
index 5eaff2395..000000000
--- a/gst/audiomixer/gstaudiomixerorc.orc
+++ /dev/null
@@ -1,176 +0,0 @@
-.function audiomixer_orc_add_s32
-.dest 4 d1 gint32
-.source 4 s1 gint32
-
-addssl d1, d1, s1
-
-
-.function audiomixer_orc_add_s16
-.dest 2 d1 gint16
-.source 2 s1 gint16
-
-addssw d1, d1, s1
-
-
-.function audiomixer_orc_add_s8
-.dest 1 d1 gint8
-.source 1 s1 gint8
-
-addssb d1, d1, s1
-
-
-.function audiomixer_orc_add_u32
-.dest 4 d1 guint32
-.source 4 s1 guint32
-
-addusl d1, d1, s1
-
-
-.function audiomixer_orc_add_u16
-.dest 2 d1 guint16
-.source 2 s1 guint16
-
-addusw d1, d1, s1
-
-
-.function audiomixer_orc_add_u8
-.dest 1 d1 guint8
-.source 1 s1 guint8
-
-addusb d1, d1, s1
-
-
-.function audiomixer_orc_add_f32
-.dest 4 d1 float
-.source 4 s1 float
-
-addf d1, d1, s1
-
-.function audiomixer_orc_add_f64
-.dest 8 d1 double
-.source 8 s1 double
-
-addd d1, d1, s1
-
-
-.function audiomixer_orc_volume_u8
-.dest 1 d1 guint8
-.param 1 p1
-.const 1 c1 0x80
-.temp 2 t1
-.temp 1 t2
-
-xorb t2, d1, c1
-mulsbw t1, t2, p1
-shrsw t1, t1, 3
-convssswb t2, t1
-xorb d1, t2, c1
-
-
-.function audiomixer_orc_add_volume_u8
-.dest 1 d1 guint8
-.source 1 s1 guint8
-.param 1 p1
-.const 1 c1 0x80
-.temp 2 t1
-.temp 1 t2
-
-xorb t2, s1, c1
-mulsbw t1, t2, p1
-shrsw t1, t1, 3
-convssswb t2, t1
-xorb t2, t2, c1
-addusb d1, d1, t2
-
-
-.function audiomixer_orc_add_volume_s8
-.dest 1 d1 gint8
-.source 1 s1 gint8
-.param 1 p1
-.temp 2 t1
-.temp 1 t2
-
-mulsbw t1, s1, p1
-shrsw t1, t1, 3
-convssswb t2, t1
-addssb d1, d1, t2
-
-
-.function audiomixer_orc_add_volume_u16
-.dest 2 d1 guint16
-.source 2 s1 guint16
-.param 2 p1
-.const 2 c1 0x8000
-.temp 4 t1
-.temp 2 t2
-
-xorw t2, s1, c1
-mulswl t1, t2, p1
-shrsl t1, t1, 11
-convssslw t2, t1
-xorw t2, t2, c1
-addusw d1, d1, t2
-
-
-.function audiomixer_orc_add_volume_s16
-.dest 2 d1 gint16
-.source 2 s1 gint16
-.param 2 p1
-.temp 4 t1
-.temp 2 t2
-
-mulswl t1, s1, p1
-shrsl t1, t1, 11
-convssslw t2, t1
-addssw d1, d1, t2
-
-
-.function audiomixer_orc_add_volume_u32
-.dest 4 d1 guint32
-.source 4 s1 guint32
-.param 4 p1
-.const 4 c1 0x80000000
-.temp 8 t1
-.temp 4 t2
-
-xorl t2, s1, c1
-mulslq t1, t2, p1
-shrsq t1, t1, 27
-convsssql t2, t1
-xorl t2, t2, c1
-addusl d1, d1, t2
-
-
-.function audiomixer_orc_add_volume_s32
-.dest 4 d1 gint32
-.source 4 s1 gint32
-.param 4 p1
-.temp 8 t1
-.temp 4 t2
-
-mulslq t1, s1, p1
-shrsq t1, t1, 27
-convsssql t2, t1
-addssl d1, d1, t2
-
-
-.function audiomixer_orc_add_volume_f32
-.dest 4 d1 float
-.source 4 s1 float
-.floatparam 4 p1
-.temp 4 t1
-
-mulf t1, s1, p1
-addf d1, d1, t1
-
-
-.function audiomixer_orc_add_volume_f64
-.dest 8 d1 double
-.source 8 s1 double
-.doubleparam 8 p1
-.temp 8 t1
-
-muld t1, s1, p1
-addd d1, d1, t1
-
-
diff --git a/gst/audiomixer/meson.build b/gst/audiomixer/meson.build
deleted file mode 100644
index ccfe1b9d3..000000000
--- a/gst/audiomixer/meson.build
+++ /dev/null
@@ -1,32 +0,0 @@
-audiomixer_sources = [
-  'gstaudiomixer.c',
-  'gstaudiointerleave.c',
-]
-
-orcsrc = 'gstaudiomixerorc'
-if have_orcc
-  orc_h = custom_target(orcsrc + '.h',
-    input : orcsrc + '.orc',
-    output : orcsrc + '.h',
-    command : orcc_args + ['--header', '-o', '@OUTPUT@', '@INPUT@'])
-  orc_c = custom_target(orcsrc + '.c',
-    input : orcsrc + '.orc',
-    output : orcsrc + '.c',
-    command : orcc_args + ['--implementation', '-o', '@OUTPUT@', '@INPUT@'])
-else
-  orc_h = configure_file(input : orcsrc + '-dist.h',
-    output : orcsrc + '.h',
-    configuration : configuration_data())
-  orc_c = configure_file(input : orcsrc + '-dist.c',
-    output : orcsrc + '.c',
-    configuration : configuration_data())
-endif
-
-gstaudiomixer = library('gstaudiomixer',
-  audiomixer_sources, orc_c, orc_h,
-  c_args : gst_plugins_bad_args + [ '-DGST_USE_UNSTABLE_API' ],
-  include_directories : [configinc],
-  dependencies : [gstbadaudio_dep, gstaudio_dep, gstbase_dep, orc_dep],
-  install : true,
-  install_dir : plugins_install_dir,
-)
diff --git a/gst/meson.build b/gst/meson.build
index b56b68d26..3eff27e55 100644
--- a/gst/meson.build
+++ b/gst/meson.build
@@ -5,7 +5,6 @@ subdir('aiff')
 subdir('asfmux')
 subdir('audiobuffersplit')
 subdir('audiofxbad')
-subdir('audiomixer')
 subdir('audiomixmatrix')
 subdir('audiovisualizers')
 subdir('autoconvert')
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 0343da7eb..5a7c497b2 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -127,7 +127,7 @@ check_kate=
 endif
 
 if HAVE_ORC
-check_orc = orc/bayer orc/audiomixer orc/compositor
+check_orc = orc/bayer orc/compositor
 else
 check_orc =
 endif
@@ -257,8 +257,6 @@ check_PROGRAMS = \
 	elements/videoframe-audiolevel \
 	elements/autoconvert \
 	elements/autovideoconvert \
-	elements/audiointerleave \
-	elements/audiomixer \
 	elements/asfmux \
 	elements/camerabin \
 	elements/gdppay \
@@ -313,12 +311,6 @@ LDADD = $(GST_CHECK_LIBS)
 generic_states_CFLAGS = $(AM_CFLAGS) $(GLIB_CFLAGS)
 generic_states_LDADD = $(LDADD) $(GLIB_LIBS)
 
-elements_audiomixer_LDADD = $(GST_BASE_LIBS) $(GST_CONTROLLER_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
-elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CONTROLLER_CFLAGS) $(AM_CFLAGS)
-
-elements_audiointerleave_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ $(GST_AUDIO_LIBS) $(LDADD)
-elements_audiointerleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
-
 elements_pnm_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
 	$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
@@ -542,14 +534,6 @@ orc/bayer.c: $(top_srcdir)/gst/bayer/gstbayerorc.orc
 	$(MKDIR_P) orc
 	$(ORCC) --test -o $@ $<
 
-orc_audiomixer_CFLAGS = $(ORC_CFLAGS)
-orc_audiomixer_LDADD = $(ORC_LIBS) -lorc-test-0.4
-nodist_orc_audiomixer_SOURCES = orc/audiomixer.c
-
-orc/audiomixer.c: $(top_srcdir)/gst/audiomixer/gstaudiomixerorc.orc
-	$(MKDIR_P) orc
-	$(ORCC) --test -o $@ $<
-
 elements_compositor_LDADD = \
 	$(GST_PLUGINS_BASE_LIBS) $(GST_VIDEO_LIBS) $(GST_BASE_LIBS) $(LDADD)
 elements_compositor_CFLAGS = \
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index 741772b5a..d264dae57 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -2,8 +2,6 @@
 aiffparse
 asfmux
 assrender
-audiointerleave
-audiomixer
 autoconvert
 autovideoconvert
 baseaudiovisualizer
diff --git a/tests/check/elements/audiointerleave.c b/tests/check/elements/audiointerleave.c
deleted file mode 100644
index 71348f459..000000000
--- a/tests/check/elements/audiointerleave.c
+++ /dev/null
@@ -1,1128 +0,0 @@
-/* GStreamer unit tests for the audiointerleave element
- * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
- * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * 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.
- */
-
-/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
- * with newer GLib versions (>= 2.31.0) */
-#define GLIB_DISABLE_DEPRECATION_WARNINGS
-
-#ifdef HAVE_CONFIG_H
-# include "config.h"
-#endif
-
-#ifdef HAVE_VALGRIND
-# include <valgrind/valgrind.h>
-#endif
-
-#include <gst/check/gstcheck.h>
-#include <gst/audio/audio.h>
-#include <gst/audio/audio-enumtypes.h>
-
-#include <gst/check/gstharness.h>
-
-static void
-gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element,
-    GstCaps * caps, GstFormat format, const gchar * stream_id)
-{
-  GstSegment segment;
-
-  gst_segment_init (&segment, format);
-
-  fail_unless (gst_pad_push_event (srcpad,
-          gst_event_new_stream_start (stream_id)));
-  if (caps)
-    fail_unless (gst_pad_push_event (srcpad, gst_event_new_caps (caps)));
-  fail_unless (gst_pad_push_event (srcpad, gst_event_new_segment (&segment)));
-}
-
-GST_START_TEST (test_create_and_unref)
-{
-  GstElement *interleave;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_object_unref (interleave);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_request_pads)
-{
-  GstElement *interleave;
-  GstPad *pad1, *pad2;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  pad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (pad1 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink_0");
-
-  pad2 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (pad2 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink_1");
-
-  gst_element_release_request_pad (interleave, pad2);
-  gst_object_unref (pad2);
-  gst_element_release_request_pad (interleave, pad1);
-  gst_object_unref (pad1);
-
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_object_unref (interleave);
-}
-
-GST_END_TEST;
-
-static GstPad **mysrcpads, *mysinkpad;
-static GstBus *bus;
-static GstElement *interleave;
-static GMutex data_mutex;
-static GCond data_cond;
-static gint have_data;
-static gfloat input[2];
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " GST_AUDIO_NE (F32) ", "
-        "channels = (int) 2, layout = (string) {interleaved, non-interleaved}, rate = (int) 48000"));
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " GST_AUDIO_NE (F32) ", "
-        "channels = (int) 1, layout = (string) interleaved, rate = (int) 48000"));
-
-#define CAPS_48khz \
-        "audio/x-raw, " \
-        "format = (string) " GST_AUDIO_NE (F32) ", " \
-        "channels = (int) 1, layout = (string) non-interleaved," \
-        "rate = (int) 48000"
-
-static GstFlowReturn
-interleave_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
-{
-  GstMapInfo map;
-  gfloat *outdata;
-  gint i;
-
-  fail_unless (GST_IS_BUFFER (buffer));
-  fail_unless (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP));
-  gst_buffer_map (buffer, &map, GST_MAP_READ);
-  outdata = (gfloat *) map.data;
-  fail_unless (outdata != NULL);
-
-#ifdef HAVE_VALGRIND
-  if (!(RUNNING_ON_VALGRIND))
-#endif
-    for (i = 0; i < map.size / sizeof (float); i += 2) {
-      fail_unless_equals_float (outdata[i], input[0]);
-      fail_unless_equals_float (outdata[i + 1], input[1]);
-    }
-
-  g_mutex_lock (&data_mutex);
-  have_data += map.size;
-  g_cond_signal (&data_cond);
-  g_mutex_unlock (&data_mutex);
-
-  gst_buffer_unmap (buffer, &map);
-  gst_buffer_unref (buffer);
-
-
-  return GST_FLOW_OK;
-}
-
-GST_START_TEST (test_audiointerleave_2ch)
-{
-  GstElement *queue;
-  GstPad *sink0, *sink1, *src, *tmp;
-  GstCaps *caps;
-  gint i;
-  GstBuffer *inbuf;
-  gfloat *indata;
-  GstMapInfo map;
-
-  mysrcpads = g_new0 (GstPad *, 2);
-
-  have_data = 0;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-
-  sink0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink0 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
-
-  sink1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink1 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
-
-  mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
-  fail_unless (mysrcpads[0] != NULL);
-
-  caps = gst_caps_from_string (CAPS_48khz);
-  gst_pad_set_active (mysrcpads[0], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
-      GST_FORMAT_TIME, "0");
-  gst_pad_use_fixed_caps (mysrcpads[0]);
-
-  mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
-  fail_unless (mysrcpads[1] != NULL);
-
-  gst_pad_set_active (mysrcpads[1], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
-      GST_FORMAT_TIME, "1");
-  gst_pad_use_fixed_caps (mysrcpads[1]);
-
-  tmp = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
-
-  mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
-  fail_unless (mysinkpad != NULL);
-  gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
-  gst_pad_set_active (mysinkpad, TRUE);
-
-  src = gst_element_get_static_pad (interleave, "src");
-  fail_unless (src != NULL);
-  fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
-  gst_object_unref (src);
-
-  bus = gst_bus_new ();
-  gst_element_set_bus (interleave, bus);
-
-  fail_unless (gst_element_set_state (interleave,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-  fail_unless (gst_element_set_state (queue,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
-  input[0] = -1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = -1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
-  input[1] = 1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = GST_SECOND;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = -1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = GST_SECOND;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  g_mutex_lock (&data_mutex);
-  while (have_data < 48000 * 2 * 2 * sizeof (float))
-    g_cond_wait (&data_cond, &data_mutex);
-  g_mutex_unlock (&data_mutex);
-
-  gst_bus_set_flushing (bus, TRUE);
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_element_set_state (queue, GST_STATE_NULL);
-
-  gst_object_unref (mysrcpads[0]);
-  gst_object_unref (mysrcpads[1]);
-  gst_object_unref (mysinkpad);
-
-  gst_element_release_request_pad (interleave, sink0);
-  gst_object_unref (sink0);
-  gst_element_release_request_pad (interleave, sink1);
-  gst_object_unref (sink1);
-
-  gst_object_unref (interleave);
-  gst_object_unref (queue);
-  gst_object_unref (bus);
-  gst_caps_unref (caps);
-
-  g_free (mysrcpads);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_1eos)
-{
-  GstElement *queue;
-  GstPad *sink0, *sink1, *src, *tmp;
-  GstCaps *caps;
-  gint i;
-  GstBuffer *inbuf;
-  gfloat *indata;
-  GstMapInfo map;
-
-  mysrcpads = g_new0 (GstPad *, 2);
-
-  have_data = 0;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-
-  sink0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink0 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
-
-  sink1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink1 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
-
-  mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
-  fail_unless (mysrcpads[0] != NULL);
-
-  caps = gst_caps_from_string (CAPS_48khz);
-  gst_pad_set_active (mysrcpads[0], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
-      GST_FORMAT_TIME, "0");
-  gst_pad_use_fixed_caps (mysrcpads[0]);
-
-  mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
-  fail_unless (mysrcpads[1] != NULL);
-
-  gst_pad_set_active (mysrcpads[1], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
-      GST_FORMAT_TIME, "1");
-  gst_pad_use_fixed_caps (mysrcpads[1]);
-
-  tmp = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
-
-  mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
-  fail_unless (mysinkpad != NULL);
-  gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
-  gst_pad_set_active (mysinkpad, TRUE);
-
-  src = gst_element_get_static_pad (interleave, "src");
-  fail_unless (src != NULL);
-  fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
-  gst_object_unref (src);
-
-  bus = gst_bus_new ();
-  gst_element_set_bus (interleave, bus);
-
-  fail_unless (gst_element_set_state (interleave,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-  fail_unless (gst_element_set_state (queue,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
-  input[0] = -1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = -1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
-  input[1] = 1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  g_mutex_lock (&data_mutex);
-  /* 48000 samples per buffer * 2 sources * 2 buffers */
-  while (have_data != 48000 * 2 * sizeof (float))
-    g_cond_wait (&data_cond, &data_mutex);
-  g_mutex_unlock (&data_mutex);
-
-  input[0] = 0.0;
-  gst_pad_push_event (mysrcpads[0], gst_event_new_eos ());
-
-  input[1] = 1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  GST_BUFFER_PTS (inbuf) = GST_SECOND;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  g_mutex_lock (&data_mutex);
-  /* 48000 samples per buffer * 2 sources * 2 buffers */
-  while (have_data != 48000 * 2 * 2 * sizeof (float))
-    g_cond_wait (&data_cond, &data_mutex);
-  g_mutex_unlock (&data_mutex);
-
-  gst_bus_set_flushing (bus, TRUE);
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_element_set_state (queue, GST_STATE_NULL);
-
-  gst_object_unref (mysrcpads[0]);
-  gst_object_unref (mysrcpads[1]);
-  gst_object_unref (mysinkpad);
-
-  gst_element_release_request_pad (interleave, sink0);
-  gst_object_unref (sink0);
-  gst_element_release_request_pad (interleave, sink1);
-  gst_object_unref (sink1);
-
-  gst_object_unref (interleave);
-  gst_object_unref (queue);
-  gst_object_unref (bus);
-  gst_caps_unref (caps);
-
-  g_free (mysrcpads);
-}
-
-GST_END_TEST;
-
-static void
-src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
-    gboolean interleaved, gpointer user_data)
-{
-  gint n = GPOINTER_TO_INT (user_data);
-  gfloat *data;
-  gint i, num_samples;
-  GstCaps *caps;
-  guint64 mask;
-  GstAudioChannelPosition pos;
-  GstMapInfo map;
-
-  fail_unless (gst_buffer_is_writable (buffer));
-
-  switch (n) {
-    case 0:
-    case 1:
-    case 2:
-      pos = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
-      break;
-    case 3:
-      pos = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
-      break;
-    default:
-      pos = GST_AUDIO_CHANNEL_POSITION_INVALID;
-      break;
-  }
-
-  mask = G_GUINT64_CONSTANT (1) << pos;
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-      "channels", G_TYPE_INT, 1,
-      "layout", G_TYPE_STRING, interleaved ? "interleaved" : "non-interleaved",
-      "channel-mask", GST_TYPE_BITMASK, mask, "rate", G_TYPE_INT, 48000, NULL);
-
-  gst_pad_set_caps (pad, caps);
-  gst_caps_unref (caps);
-
-  fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE));
-  fail_unless (map.size % sizeof (gfloat) == 0);
-
-  fail_unless (map.size > 480);
-
-  num_samples = map.size / sizeof (gfloat);
-  data = (gfloat *) map.data;
-
-  for (i = 0; i < num_samples; i++)
-    data[i] = (n % 2 == 0) ? -1.0 : 1.0;
-
-  gst_buffer_unmap (buffer, &map);
-}
-
-static void
-src_handoff_float32_audiointerleaved (GstElement * element, GstBuffer * buffer,
-    GstPad * pad, gpointer user_data)
-{
-  src_handoff_float32 (element, buffer, pad, TRUE, user_data);
-}
-
-static void
-src_handoff_float32_non_audiointerleaved (GstElement * element,
-    GstBuffer * buffer, GstPad * pad, gpointer user_data)
-{
-  src_handoff_float32 (element, buffer, pad, FALSE, user_data);
-}
-
-static void
-sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
-    gpointer user_data)
-{
-  gint i;
-  GstMapInfo map;
-  gfloat *data;
-  GstCaps *caps, *ccaps;
-  gint n = GPOINTER_TO_INT (user_data);
-  guint64 mask;
-
-  fail_unless (GST_IS_BUFFER (buffer));
-  gst_buffer_map (buffer, &map, GST_MAP_READ);
-  data = (gfloat *) map.data;
-
-  /* Give a little leeway for rounding errors */
-  fail_unless (gst_util_uint64_scale (map.size, GST_SECOND,
-          48000 * 2 * sizeof (gfloat)) <= GST_BUFFER_DURATION (buffer) + 1 ||
-      gst_util_uint64_scale (map.size, GST_SECOND,
-          48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1);
-
-  if (n == 0 || n == 3) {
-    GstAudioChannelPosition pos[2] =
-        { GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
-    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
-  } else if (n == 1) {
-    GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
-    };
-    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
-  } else if (n == 2) {
-    GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-      GST_AUDIO_CHANNEL_POSITION_REAR_CENTER
-    };
-    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
-  } else {
-    g_assert_not_reached ();
-  }
-
-  if (pad) {
-    caps = gst_caps_new_simple ("audio/x-raw",
-        "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-        "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
-        "layout", G_TYPE_STRING, "interleaved",
-        "channel-mask", GST_TYPE_BITMASK, mask, NULL);
-
-    ccaps = gst_pad_get_current_caps (pad);
-    fail_unless (gst_caps_is_equal (caps, ccaps));
-    gst_caps_unref (ccaps);
-    gst_caps_unref (caps);
-  }
-#ifdef HAVE_VALGRIND
-  if (!(RUNNING_ON_VALGRIND))
-#endif
-    for (i = 0; i < map.size / sizeof (float); i += 2) {
-      fail_unless_equals_float (data[i], -1.0);
-      if (n != 3)
-        fail_unless_equals_float (data[i + 1], 1.0);
-    }
-  have_data += map.size;
-
-  gst_buffer_unmap (buffer, &map);
-
-}
-
-static void
-test_audiointerleave_2ch_pipeline (gboolean interleaved)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-  void *src_handoff_float32 =
-      interleaved ? &src_handoff_float32_audiointerleaved :
-      &src_handoff_float32_non_audiointerleaved;
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
-      GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
-      GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_audiointerleaved)
-{
-  test_audiointerleave_2ch_pipeline (TRUE);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_non_audiointerleaved)
-{
-  test_audiointerleave_2ch_pipeline (FALSE);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (3));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  g_object_set (interleave, "channel-positions-from-input", TRUE, NULL);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-  GValueArray *arr;
-  GValue val = { 0, };
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
-  arr = g_value_array_new (2);
-
-  g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
-  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER);
-  g_value_array_append (arr, &val);
-  g_value_reset (&val);
-  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER);
-  g_value_array_append (arr, &val);
-  g_value_unset (&val);
-
-  g_object_set (interleave, "channel-positions", arr, NULL);
-  g_value_array_free (arr);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (2));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type)
-{
-  GstEvent *e;
-
-  e = gst_harness_pull_event (hsrc);
-  fail_unless (GST_EVENT_TYPE (e) == type);
-  gst_harness_push_event (h, e);
-}
-
-GST_START_TEST (test_audiointerleave_2ch_smallbuf)
-{
-  GstElement *audiointerleave;
-  GstHarness *hsrc;
-  GstHarness *h;
-  GstHarness *h2;
-  GstBuffer *buffer;
-  gint i;
-  GstEvent *ev;
-  GstCaps *ecaps, *caps;
-
-  audiointerleave = gst_element_factory_make ("audiointerleave", NULL);
-
-  g_object_set (audiointerleave, "latency", GST_SECOND / 2,
-      "output-buffer-duration", GST_SECOND / 4, NULL);
-
-  h = gst_harness_new_with_element (audiointerleave, "sink_0", "src");
-  gst_harness_use_testclock (h);
-
-  h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL);
-  gst_harness_set_src_caps_str (h2, "audio/x-raw, "
-      "format=" GST_AUDIO_NE (F32) ", channels=(int)1,"
-      " layout=interleaved, rate=48000, channel-mask=(bitmask)8");
-
-  hsrc = gst_harness_new ("fakesrc");
-  gst_harness_use_testclock (hsrc);
-  g_object_set (hsrc->element,
-      "is-live", TRUE,
-      "sync", TRUE,
-      "signal-handoffs", TRUE,
-      "format", GST_FORMAT_TIME,
-      "sizetype", 2,
-      "sizemax", (int) 480 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_signal_connect (hsrc->element, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
-  gst_harness_play (hsrc);
-
-  gst_harness_crank_single_clock_wait (hsrc);
-  forward_check_event (h, hsrc, GST_EVENT_STREAM_START);
-  forward_check_event (h, hsrc, GST_EVENT_CAPS);
-  forward_check_event (h, hsrc, GST_EVENT_SEGMENT);
-  gst_harness_push (h, gst_harness_pull (hsrc));        /* buffer */
-
-  for (i = 0; i < 24; i++) {
-    gst_harness_crank_single_clock_wait (hsrc);
-    forward_check_event (h, hsrc, GST_EVENT_CAPS);
-    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
-  }
-
-  gst_harness_crank_single_clock_wait (h);
-
-
-  gst_event_unref (gst_harness_pull_event (h)); /* stream-start */
-  ev = gst_harness_pull_event (h);      /* caps */
-  fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-      "channels", G_TYPE_INT, 2,
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK,
-      (guint64) 0x9, NULL);
-
-  gst_event_parse_caps (ev, &ecaps);
-  gst_check_caps_equal (ecaps, caps);
-  gst_caps_unref (caps);
-  gst_event_unref (ev);
-
-  /* eat the caps processing */
-  gst_harness_crank_single_clock_wait (h);
-  for (i = 0; i < 23; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 750 * GST_MSECOND);
-
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-  fail_unless_equals_int (gst_harness_buffers_received (h), 1);
-
-  for (i = 0; i < 50; i++) {
-    gst_harness_crank_single_clock_wait (hsrc);
-    forward_check_event (h, hsrc, GST_EVENT_CAPS);
-    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
-  }
-  for (i = 0; i < 25; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 1000 * GST_MSECOND);
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-  fail_unless_equals_int (gst_harness_buffers_received (h), 2);
-
-  for (i = 0; i < 25; i++) {
-    gst_harness_crank_single_clock_wait (hsrc);
-    forward_check_event (h, hsrc, GST_EVENT_CAPS);
-    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
-  }
-  for (i = 0; i < 25; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 1250 * GST_MSECOND);
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-  fail_unless_equals_int (gst_harness_buffers_received (h), 3);
-
-  gst_harness_push_event (h, gst_event_new_eos ());
-
-  for (i = 0; i < 25; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 1500 * GST_MSECOND);
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-
-  fail_unless_equals_int (gst_harness_buffers_received (h), 4);
-
-  gst_harness_teardown (h2);
-  gst_harness_teardown (h);
-  gst_harness_teardown (hsrc);
-  gst_object_unref (audiointerleave);
-}
-
-GST_END_TEST;
-
-static Suite *
-audiointerleave_suite (void)
-{
-  Suite *s = suite_create ("audiointerleave");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_set_timeout (tc_chain, 180);
-  tcase_add_test (tc_chain, test_create_and_unref);
-  tcase_add_test (tc_chain, test_request_pads);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_1eos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_audiointerleaved);
-  tcase_add_test (tc_chain,
-      test_audiointerleave_2ch_pipeline_non_audiointerleaved);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf);
-
-  return s;
-}
-
-GST_CHECK_MAIN (audiointerleave);
diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c
deleted file mode 100644
index 4a8a8233b..000000000
--- a/tests/check/elements/audiomixer.c
+++ /dev/null
@@ -1,1894 +0,0 @@
-/* GStreamer
- *
- * unit test for audiomixer
- *
- * Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
- * Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com>
- *
- * 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.
- */
-
-#ifdef HAVE_CONFIG_H
-# include <config.h>
-#endif
-
-#ifdef HAVE_VALGRIND
-# include <valgrind/valgrind.h>
-#endif
-
-#include <unistd.h>
-
-#include <gst/check/gstcheck.h>
-#include <gst/check/gstconsistencychecker.h>
-#include <gst/audio/audio.h>
-#include <gst/base/gstbasesrc.h>
-#include <gst/controller/gstdirectcontrolbinding.h>
-#include <gst/controller/gstinterpolationcontrolsource.h>
-
-static GMainLoop *main_loop;
-
-/* fixtures */
-
-static void
-test_setup (void)
-{
-  main_loop = g_main_loop_new (NULL, FALSE);
-}
-
-static void
-test_teardown (void)
-{
-  g_main_loop_unref (main_loop);
-  main_loop = NULL;
-}
-
-
-/* some test helpers */
-
-static GstElement *
-setup_pipeline (GstElement * audiomixer, gint num_srcs, GstElement * capsfilter)
-{
-  GstElement *pipeline, *src, *sink;
-  gint i;
-
-  pipeline = gst_pipeline_new ("pipeline");
-  if (!audiomixer) {
-    audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  }
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (pipeline), audiomixer, sink, NULL);
-
-  if (capsfilter) {
-    gst_bin_add (GST_BIN (pipeline), capsfilter);
-    gst_element_link_many (audiomixer, capsfilter, sink, NULL);
-  } else {
-    gst_element_link (audiomixer, sink);
-  }
-
-  for (i = 0; i < num_srcs; i++) {
-    src = gst_element_factory_make ("audiotestsrc", NULL);
-    g_object_set (src, "wave", 4, NULL);        /* silence */
-    gst_bin_add (GST_BIN (pipeline), src);
-    gst_element_link (src, audiomixer);
-  }
-  return pipeline;
-}
-
-static GstCaps *
-get_element_sink_pad_caps (GstElement * pipeline, const gchar * element_name)
-{
-  GstElement *sink;
-  GstCaps *caps;
-  GstPad *pad;
-
-  sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
-  pad = gst_element_get_static_pad (sink, "sink");
-  caps = gst_pad_get_current_caps (pad);
-  gst_object_unref (pad);
-  gst_object_unref (sink);
-
-  return caps;
-}
-
-static void
-set_state_and_wait (GstElement * pipeline, GstState state)
-{
-  GstStateChangeReturn state_res;
-
-  /* prepare paused/playing */
-  state_res = gst_element_set_state (pipeline, state);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* wait for preroll */
-  state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-}
-
-static gboolean
-set_playing (GstElement * element)
-{
-  GstStateChangeReturn state_res;
-
-  state_res = gst_element_set_state (element, GST_STATE_PLAYING);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  return FALSE;
-}
-
-static void
-play_and_wait (GstElement * pipeline)
-{
-  GstStateChangeReturn state_res;
-
-  g_idle_add ((GSourceFunc) set_playing, pipeline);
-
-  GST_INFO ("running main loop");
-  g_main_loop_run (main_loop);
-
-  state_res = gst_element_set_state (pipeline, GST_STATE_NULL);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-}
-
-static void
-message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
-{
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  switch (message->type) {
-    case GST_MESSAGE_EOS:
-      g_main_loop_quit (main_loop);
-      break;
-    case GST_MESSAGE_WARNING:{
-      GError *gerror;
-      gchar *debug;
-
-      gst_message_parse_warning (message, &gerror, &debug);
-      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
-      g_error_free (gerror);
-      g_free (debug);
-      break;
-    }
-    case GST_MESSAGE_ERROR:{
-      GError *gerror;
-      gchar *debug;
-
-      gst_message_parse_error (message, &gerror, &debug);
-      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
-      g_error_free (gerror);
-      g_free (debug);
-      g_main_loop_quit (main_loop);
-      break;
-    }
-    default:
-      break;
-  }
-}
-
-static GstBuffer *
-new_buffer (gsize num_bytes, gint data, GstClockTime ts, GstClockTime dur,
-    GstBufferFlags flags)
-{
-  GstMapInfo map;
-  GstBuffer *buffer = gst_buffer_new_and_alloc (num_bytes);
-
-  gst_buffer_map (buffer, &map, GST_MAP_WRITE);
-  memset (map.data, data, map.size);
-  gst_buffer_unmap (buffer, &map);
-  GST_BUFFER_TIMESTAMP (buffer) = ts;
-  GST_BUFFER_DURATION (buffer) = dur;
-  if (flags)
-    GST_BUFFER_FLAG_SET (buffer, flags);
-  GST_DEBUG ("created buffer %p", buffer);
-  return buffer;
-}
-
-/* make sure downstream gets a CAPS event before buffers are sent */
-GST_START_TEST (test_caps)
-{
-  GstElement *pipeline;
-  GstCaps *caps;
-
-  /* build pipeline */
-  pipeline = setup_pipeline (NULL, 1, NULL);
-
-  /* prepare playing */
-  set_state_and_wait (pipeline, GST_STATE_PAUSED);
-
-  /* check caps on fakesink */
-  caps = get_element_sink_pad_caps (pipeline, "sink");
-  fail_unless (caps != NULL);
-  gst_caps_unref (caps);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-/* check that caps set on the property are honoured */
-GST_START_TEST (test_filter_caps)
-{
-  GstElement *pipeline, *audiomixer, *capsfilter;
-  GstCaps *filter_caps, *caps;
-
-  filter_caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
-      "channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL);
-
-  capsfilter = gst_element_factory_make ("capsfilter", NULL);
-
-  /* build pipeline */
-  audiomixer = gst_element_factory_make ("audiomixer", NULL);
-  g_object_set (capsfilter, "caps", filter_caps, NULL);
-  pipeline = setup_pipeline (audiomixer, 1, capsfilter);
-
-  /* prepare playing */
-  set_state_and_wait (pipeline, GST_STATE_PAUSED);
-
-  /* check caps on fakesink */
-  caps = get_element_sink_pad_caps (pipeline, "sink");
-  fail_unless (caps != NULL);
-  GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps);
-  fail_unless (gst_caps_is_equal_fixed (caps, filter_caps));
-  gst_caps_unref (caps);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-
-  gst_caps_unref (filter_caps);
-}
-
-GST_END_TEST;
-
-static GstFormat format = GST_FORMAT_UNDEFINED;
-static gint64 position = -1;
-
-static void
-test_event_message_received (GstBus * bus, GstMessage * message,
-    GstPipeline * bin)
-{
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  switch (message->type) {
-    case GST_MESSAGE_SEGMENT_DONE:
-      gst_message_parse_segment_done (message, &format, &position);
-      GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position);
-      g_main_loop_quit (main_loop);
-      break;
-    default:
-      g_assert_not_reached ();
-      break;
-  }
-}
-
-
-GST_START_TEST (test_event)
-{
-  GstElement *bin, *src1, *src2, *audiomixer, *sink;
-  GstBus *bus;
-  GstEvent *seek_event;
-  gboolean res;
-  GstPad *srcpad, *sinkpad;
-  GstStreamConsistency *chk_1, *chk_2, *chk_3;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "wave", 4, NULL); /* silence */
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "wave", 4, NULL); /* silence */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
-
-  res = gst_element_link (src1, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (src2, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  chk_3 = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  /* create consistency checkers for the pads */
-  srcpad = gst_element_get_static_pad (src1, "src");
-  chk_1 = gst_consistency_checker_new (srcpad);
-  sinkpad = gst_pad_get_peer (srcpad);
-  gst_consistency_checker_add_pad (chk_3, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  srcpad = gst_element_get_static_pad (src2, "src");
-  chk_2 = gst_consistency_checker_new (srcpad);
-  sinkpad = gst_pad_get_peer (srcpad);
-  gst_consistency_checker_add_pad (chk_3, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  format = GST_FORMAT_UNDEFINED;
-  position = -1;
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) test_event_message_received, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  res = gst_element_send_event (bin, seek_event);
-  fail_unless (res == TRUE, NULL);
-
-  /* run pipeline */
-  play_and_wait (bin);
-
-  ck_assert_int_eq (position, 2 * GST_SECOND);
-
-  /* cleanup */
-  gst_consistency_checker_free (chk_1);
-  gst_consistency_checker_free (chk_2);
-  gst_consistency_checker_free (chk_3);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-static guint play_count = 0;
-static GstEvent *play_seek_event = NULL;
-
-static void
-test_play_twice_message_received (GstBus * bus, GstMessage * message,
-    GstElement * bin)
-{
-  gboolean res;
-  GstStateChangeReturn state_res;
-
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  switch (message->type) {
-    case GST_MESSAGE_SEGMENT_DONE:
-      play_count++;
-      if (play_count == 1) {
-        state_res = gst_element_set_state (bin, GST_STATE_READY);
-        ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-        /* prepare playing again */
-        set_state_and_wait (bin, GST_STATE_PAUSED);
-
-        res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-        fail_unless (res == TRUE, NULL);
-
-        state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
-        ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-      } else {
-        g_main_loop_quit (main_loop);
-      }
-      break;
-    default:
-      g_assert_not_reached ();
-      break;
-  }
-}
-
-
-GST_START_TEST (test_play_twice)
-{
-  GstElement *bin, *audiomixer;
-  GstBus *bus;
-  gboolean res;
-  GstPad *srcpad;
-  GstStreamConsistency *consist;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  bin = setup_pipeline (audiomixer, 2, NULL);
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  consist = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  play_count = 0;
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) test_play_twice_message_received, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-  fail_unless (res == TRUE, NULL);
-
-  GST_INFO ("seeked");
-
-  /* run pipeline */
-  play_and_wait (bin);
-
-  ck_assert_int_eq (play_count, 2);
-
-  /* cleanup */
-  gst_consistency_checker_free (consist);
-  gst_event_unref (play_seek_event);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_play_twice_then_add_and_play_again)
-{
-  GstElement *bin, *src, *audiomixer;
-  GstBus *bus;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  gint i;
-  GstPad *srcpad;
-  GstStreamConsistency *consist;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  bin = setup_pipeline (audiomixer, 2, NULL);
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  consist = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) test_play_twice_message_received, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  /* run it twice */
-  for (i = 0; i < 2; i++) {
-    play_count = 0;
-
-    GST_INFO ("starting test-loop %d", i);
-
-    /* prepare playing */
-    set_state_and_wait (bin, GST_STATE_PAUSED);
-
-    res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-    fail_unless (res == TRUE, NULL);
-
-    GST_INFO ("seeked");
-
-    /* run pipeline */
-    play_and_wait (bin);
-
-    ck_assert_int_eq (play_count, 2);
-
-    /* plug another source */
-    if (i == 0) {
-      src = gst_element_factory_make ("audiotestsrc", NULL);
-      g_object_set (src, "wave", 4, NULL);      /* silence */
-      gst_bin_add (GST_BIN (bin), src);
-
-      res = gst_element_link (src, audiomixer);
-      fail_unless (res == TRUE, NULL);
-    }
-
-    gst_consistency_checker_reset (consist);
-  }
-
-  state_res = gst_element_set_state (bin, GST_STATE_NULL);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* cleanup */
-  gst_event_unref (play_seek_event);
-  gst_consistency_checker_free (consist);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-
-static GstElement *
-test_live_seeking_try_audiosrc (const gchar * factory_name)
-{
-  GstElement *src;
-  GstStateChangeReturn state_res;
-
-  if (!(src = gst_element_factory_make (factory_name, NULL))) {
-    GST_INFO ("can't make '%s', skipping", factory_name);
-    return NULL;
-  }
-
-  /* Test that the audio source can get to ready, else skip */
-  state_res = gst_element_set_state (src, GST_STATE_READY);
-  gst_element_set_state (src, GST_STATE_NULL);
-
-  if (state_res == GST_STATE_CHANGE_FAILURE) {
-    GST_INFO_OBJECT (src, "can't go to ready, skipping");
-    gst_object_unref (src);
-    return NULL;
-  }
-
-  return src;
-}
-
-/* test failing seeks on live-sources */
-GST_START_TEST (test_live_seeking)
-{
-  GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink;
-  GstCaps *caps;
-  GstBus *bus;
-  gboolean res;
-  GstPad *srcpad;
-  GstPad *sinkpad;
-  gint i;
-  GstStreamConsistency *consist;
-  /* don't use autoaudiosrc, as then we can't set anything here */
-  const gchar *audio_src_factories[] = {
-    "alsasrc",
-    "pulseaudiosrc"
-  };
-
-  GST_INFO ("preparing test");
-  play_seek_event = NULL;
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  for (i = 0; (i < G_N_ELEMENTS (audio_src_factories) && src1 == NULL); i++) {
-    src1 = test_live_seeking_try_audiosrc (audio_src_factories[i]);
-  }
-  if (!src1) {
-    /* normal audiosources behave differently than audiotestsrc */
-    GST_WARNING ("no real audiosrc found, using audiotestsrc is-live");
-    src1 = gst_element_factory_make ("audiotestsrc", "src1");
-    g_object_set (src1, "wave", 4, "is-live", TRUE, NULL);      /* silence */
-  } else {
-    /* live sources ignore seeks, force eos after 2 sec (4 buffers half second
-     * each)
-     */
-    g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL);
-  }
-
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  cf = gst_element_factory_make ("capsfilter", "capsfilter");
-  sink = gst_element_factory_make ("fakesink", "sink");
-
-  gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL);
-  res = gst_element_link_many (src1, cf, audiomixer, sink, NULL);
-  fail_unless (res == TRUE, NULL);
-
-  /* get the caps for the livesrc, we'll reuse this for the non-live source */
-  set_state_and_wait (bin, GST_STATE_PLAYING);
-
-  sinkpad = gst_element_get_static_pad (sink, "sink");
-  fail_unless (sinkpad != NULL);
-  caps = gst_pad_get_current_caps (sinkpad);
-  fail_unless (caps != NULL);
-  gst_object_unref (sinkpad);
-
-  gst_element_set_state (bin, GST_STATE_NULL);
-
-  g_object_set (cf, "caps", caps, NULL);
-
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "wave", 4, NULL); /* silence */
-  gst_bin_add (GST_BIN (bin), src2);
-
-  res = gst_element_link_filtered (src2, audiomixer, caps);
-  fail_unless (res == TRUE, NULL);
-
-  gst_caps_unref (caps);
-
-  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  consist = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  GST_INFO ("starting test");
-
-  /* run it twice */
-  for (i = 0; i < 2; i++) {
-
-    GST_INFO ("starting test-loop %d", i);
-
-    /* prepare playing */
-    set_state_and_wait (bin, GST_STATE_PAUSED);
-
-    res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-    fail_unless (res == TRUE, NULL);
-
-    GST_INFO ("seeked");
-
-    /* run pipeline */
-    play_and_wait (bin);
-
-    gst_consistency_checker_reset (consist);
-  }
-
-  /* cleanup */
-  GST_INFO ("cleaning up");
-  gst_consistency_checker_free (consist);
-  if (play_seek_event)
-    gst_event_unref (play_seek_event);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-/* check if adding pads work as expected */
-GST_START_TEST (test_add_pad)
-{
-  GstElement *bin, *src1, *src2, *audiomixer, *sink;
-  GstBus *bus;
-  GstPad *srcpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "num-buffers", 4, "wave", /* silence */ 4, NULL);
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  /* one buffer less, we connect with 1 buffer of delay */
-  g_object_set (src2, "num-buffers", 3, "wave", /* silence */ 4, NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL);
-
-  res = gst_element_link (src1, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  gst_object_unref (srcpad);
-
-  g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
-      bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  /* add other element */
-  gst_bin_add_many (GST_BIN (bin), src2, NULL);
-
-  /* now link the second element */
-  res = gst_element_link (src2, audiomixer);
-  fail_unless (res == TRUE, NULL);
-
-  /* set to PAUSED as well */
-  state_res = gst_element_set_state (src2, GST_STATE_PAUSED);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* now play all */
-  play_and_wait (bin);
-
-  /* cleanup */
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-/* check if removing pads work as expected */
-GST_START_TEST (test_remove_pad)
-{
-  GstElement *bin, *src, *audiomixer, *sink;
-  GstBus *bus;
-  GstPad *pad, *srcpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  src = gst_element_factory_make ("audiotestsrc", "src");
-  g_object_set (src, "num-buffers", 4, "wave", 4, NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL);
-
-  res = gst_element_link (src, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  /* create an unconnected sinkpad in audiomixer */
-  pad = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (pad == NULL, NULL);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  gst_object_unref (srcpad);
-
-  g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
-      bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing, this will not preroll as audiomixer is waiting
-   * on the unconnected sinkpad. */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* wait for completion for one second, will return ASYNC */
-  state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND);
-  ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC);
-
-  /* get rid of the pad now, audiomixer should stop waiting on it and
-   * continue the preroll */
-  gst_element_release_request_pad (audiomixer, pad);
-  gst_object_unref (pad);
-
-  /* wait for completion, should work now */
-  state_res =
-      gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
-      GST_CLOCK_TIME_NONE);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* now play all */
-  play_and_wait (bin);
-
-  /* cleanup */
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (G_OBJECT (bus));
-  gst_object_unref (G_OBJECT (bin));
-}
-
-GST_END_TEST;
-
-
-static GstBuffer *handoff_buffer = NULL;
-
-static void
-handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
-    gpointer user_data)
-{
-  GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT
-      " -- %p PTS is %" GST_TIME_FORMAT " END is %" GST_TIME_FORMAT,
-      gst_buffer_get_size (buffer), buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-
-  gst_buffer_replace (&handoff_buffer, buffer);
-}
-
-/* check if clipping works as expected */
-GST_START_TEST (test_clip)
-{
-  GstSegment segment;
-  GstElement *bin, *audiomixer, *sink;
-  GstBus *bus;
-  GstPad *sinkpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  GstFlowReturn ret;
-  GstEvent *event;
-  GstBuffer *buffer;
-  GstCaps *caps;
-  GstQuery *drain = gst_query_new_drain ();
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  /* just an audiomixer and a fakesink */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  g_object_set (audiomixer, "output-buffer-duration", 50 * GST_MSECOND, NULL);
-  sink = gst_element_factory_make ("fakesink", "sink");
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
-  gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
-
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  /* set to playing */
-  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* create an unconnected sinkpad in audiomixer, should also automatically activate
-   * the pad */
-  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad == NULL, NULL);
-
-  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
-
-  gst_pad_set_caps (sinkpad, caps);
-  gst_caps_unref (caps);
-
-  /* send segment to audiomixer */
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  segment.start = GST_SECOND;
-  segment.stop = 2 * GST_SECOND;
-  segment.time = 0;
-  event = gst_event_new_segment (&segment);
-  gst_pad_send_event (sinkpad, event);
-
-  /* should be clipped and ok */
-  buffer = new_buffer (44100, 0, 0, 250 * GST_MSECOND, 0);
-  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
-      buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  /* The aggregation is done in a dedicated thread, so we can't
-   * know when it is actually going to happen, so we use a DRAIN query
-   * to wait for it to complete.
-   */
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer == NULL);
-
-  /* should be partially clipped */
-  buffer = new_buffer (44100, 0, 900 * GST_MSECOND, 250 * GST_MSECOND,
-      GST_BUFFER_FLAG_DISCONT);
-  GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
-      GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
-      GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-
-  fail_unless (handoff_buffer != NULL);
-  ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
-      GST_BUFFER_DURATION (handoff_buffer), 150 * GST_MSECOND);
-  gst_buffer_replace (&handoff_buffer, NULL);
-
-  /* should not be clipped */
-  buffer = new_buffer (44100, 0, 1150 * GST_MSECOND, 250 * GST_MSECOND, 0);
-  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
-      buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer != NULL);
-  ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
-      GST_BUFFER_DURATION (handoff_buffer), 400 * GST_MSECOND);
-  gst_buffer_replace (&handoff_buffer, NULL);
-  fail_unless (handoff_buffer == NULL);
-
-  /* should be clipped and ok */
-  buffer = new_buffer (44100, 0, 2 * GST_SECOND, 250 * GST_MSECOND,
-      GST_BUFFER_FLAG_DISCONT);
-  GST_DEBUG ("pushing buffer %p PTS is %" GST_TIME_FORMAT
-      " END is %" GST_TIME_FORMAT,
-      buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer == NULL);
-
-  gst_element_release_request_pad (audiomixer, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-  gst_query_unref (drain);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_duration_is_max)
-{
-  GstElement *bin, *src[3], *audiomixer, *sink;
-  GstStateChangeReturn state_res;
-  GstFormat format = GST_FORMAT_TIME;
-  gboolean res;
-  gint64 duration;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-
-  /* 3 sources, an audiomixer and a fakesink */
-  src[0] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[1] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[2] = gst_element_factory_make ("audiotestsrc", NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
-      NULL);
-
-  gst_element_link (src[0], audiomixer);
-  gst_element_link (src[1], audiomixer);
-  gst_element_link (src[2], audiomixer);
-  gst_element_link (audiomixer, sink);
-
-  /* irks, duration is reset on basesrc */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
-
-  /* set durations on src */
-  GST_BASE_SRC (src[0])->segment.duration = 1000;
-  GST_BASE_SRC (src[1])->segment.duration = 3000;
-  GST_BASE_SRC (src[2])->segment.duration = 2000;
-
-  /* set to playing */
-  set_state_and_wait (bin, GST_STATE_PLAYING);
-
-  res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
-  fail_unless (res, NULL);
-
-  ck_assert_int_eq (duration, 3000);
-
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_duration_unknown_overrides)
-{
-  GstElement *bin, *src[3], *audiomixer, *sink;
-  GstStateChangeReturn state_res;
-  GstFormat format = GST_FORMAT_TIME;
-  gboolean res;
-  gint64 duration;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-
-  /* 3 sources, an audiomixer and a fakesink */
-  src[0] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[1] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[2] = gst_element_factory_make ("audiotestsrc", NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
-      NULL);
-
-  gst_element_link (src[0], audiomixer);
-  gst_element_link (src[1], audiomixer);
-  gst_element_link (src[2], audiomixer);
-  gst_element_link (audiomixer, sink);
-
-  /* irks, duration is reset on basesrc */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
-
-  /* set durations on src */
-  GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE;
-  GST_BASE_SRC (src[1])->segment.duration = 3000;
-  GST_BASE_SRC (src[2])->segment.duration = 2000;
-
-  /* set to playing */
-  set_state_and_wait (bin, GST_STATE_PLAYING);
-
-  res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
-  fail_unless (res, NULL);
-
-  ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE);
-
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-
-static gboolean looped = FALSE;
-
-static void
-loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin)
-{
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  if (looped) {
-    g_main_loop_quit (main_loop);
-  } else {
-    GstEvent *seek_event;
-    gboolean res;
-
-    seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-        GST_SEEK_FLAG_SEGMENT,
-        GST_SEEK_TYPE_SET, (GstClockTime) 0,
-        GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
-
-    res = gst_element_send_event (bin, seek_event);
-    fail_unless (res == TRUE, NULL);
-    looped = TRUE;
-  }
-}
-
-GST_START_TEST (test_loop)
-{
-  GstElement *bin;
-  GstBus *bus;
-  GstEvent *seek_event;
-  gboolean res;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = setup_pipeline (NULL, 2, NULL);
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) loop_segment_done, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  res = gst_element_send_event (bin, seek_event);
-  fail_unless (res == TRUE, NULL);
-
-  /* run pipeline */
-  play_and_wait (bin);
-
-  fail_unless (looped);
-
-  /* cleanup */
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_flush_start_flush_stop)
-{
-  GstPadTemplate *sink_template;
-  GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src;
-  GstElement *pipeline, *src1, *src2, *audiomixer, *sink;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  pipeline = gst_pipeline_new ("pipeline");
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "wave", 4, NULL); /* silence */
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "wave", 4, NULL); /* silence */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL);
-
-  sink_template =
-      gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer),
-      "sink_%u");
-  fail_unless (GST_IS_PAD_TEMPLATE (sink_template));
-  sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
-  srcpad1 = gst_element_get_static_pad (src1, "src");
-  gst_pad_link (srcpad1, sinkpad1);
-
-  sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
-  tmppad = gst_element_get_static_pad (src2, "src");
-  gst_pad_link (tmppad, sinkpad2);
-  gst_object_unref (tmppad);
-
-  gst_element_link (audiomixer, sink);
-
-  /* prepare playing */
-  set_state_and_wait (pipeline, GST_STATE_PLAYING);
-
-  audiomixer_src = gst_element_get_static_pad (audiomixer, "src");
-  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
-  gst_pad_send_event (sinkpad1, gst_event_new_flush_start ());
-  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
-  fail_unless (GST_PAD_IS_FLUSHING (sinkpad1));
-  /* Hold the streamlock to make sure the flush stop is not between
-     the attempted push of a segment event and of the following buffer. */
-  GST_PAD_STREAM_LOCK (srcpad1);
-  gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE));
-  GST_PAD_STREAM_UNLOCK (srcpad1);
-  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
-  fail_if (GST_PAD_IS_FLUSHING (sinkpad1));
-  gst_object_unref (audiomixer_src);
-
-  gst_element_release_request_pad (audiomixer, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_element_release_request_pad (audiomixer, sinkpad2);
-  gst_object_unref (sinkpad2);
-  gst_object_unref (srcpad1);
-
-  /* cleanup */
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer,
-    GstPad * pad, gpointer user_data)
-{
-  GList **received_buffers = user_data;
-
-  GST_DEBUG ("got buffer %p", buffer);
-  *received_buffers =
-      g_list_append (*received_buffers, gst_buffer_ref (buffer));
-}
-
-typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2);
-typedef void (*CheckBuffersFunction) (GList * buffers);
-
-static void
-run_sync_test (SendBuffersFunction send_buffers,
-    CheckBuffersFunction check_buffers)
-{
-  GstSegment segment;
-  GstElement *bin, *audiomixer, *queue1, *queue2, *sink;
-  GstBus *bus;
-  GstPad *sinkpad1, *sinkpad2;
-  GstPad *queue1_sinkpad, *queue2_sinkpad;
-  GstPad *pad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  GstEvent *event;
-  GstCaps *caps;
-  GList *received_buffers = NULL;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  /* just an audiomixer and a fakesink */
-  queue1 = gst_element_factory_make ("queue", "queue1");
-  queue2 = gst_element_factory_make ("queue", "queue2");
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL);
-  sink = gst_element_factory_make ("fakesink", "sink");
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,
-      &received_buffers);
-  gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL);
-
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  /* set to paused */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* create an unconnected sinkpad in audiomixer, should also automatically activate
-   * the pad */
-  sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad1 == NULL, NULL);
-
-  queue1_sinkpad = gst_element_get_static_pad (queue1, "sink");
-  pad = gst_element_get_static_pad (queue1, "src");
-  fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (pad);
-
-  sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad2 == NULL, NULL);
-
-  queue2_sinkpad = gst_element_get_static_pad (queue2, "sink");
-  pad = gst_element_get_static_pad (queue2, "src");
-  fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK);
-  gst_object_unref (pad);
-
-  gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test"));
-  gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test"));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
-
-  gst_pad_set_caps (queue1_sinkpad, caps);
-  gst_pad_set_caps (queue2_sinkpad, caps);
-  gst_caps_unref (caps);
-
-  /* send segment to audiomixer */
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  event = gst_event_new_segment (&segment);
-  gst_pad_send_event (queue1_sinkpad, gst_event_ref (event));
-  gst_pad_send_event (queue2_sinkpad, event);
-
-  /* Push buffers */
-  send_buffers (queue1_sinkpad, queue2_sinkpad);
-
-  /* Set PLAYING */
-  g_idle_add ((GSourceFunc) set_playing, bin);
-
-  /* Collect buffers and messages */
-  g_main_loop_run (main_loop);
-
-  /* Here we get once we got EOS, for errors we failed */
-
-  check_buffers (received_buffers);
-
-  g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref);
-
-  gst_element_release_request_pad (audiomixer, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (queue1_sinkpad);
-  gst_element_release_request_pad (audiomixer, sinkpad2);
-  gst_object_unref (sinkpad2);
-  gst_object_unref (queue2_sinkpad);
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-static void
-send_buffers_sync (GstPad * pad1, GstPad * pad2)
-{
-  GstBuffer *buffer;
-  GstFlowReturn ret;
-
-  buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 1, 2 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad1, gst_event_new_eos ());
-
-  buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad2, gst_event_new_eos ());
-}
-
-static void
-check_buffers_sync (GList * received_buffers)
-{
-  GstBuffer *buffer;
-  GList *l;
-  gint i;
-  GstMapInfo map;
-
-  /* Should have 8 * 0.5s buffers */
-  fail_unless_equals_int (g_list_length (received_buffers), 8);
-  for (i = 0, l = received_buffers; l; l = l->next, i++) {
-    buffer = l->data;
-
-    gst_buffer_map (buffer, &map, GST_MAP_READ);
-
-    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else {
-      g_assert_not_reached ();
-    }
-
-    gst_buffer_unmap (buffer, &map);
-
-  }
-}
-
-GST_START_TEST (test_sync)
-{
-  run_sync_test (send_buffers_sync, check_buffers_sync);
-}
-
-GST_END_TEST;
-
-static void
-send_buffers_sync_discont (GstPad * pad1, GstPad * pad2)
-{
-  GstBuffer *buffer;
-  GstFlowReturn ret;
-
-  buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 1, 3 * GST_SECOND, 1 * GST_SECOND,
-      GST_BUFFER_FLAG_DISCONT);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad1, gst_event_new_eos ());
-
-  buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad2, gst_event_new_eos ());
-}
-
-static void
-check_buffers_sync_discont (GList * received_buffers)
-{
-  GstBuffer *buffer;
-  GList *l;
-  gint i;
-  GstMapInfo map;
-
-  /* Should have 8 * 0.5s buffers */
-  fail_unless_equals_int (g_list_length (received_buffers), 8);
-  for (i = 0, l = received_buffers; l; l = l->next, i++) {
-    buffer = l->data;
-
-    gst_buffer_map (buffer, &map, GST_MAP_READ);
-
-    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else {
-      g_assert_not_reached ();
-    }
-
-    gst_buffer_unmap (buffer, &map);
-
-  }
-}
-
-GST_START_TEST (test_sync_discont)
-{
-  run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont);
-}
-
-GST_END_TEST;
-
-static void
-send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2)
-{
-  GstBuffer *buffer;
-  GstFlowReturn ret;
-
-  buffer = new_buffer (2000, 1, 750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 1, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad1, gst_event_new_eos ());
-
-  buffer = new_buffer (2000, 2, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 2, 2750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad2, gst_event_new_eos ());
-}
-
-static void
-check_buffers_sync_unaligned (GList * received_buffers)
-{
-  GstBuffer *buffer;
-  GList *l;
-  gint i;
-  GstMapInfo map;
-
-  /* Should have 8 * 0.5s buffers */
-  fail_unless_equals_int (g_list_length (received_buffers), 8);
-  for (i = 0, l = received_buffers; l; l = l->next, i++) {
-    buffer = l->data;
-
-    gst_buffer_map (buffer, &map, GST_MAP_READ);
-
-    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[499] == 0);
-      fail_unless (map.data[500] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[499] == 1);
-      fail_unless (map.data[500] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[499] == 3);
-      fail_unless (map.data[500] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[499] == 3);
-      fail_unless (map.data[500] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[499] == 2);
-      fail_unless (map.data[500] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
-      fail_unless (map.size == 500);
-      fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND);
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[499] == 2);
-    } else {
-      g_assert_not_reached ();
-    }
-
-    gst_buffer_unmap (buffer, &map);
-
-  }
-}
-
-GST_START_TEST (test_sync_unaligned)
-{
-  run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_segment_base_handling)
-{
-  GstElement *pipeline, *sink, *mix, *src1, *src2;
-  GstPad *srcpad, *sinkpad;
-  GstClockTime end_time;
-  GstSample *last_sample = NULL;
-  GstSample *sample;
-  GstBuffer *buf;
-  GstCaps *caps;
-
-  caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100,
-      "channels", G_TYPE_INT, 2, NULL);
-
-  pipeline = gst_pipeline_new ("pipeline");
-  mix = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("appsink", "sink");
-  g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
-  gst_caps_unref (caps);
-  /* 50 buffers of 1/10 sec = 5 sec */
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
-  gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL);
-  fail_unless (gst_element_link (mix, sink));
-
-  srcpad = gst_element_get_static_pad (src1, "src");
-  sinkpad = gst_element_get_request_pad (mix, "sink_1");
-  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  srcpad = gst_element_get_static_pad (src2, "src");
-  sinkpad = gst_element_get_request_pad (mix, "sink_2");
-  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
-  /* set a pad offset of another 5 seconds */
-  gst_pad_set_offset (sinkpad, 5 * GST_SECOND);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  do {
-    g_signal_emit_by_name (sink, "pull-sample", &sample);
-    if (sample == NULL)
-      break;
-    if (last_sample)
-      gst_sample_unref (last_sample);
-    last_sample = sample;
-  } while (TRUE);
-
-  buf = gst_sample_get_buffer (last_sample);
-  end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
-  fail_unless_equals_int64 (end_time, 10 * GST_SECOND);
-  gst_sample_unref (last_sample);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
-    GstClockTime end, gdouble end_value)
-{
-  GstControlSource *cs;
-  GstTimedValueControlSource *tvcs;
-
-  cs = gst_interpolation_control_source_new ();
-  fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
-          gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
-              "volume", cs)));
-
-  /* set volume interpolation mode */
-  g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
-
-  tvcs = (GstTimedValueControlSource *) cs;
-  fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
-  fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
-  gst_object_unref (cs);
-}
-
-GST_START_TEST (test_sinkpad_property_controller)
-{
-  GstBus *bus;
-  GstMessage *msg;
-  GstElement *pipeline, *sink, *mix, *src1;
-  GstPad *srcpad, *sinkpad;
-  GError *error = NULL;
-  gchar *debug;
-
-  pipeline = gst_pipeline_new ("pipeline");
-  mix = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "num-buffers", 100, NULL);
-  gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
-  fail_unless (gst_element_link (mix, sink));
-
-  srcpad = gst_element_get_static_pad (src1, "src");
-  sinkpad = gst_element_get_request_pad (mix, "sink_0");
-  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
-  set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
-  msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
-      GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
-  switch (GST_MESSAGE_TYPE (msg)) {
-    case GST_MESSAGE_ERROR:
-      gst_message_parse_error (msg, &error, &debug);
-      g_printerr ("ERROR from element %s: %s\n",
-          GST_OBJECT_NAME (msg->src), error->message);
-      g_printerr ("Debug info: %s\n", debug);
-      g_error_free (error);
-      g_free (debug);
-      break;
-    case GST_MESSAGE_EOS:
-      break;
-    default:
-      g_assert_not_reached ();
-  }
-  gst_message_unref (msg);
-  g_object_unref (bus);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-change_src_caps (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
-    GstElement * capsfilter)
-{
-  GstCaps *caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
-
-  g_object_set (capsfilter, "caps", caps, NULL);
-  g_signal_connect (fakesink, "handoff", (GCallback) handoff_buffer_cb, NULL);
-  g_signal_handlers_disconnect_by_func (fakesink, change_src_caps, capsfilter);
-}
-
-/* In this test, we create an input buffer with a duration of 2 seconds,
- * and require the audiomixer to output 1 second long buffers.
- * The input buffer will thus be mixed twice, and the audiomixer will
- * output two buffers.
- *
- * After audiomixer has output a first buffer, we change its output format
- * from S8 to S32. As our sample rate stays the same at 10 fps, and we use
- * mono, the first buffer should be 10 bytes long, and the second 40.
- *
- * The input buffer is made up of 15 0-valued bytes, and 5 1-valued bytes.
- * We verify that the second buffer contains 5 0-valued integers, and
- * 5 1 << 24 valued integers.
- */
-GST_START_TEST (test_change_output_caps)
-{
-  GstSegment segment;
-  GstElement *bin, *audiomixer, *capsfilter, *sink;
-  GstBus *bus;
-  GstPad *sinkpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  GstFlowReturn ret;
-  GstEvent *event;
-  GstBuffer *buffer;
-  GstCaps *caps;
-  GstQuery *drain = gst_query_new_drain ();
-  GstMapInfo inmap;
-  GstMapInfo outmap;
-  gsize i;
-
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  g_object_set (audiomixer, "output-buffer-duration", GST_SECOND, NULL);
-  capsfilter = gst_element_factory_make ("capsfilter", NULL);
-  sink = gst_element_factory_make ("fakesink", "sink");
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) change_src_caps, capsfilter);
-  gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
-
-  res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
-  fail_unless (res == TRUE, NULL);
-
-  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad == NULL, NULL);
-
-  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, "S8",
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
-
-  gst_pad_set_caps (sinkpad, caps);
-  g_object_set (capsfilter, "caps", caps, NULL);
-  gst_caps_unref (caps);
-
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  segment.start = 0;
-  segment.stop = 2 * GST_SECOND;
-  segment.time = 0;
-  event = gst_event_new_segment (&segment);
-  gst_pad_send_event (sinkpad, event);
-
-  gst_buffer_replace (&handoff_buffer, NULL);
-
-  buffer = new_buffer (20, 0, 0, 2 * GST_SECOND, 0);
-  gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
-  memset (inmap.data + 15, 1, 5);
-  gst_buffer_unmap (buffer, &inmap);
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer != NULL);
-  fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 40);
-
-  gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
-  for (i = 0; i < 10; i++) {
-    guint32 sample;
-
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
-    sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
-#else
-    sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
-#endif
-
-    if (i < 5) {
-      fail_unless_equals_int (sample, 0);
-    } else {
-      fail_unless_equals_int (sample, 1 << 24);
-    }
-  }
-  gst_buffer_unmap (handoff_buffer, &outmap);
-
-  gst_element_release_request_pad (audiomixer, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-  gst_query_unref (drain);
-}
-
-GST_END_TEST;
-
-static Suite *
-audiomixer_suite (void)
-{
-  Suite *s = suite_create ("audiomixer");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_caps);
-  tcase_add_test (tc_chain, test_filter_caps);
-  tcase_add_test (tc_chain, test_event);
-  tcase_add_test (tc_chain, test_play_twice);
-  tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again);
-  tcase_add_test (tc_chain, test_live_seeking);
-  tcase_add_test (tc_chain, test_add_pad);
-  tcase_add_test (tc_chain, test_remove_pad);
-  tcase_add_test (tc_chain, test_clip);
-  tcase_add_test (tc_chain, test_duration_is_max);
-  tcase_add_test (tc_chain, test_duration_unknown_overrides);
-  tcase_add_test (tc_chain, test_loop);
-  tcase_add_test (tc_chain, test_flush_start_flush_stop);
-  tcase_add_test (tc_chain, test_sync);
-  tcase_add_test (tc_chain, test_sync_discont);
-  tcase_add_test (tc_chain, test_sync_unaligned);
-  tcase_add_test (tc_chain, test_segment_base_handling);
-  tcase_add_test (tc_chain, test_sinkpad_property_controller);
-  tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
-  tcase_add_test (tc_chain, test_change_output_caps);
-
-  /* Use a longer timeout */
-#ifdef HAVE_VALGRIND
-  if (RUNNING_ON_VALGRIND) {
-    tcase_set_timeout (tc_chain, 5 * 60);
-  } else
-#endif
-  {
-    /* this is shorter than the default 60 seconds?! (tpm) */
-    /* tcase_set_timeout (tc_chain, 6); */
-  }
-
-  return s;
-}
-
-GST_CHECK_MAIN (audiomixer);
diff --git a/tests/check/meson.build b/tests/check/meson.build
index 55f1513e8..1cb817164 100644
--- a/tests/check/meson.build
+++ b/tests/check/meson.build
@@ -18,8 +18,6 @@ base_tests = [
   [['elements/aiffparse.c']],
   [['elements/asfmux.c']],
   [['elements/assrender.c'], not ass_dep.found(), [ass_dep]],
-  [['elements/audiointerleave.c']],
-  [['elements/audiomixer.c']],
   [['elements/autoconvert.c']],
   [['elements/autovideoconvert.c']],
   [['elements/camerabin.c']],