audioaggregator, audiomixer, audiointerleave: move from -bad to -base

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

9 files changed, 4606 insertions, 0 deletions

diff --git a/gst/audiomixer/Makefile.am b/gst/audiomixer/Makefile.am
new file mode 100644
index 000000000..f1a4d7395
--- /dev/null
+++ b/gst/audiomixer/Makefile.am
@@ -0,0 +1,21 @@
+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
new file mode 100644
index 000000000..90ec363ea
--- /dev/null
+++ b/gst/audiomixer/gstaudiointerleave.c
@@ -0,0 +1,902 @@
+/* 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
new file mode 100644
index 000000000..bf46f4a50
--- /dev/null
+++ b/gst/audiomixer/gstaudiointerleave.h
@@ -0,0 +1,100 @@
+/* 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
new file mode 100644
index 000000000..a0f569010
--- /dev/null
+++ b/gst/audiomixer/gstaudiomixer.c
@@ -0,0 +1,577 @@
+/* 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
new file mode 100644
index 000000000..67ccb27e6
--- /dev/null
+++ b/gst/audiomixer/gstaudiomixer.h
@@ -0,0 +1,87 @@
+/* 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
new file mode 100644
index 000000000..be377f705
--- /dev/null
+++ b/gst/audiomixer/gstaudiomixerorc-dist.c
@@ -0,0 +1,2605 @@
+
+/* 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
new file mode 100644
index 000000000..af0de0139
--- /dev/null
+++ b/gst/audiomixer/gstaudiomixerorc-dist.h
@@ -0,0 +1,106 @@
+
+/* 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
new file mode 100644
index 000000000..5eaff2395
--- /dev/null
+++ b/gst/audiomixer/gstaudiomixerorc.orc
@@ -0,0 +1,176 @@
+.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
new file mode 100644
index 000000000..ccfe1b9d3
--- /dev/null
+++ b/gst/audiomixer/meson.build
@@ -0,0 +1,32 @@
+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,
+)