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

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

13 files changed, 9851 insertions, 0 deletions

diff --git a/gst-libs/gst/audio/gstaudioaggregator.c b/gst-libs/gst/audio/gstaudioaggregator.c
new file mode 100644
index 000000000..fa9911b31
--- /dev/null
+++ b/gst-libs/gst/audio/gstaudioaggregator.c
@@ -0,0 +1,1995 @@
+/* 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>
+ *                    2014 Collabora
+ *                             Olivier Crete <olivier.crete@collabora.com>
+ *
+ * gstaudioaggregator.c:
+ *
+ * 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: gstaudioaggregator
+ * @short_description: manages a set of pads with the purpose of
+ * aggregating their buffers for raw audio
+ * @see_also: #GstAggregator
+ *
+ * #GstAudioAggregator will perform conversion on the data arriving
+ * on its sink pads, based on the format expected downstream.
+ *
+ * Subclasses can opt out of the conversion behaviour by setting
+ * #GstAudioAggregator.convert_buffer() to %NULL.
+ *
+ * Subclasses that wish to use the default conversion implementation
+ * should use a (subclass of) #GstAudioAggregatorConvertPad as their
+ * #GstAggregatorClass.sinkpads_type, as it will cache the created
+ * #GstAudioConverter and install a property allowing to configure it,
+ * #GstAudioAggregatorPadClass:converter-config.
+ *
+ * Subclasses that wish to perform custom conversion should override
+ * #GstAudioAggregator.convert_buffer().
+ *
+ * When conversion is enabled, #GstAudioAggregator will accept
+ * any type of raw audio caps and perform conversion
+ * on the data arriving on its sink pads, with whatever downstream
+ * expects as the target format.
+ *
+ * In case downstream caps are not fully fixated, it will use
+ * the first configured sink pad to finish fixating its source pad
+ * caps.
+ *
+ * Additionally, handling audio conversion directly in the element
+ * means that this base class supports safely reconfiguring its
+ * source pad.
+ *
+ * A notable exception for now is the sample rate, sink pads must
+ * have the same sample rate as either the downstream requirement,
+ * or the first configured pad, or a combination of both (when
+ * downstream specifies a range or a set of acceptable rates).
+ */
+
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include "gstaudioaggregator.h"
+
+#include <string.h>
+
+GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug);
+#define GST_CAT_DEFAULT audio_aggregator_debug
+
+struct _GstAudioAggregatorPadPrivate
+{
+  /* All members are protected by the pad object lock */
+
+  GstBuffer *buffer;            /* current buffer we're mixing, for
+                                   comparison with a new input buffer from
+                                   aggregator to see if we need to update our
+                                   cached values. */
+
+  guint position, size;         /* position in the input buffer and size of the
+                                   input buffer in number of samples */
+
+  GstBuffer *input_buffer;
+
+  guint64 output_offset;        /* Sample offset in output segment relative to
+                                   pad.segment.start that position refers to
+                                   in the current buffer. */
+
+  guint64 next_offset;          /* Next expected sample offset relative to
+                                   pad.segment.start */
+
+  /* Last time we noticed a discont */
+  GstClockTime discont_time;
+
+  /* A new unhandled segment event has been received */
+  gboolean new_segment;
+};
+
+
+/*****************************************
+ * GstAudioAggregatorPad implementation  *
+ *****************************************/
+G_DEFINE_TYPE (GstAudioAggregatorPad, gst_audio_aggregator_pad,
+    GST_TYPE_AGGREGATOR_PAD);
+
+enum
+{
+  PROP_PAD_0,
+  PROP_PAD_CONVERTER_CONFIG,
+};
+
+static GstFlowReturn
+gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
+    GstAggregator * aggregator);
+
+static void
+gst_audio_aggregator_pad_finalize (GObject * object)
+{
+  GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object;
+
+  gst_buffer_replace (&pad->priv->buffer, NULL);
+  gst_buffer_replace (&pad->priv->input_buffer, NULL);
+
+  G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object);
+}
+
+static void
+gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass;
+
+  g_type_class_add_private (klass, sizeof (GstAudioAggregatorPadPrivate));
+
+  gobject_class->finalize = gst_audio_aggregator_pad_finalize;
+  aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad);
+}
+
+static void
+gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad)
+{
+  pad->priv =
+      G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_PAD,
+      GstAudioAggregatorPadPrivate);
+
+  gst_audio_info_init (&pad->info);
+
+  pad->priv->buffer = NULL;
+  pad->priv->input_buffer = NULL;
+  pad->priv->position = 0;
+  pad->priv->size = 0;
+  pad->priv->output_offset = -1;
+  pad->priv->next_offset = -1;
+  pad->priv->discont_time = GST_CLOCK_TIME_NONE;
+}
+
+
+static GstFlowReturn
+gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
+    GstAggregator * aggregator)
+{
+  GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
+
+  GST_OBJECT_LOCK (aggpad);
+  pad->priv->position = pad->priv->size = 0;
+  pad->priv->output_offset = pad->priv->next_offset = -1;
+  pad->priv->discont_time = GST_CLOCK_TIME_NONE;
+  gst_buffer_replace (&pad->priv->buffer, NULL);
+  gst_buffer_replace (&pad->priv->input_buffer, NULL);
+  GST_OBJECT_UNLOCK (aggpad);
+
+  return GST_FLOW_OK;
+}
+
+struct _GstAudioAggregatorConvertPadPrivate
+{
+  /* All members are protected by the pad object lock */
+  GstAudioConverter *converter;
+  GstStructure *converter_config;
+  gboolean converter_config_changed;
+};
+
+
+G_DEFINE_TYPE (GstAudioAggregatorConvertPad, gst_audio_aggregator_convert_pad,
+    GST_TYPE_AUDIO_AGGREGATOR_PAD);
+
+static void
+gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
+    * aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info)
+{
+  if (!aaggcpad->priv->converter_config_changed)
+    return;
+
+  if (aaggcpad->priv->converter) {
+    gst_audio_converter_free (aaggcpad->priv->converter);
+    aaggcpad->priv->converter = NULL;
+  }
+
+  if (gst_audio_info_is_equal (in_info, out_info) ||
+      in_info->finfo->format == GST_AUDIO_FORMAT_UNKNOWN) {
+    if (aaggcpad->priv->converter) {
+      gst_audio_converter_free (aaggcpad->priv->converter);
+      aaggcpad->priv->converter = NULL;
+    }
+  } else {
+    /* If we haven't received caps yet, this pad should not have
+     * a buffer to convert anyway */
+    aaggcpad->priv->converter =
+        gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
+        in_info, out_info,
+        aaggcpad->priv->converter_config ? gst_structure_copy (aaggcpad->
+            priv->converter_config) : NULL);
+  }
+
+  aaggcpad->priv->converter_config_changed = FALSE;
+}
+
+static GstBuffer *
+gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorConvertPad *
+    aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
+    GstBuffer * input_buffer)
+{
+  GstBuffer *res;
+
+  gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info,
+      out_info);
+
+  if (aaggcpad->priv->converter) {
+    gint insize = gst_buffer_get_size (input_buffer);
+    gsize insamples = insize / in_info->bpf;
+    gsize outsamples =
+        gst_audio_converter_get_out_frames (aaggcpad->priv->converter,
+        insamples);
+    gint outsize = outsamples * out_info->bpf;
+    GstMapInfo inmap, outmap;
+
+    res = gst_buffer_new_allocate (NULL, outsize, NULL);
+
+    /* We create a perfectly similar buffer, except obviously for
+     * its converted contents */
+    gst_buffer_copy_into (res, input_buffer,
+        GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
+        GST_BUFFER_COPY_META, 0, -1);
+
+    gst_buffer_map (input_buffer, &inmap, GST_MAP_READ);
+    gst_buffer_map (res, &outmap, GST_MAP_WRITE);
+
+    gst_audio_converter_samples (aaggcpad->priv->converter,
+        GST_AUDIO_CONVERTER_FLAG_NONE,
+        (gpointer *) & inmap.data, insamples,
+        (gpointer *) & outmap.data, outsamples);
+
+    gst_buffer_unmap (input_buffer, &inmap);
+    gst_buffer_unmap (res, &outmap);
+  } else {
+    res = gst_buffer_ref (input_buffer);
+  }
+
+  return res;
+}
+
+static void
+gst_audio_aggregator_convert_pad_finalize (GObject * object)
+{
+  GstAudioAggregatorConvertPad *pad = (GstAudioAggregatorConvertPad *) object;
+
+  if (pad->priv->converter)
+    gst_audio_converter_free (pad->priv->converter);
+
+  if (pad->priv->converter_config)
+    gst_structure_free (pad->priv->converter_config);
+
+  G_OBJECT_CLASS (gst_audio_aggregator_convert_pad_parent_class)->finalize
+      (object);
+}
+
+static void
+gst_audio_aggregator_convert_pad_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
+
+  switch (prop_id) {
+    case PROP_PAD_CONVERTER_CONFIG:
+      GST_OBJECT_LOCK (pad);
+      if (pad->priv->converter_config)
+        g_value_set_boxed (value, pad->priv->converter_config);
+      GST_OBJECT_UNLOCK (pad);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_aggregator_convert_pad_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
+
+  switch (prop_id) {
+    case PROP_PAD_CONVERTER_CONFIG:
+      GST_OBJECT_LOCK (pad);
+      if (pad->priv->converter_config)
+        gst_structure_free (pad->priv->converter_config);
+      pad->priv->converter_config = g_value_dup_boxed (value);
+      pad->priv->converter_config_changed = TRUE;
+      GST_OBJECT_UNLOCK (pad);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
+    klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  g_type_class_add_private (klass,
+      sizeof (GstAudioAggregatorConvertPadPrivate));
+
+  gobject_class->set_property = gst_audio_aggregator_convert_pad_set_property;
+  gobject_class->get_property = gst_audio_aggregator_convert_pad_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_PAD_CONVERTER_CONFIG,
+      g_param_spec_boxed ("converter-config", "Converter configuration",
+          "A GstStructure describing the configuration that should be used "
+          "when converting this pad's audio buffers",
+          GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize;
+}
+
+static void
+gst_audio_aggregator_convert_pad_init (GstAudioAggregatorConvertPad * pad)
+{
+  pad->priv =
+      G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD,
+      GstAudioAggregatorConvertPadPrivate);
+}
+
+/**************************************
+ * GstAudioAggregator implementation  *
+ **************************************/
+
+struct _GstAudioAggregatorPrivate
+{
+  GMutex mutex;
+
+  /* All three properties are unprotected, can't be modified while streaming */
+  /* Size in frames that is output per buffer */
+  GstClockTime output_buffer_duration;
+  GstClockTime alignment_threshold;
+  GstClockTime discont_wait;
+
+  /* Protected by srcpad stream clock */
+  /* Output buffer starting at offset containing blocksize frames (calculated
+   * from output_buffer_duration) */
+  GstBuffer *current_buffer;
+
+  /* counters to keep track of timestamps */
+  /* Readable with object lock, writable with both aag lock and object lock */
+
+  /* Sample offset starting from 0 at aggregator.segment.start */
+  gint64 offset;
+};
+
+#define GST_AUDIO_AGGREGATOR_LOCK(self)   g_mutex_lock (&(self)->priv->mutex);
+#define GST_AUDIO_AGGREGATOR_UNLOCK(self) g_mutex_unlock (&(self)->priv->mutex);
+
+static void gst_audio_aggregator_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_audio_aggregator_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_audio_aggregator_dispose (GObject * object);
+
+static gboolean gst_audio_aggregator_src_event (GstAggregator * agg,
+    GstEvent * event);
+static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg,
+    GstAggregatorPad * aggpad, GstEvent * event);
+static gboolean gst_audio_aggregator_src_query (GstAggregator * agg,
+    GstQuery * query);
+static gboolean
+gst_audio_aggregator_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
+    GstQuery * query);
+static gboolean gst_audio_aggregator_start (GstAggregator * agg);
+static gboolean gst_audio_aggregator_stop (GstAggregator * agg);
+static GstFlowReturn gst_audio_aggregator_flush (GstAggregator * agg);
+
+static GstBuffer *gst_audio_aggregator_create_output_buffer (GstAudioAggregator
+    * aagg, guint num_frames);
+static GstBuffer *gst_audio_aggregator_do_clip (GstAggregator * agg,
+    GstAggregatorPad * bpad, GstBuffer * buffer);
+static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg,
+    gboolean timeout);
+static gboolean sync_pad_values (GstElement * aagg, GstPad * pad, gpointer ud);
+static gboolean gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg,
+    GstCaps * caps);
+static GstFlowReturn
+gst_audio_aggregator_update_src_caps (GstAggregator * agg,
+    GstCaps * caps, GstCaps ** ret);
+static GstCaps *gst_audio_aggregator_fixate_src_caps (GstAggregator * agg,
+    GstCaps * caps);
+
+#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
+#define DEFAULT_ALIGNMENT_THRESHOLD   (40 * GST_MSECOND)
+#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
+
+enum
+{
+  PROP_0,
+  PROP_OUTPUT_BUFFER_DURATION,
+  PROP_ALIGNMENT_THRESHOLD,
+  PROP_DISCONT_WAIT,
+};
+
+G_DEFINE_ABSTRACT_TYPE (GstAudioAggregator, gst_audio_aggregator,
+    GST_TYPE_AGGREGATOR);
+
+static GstClockTime
+gst_audio_aggregator_get_next_time (GstAggregator * agg)
+{
+  GstClockTime next_time;
+
+  GST_OBJECT_LOCK (agg);
+  if (agg->segment.position == -1 || agg->segment.position < agg->segment.start)
+    next_time = agg->segment.start;
+  else
+    next_time = agg->segment.position;
+
+  if (agg->segment.stop != -1 && next_time > agg->segment.stop)
+    next_time = agg->segment.stop;
+
+  next_time =
+      gst_segment_to_running_time (&agg->segment, GST_FORMAT_TIME, next_time);
+  GST_OBJECT_UNLOCK (agg);
+
+  return next_time;
+}
+
+static GstBuffer *
+gst_audio_aggregator_convert_once (GstAudioAggregator * aagg, GstPad * pad,
+    GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
+{
+  GstAudioConverter *converter =
+      gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
+      in_info, out_info, NULL);
+  gint insize = gst_buffer_get_size (buffer);
+  gsize insamples = insize / in_info->bpf;
+  gsize outsamples = gst_audio_converter_get_out_frames (converter,
+      insamples);
+  gint outsize = outsamples * out_info->bpf;
+  GstMapInfo inmap, outmap;
+  GstBuffer *converted = gst_buffer_new_allocate (NULL, outsize, NULL);
+
+  gst_buffer_copy_into (converted, buffer,
+      GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
+      GST_BUFFER_COPY_META, 0, -1);
+
+  gst_buffer_map (buffer, &inmap, GST_MAP_READ);
+  gst_buffer_map (converted, &outmap, GST_MAP_WRITE);
+
+  gst_audio_converter_samples (converter,
+      GST_AUDIO_CONVERTER_FLAG_NONE,
+      (gpointer *) & inmap.data, insamples,
+      (gpointer *) & outmap.data, outsamples);
+
+  gst_buffer_unmap (buffer, &inmap);
+  gst_buffer_unmap (converted, &outmap);
+  gst_audio_converter_free (converter);
+
+  return converted;
+}
+
+static GstBuffer *
+gst_audio_aggregator_default_convert_buffer (GstAudioAggregator * aagg,
+    GstPad * pad, GstAudioInfo * in_info, GstAudioInfo * out_info,
+    GstBuffer * buffer)
+{
+  if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
+    return
+        gst_audio_aggregator_convert_pad_convert_buffer
+        (GST_AUDIO_AGGREGATOR_CONVERT_PAD (pad),
+        &GST_AUDIO_AGGREGATOR_PAD (pad)->info, out_info, buffer);
+  else
+    return gst_audio_aggregator_convert_once (aagg, pad, in_info, out_info,
+        buffer);
+}
+
+static GstBuffer *
+gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
+    GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
+{
+  GstAudioAggregatorClass *klass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
+
+  g_assert (klass->convert_buffer);
+
+  return klass->convert_buffer (aagg, pad, in_info, out_info, buffer);
+}
+
+static void
+gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass;
+
+  g_type_class_add_private (klass, sizeof (GstAudioAggregatorPrivate));
+
+  gobject_class->set_property = gst_audio_aggregator_set_property;
+  gobject_class->get_property = gst_audio_aggregator_get_property;
+  gobject_class->dispose = gst_audio_aggregator_dispose;
+
+  gstaggregator_class->src_event =
+      GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_event);
+  gstaggregator_class->sink_event =
+      GST_DEBUG_FUNCPTR (gst_audio_aggregator_sink_event);
+  gstaggregator_class->src_query =
+      GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_query);
+  gstaggregator_class->sink_query = gst_audio_aggregator_sink_query;
+  gstaggregator_class->start = gst_audio_aggregator_start;
+  gstaggregator_class->stop = gst_audio_aggregator_stop;
+  gstaggregator_class->flush = gst_audio_aggregator_flush;
+  gstaggregator_class->aggregate =
+      GST_DEBUG_FUNCPTR (gst_audio_aggregator_aggregate);
+  gstaggregator_class->clip = GST_DEBUG_FUNCPTR (gst_audio_aggregator_do_clip);
+  gstaggregator_class->get_next_time = gst_audio_aggregator_get_next_time;
+  gstaggregator_class->update_src_caps =
+      GST_DEBUG_FUNCPTR (gst_audio_aggregator_update_src_caps);
+  gstaggregator_class->fixate_src_caps = gst_audio_aggregator_fixate_src_caps;
+  gstaggregator_class->negotiated_src_caps =
+      gst_audio_aggregator_negotiated_src_caps;
+
+  klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;
+  klass->convert_buffer = gst_audio_aggregator_default_convert_buffer;
+
+  GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
+      GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");
+
+  g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
+      g_param_spec_uint64 ("output-buffer-duration", "Output Buffer Duration",
+          "Output block size in nanoseconds", 1,
+          G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
+      g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
+          "Timestamp alignment threshold in nanoseconds", 0,
+          G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
+      g_param_spec_uint64 ("discont-wait", "Discont Wait",
+          "Window of time in nanoseconds to wait before "
+          "creating a discontinuity", 0,
+          G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_audio_aggregator_init (GstAudioAggregator * aagg)
+{
+  aagg->priv =
+      G_TYPE_INSTANCE_GET_PRIVATE (aagg, GST_TYPE_AUDIO_AGGREGATOR,
+      GstAudioAggregatorPrivate);
+
+  g_mutex_init (&aagg->priv->mutex);
+
+  aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
+  aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
+  aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;
+
+  aagg->current_caps = NULL;
+  gst_audio_info_init (&aagg->info);
+
+  gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
+      aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration);
+}
+
+static void
+gst_audio_aggregator_dispose (GObject * object)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
+
+  gst_caps_replace (&aagg->current_caps, NULL);
+
+  g_mutex_clear (&aagg->priv->mutex);
+
+  G_OBJECT_CLASS (gst_audio_aggregator_parent_class)->dispose (object);
+}
+
+static void
+gst_audio_aggregator_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
+
+  switch (prop_id) {
+    case PROP_OUTPUT_BUFFER_DURATION:
+      aagg->priv->output_buffer_duration = g_value_get_uint64 (value);
+      gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
+          aagg->priv->output_buffer_duration,
+          aagg->priv->output_buffer_duration);
+      break;
+    case PROP_ALIGNMENT_THRESHOLD:
+      aagg->priv->alignment_threshold = g_value_get_uint64 (value);
+      break;
+    case PROP_DISCONT_WAIT:
+      aagg->priv->discont_wait = g_value_get_uint64 (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_aggregator_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
+
+  switch (prop_id) {
+    case PROP_OUTPUT_BUFFER_DURATION:
+      g_value_set_uint64 (value, aagg->priv->output_buffer_duration);
+      break;
+    case PROP_ALIGNMENT_THRESHOLD:
+      g_value_set_uint64 (value, aagg->priv->alignment_threshold);
+      break;
+    case PROP_DISCONT_WAIT:
+      g_value_set_uint64 (value, aagg->priv->discont_wait);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+/* Caps negotiation */
+
+/* Unref after usage */
+static GstAudioAggregatorPad *
+gst_audio_aggregator_get_first_configured_pad (GstAggregator * agg)
+{
+  GstAudioAggregatorPad *res = NULL;
+  GList *l;
+
+  GST_OBJECT_LOCK (agg);
+  for (l = GST_ELEMENT (agg)->sinkpads; l; l = l->next) {
+    GstAudioAggregatorPad *aaggpad = l->data;
+
+    if (GST_AUDIO_INFO_FORMAT (&aaggpad->info) != GST_AUDIO_FORMAT_UNKNOWN) {
+      res = gst_object_ref (aaggpad);
+      break;
+    }
+  }
+  GST_OBJECT_UNLOCK (agg);
+
+  return res;
+}
+
+static GstCaps *
+gst_audio_aggregator_sink_getcaps (GstPad * pad, GstAggregator * agg,
+    GstCaps * filter)
+{
+  GstAudioAggregatorPad *first_configured_pad =
+      gst_audio_aggregator_get_first_configured_pad (agg);
+  GstCaps *sink_template_caps = gst_pad_get_pad_template_caps (pad);
+  GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
+  GstCaps *sink_caps;
+  GstStructure *s, *s2;
+  gint downstream_rate;
+
+  sink_template_caps = gst_caps_make_writable (sink_template_caps);
+  s = gst_caps_get_structure (sink_template_caps, 0);
+
+  if (downstream_caps && !gst_caps_is_empty (downstream_caps))
+    s2 = gst_caps_get_structure (downstream_caps, 0);
+  else
+    s2 = NULL;
+
+  if (s2 && gst_structure_get_int (s2, "rate", &downstream_rate)) {
+    gst_structure_fixate_field_nearest_int (s, "rate", downstream_rate);
+  } else if (first_configured_pad) {
+    gst_structure_fixate_field_nearest_int (s, "rate",
+        first_configured_pad->info.rate);
+  }
+
+  if (first_configured_pad)
+    gst_object_unref (first_configured_pad);
+
+  sink_caps = filter ? gst_caps_intersect (sink_template_caps,
+      filter) : gst_caps_ref (sink_template_caps);
+
+  GST_INFO_OBJECT (pad, "Getting caps with filter %" GST_PTR_FORMAT, filter);
+  GST_DEBUG_OBJECT (pad, "sink template caps : %" GST_PTR_FORMAT,
+      sink_template_caps);
+  GST_DEBUG_OBJECT (pad, "downstream caps %" GST_PTR_FORMAT, downstream_caps);
+  GST_INFO_OBJECT (pad, "returned sink caps : %" GST_PTR_FORMAT, sink_caps);
+
+  gst_caps_unref (sink_template_caps);
+
+  if (downstream_caps)
+    gst_caps_unref (downstream_caps);
+
+  return sink_caps;
+}
+
+static gboolean
+gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
+    GstAggregator * agg, GstCaps * caps)
+{
+  GstAudioAggregatorPad *first_configured_pad =
+      gst_audio_aggregator_get_first_configured_pad (agg);
+  GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
+  GstAudioInfo info;
+  gboolean ret = TRUE;
+  gint downstream_rate;
+  GstStructure *s;
+
+  if (!downstream_caps || gst_caps_is_empty (downstream_caps)) {
+    ret = FALSE;
+    goto done;
+  }
+
+  gst_audio_info_from_caps (&info, caps);
+  s = gst_caps_get_structure (downstream_caps, 0);
+
+  /* TODO: handle different rates on sinkpads, a bit complex
+   * because offsets will have to be updated, and audio resampling
+   * has a latency to take into account
+   */
+  if ((gst_structure_get_int (s, "rate", &downstream_rate)
+          && info.rate != downstream_rate) || (first_configured_pad
+          && info.rate != first_configured_pad->info.rate)) {
+    gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
+    ret = FALSE;
+  } else {
+    GST_OBJECT_LOCK (aaggpad);
+    gst_audio_info_from_caps (&aaggpad->info, caps);
+    if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
+      GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
+          priv->converter_config_changed = TRUE;
+    GST_OBJECT_UNLOCK (aaggpad);
+  }
+
+done:
+  if (first_configured_pad)
+    gst_object_unref (first_configured_pad);
+
+  if (downstream_caps)
+    gst_caps_unref (downstream_caps);
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_audio_aggregator_update_src_caps (GstAggregator * agg,
+    GstCaps * caps, GstCaps ** ret)
+{
+  GstCaps *src_template_caps = gst_pad_get_pad_template_caps (agg->srcpad);
+  GstCaps *downstream_caps =
+      gst_pad_peer_query_caps (agg->srcpad, src_template_caps);
+
+  gst_caps_unref (src_template_caps);
+
+  *ret = gst_caps_intersect (caps, downstream_caps);
+
+  GST_INFO ("Updated src caps to %" GST_PTR_FORMAT, *ret);
+
+  if (downstream_caps)
+    gst_caps_unref (downstream_caps);
+
+  return GST_FLOW_OK;
+}
+
+/* At that point if the caps are not fixed, this means downstream
+ * didn't have fully specified requirements, we'll just go ahead
+ * and fixate raw audio fields using our first configured pad, we don't for
+ * now need a more complicated heuristic
+ */
+static GstCaps *
+gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps)
+{
+  GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
+  GstAudioAggregatorPad *first_configured_pad;
+
+  if (!aaggclass->convert_buffer)
+    return
+        GST_AGGREGATOR_CLASS
+        (gst_audio_aggregator_parent_class)->fixate_src_caps (agg, caps);
+
+  first_configured_pad = gst_audio_aggregator_get_first_configured_pad (agg);
+
+  if (first_configured_pad) {
+    GstStructure *s, *s2;
+    GstCaps *first_configured_caps =
+        gst_audio_info_to_caps (&first_configured_pad->info);
+    gint first_configured_rate, first_configured_channels;
+
+    caps = gst_caps_make_writable (caps);
+    s = gst_caps_get_structure (caps, 0);
+    s2 = gst_caps_get_structure (first_configured_caps, 0);
+
+    gst_structure_get_int (s2, "rate", &first_configured_rate);
+    gst_structure_get_int (s2, "channels", &first_configured_channels);
+
+    gst_structure_fixate_field_string (s, "format",
+        gst_structure_get_string (s2, "format"));
+    gst_structure_fixate_field_string (s, "layout",
+        gst_structure_get_string (s2, "layout"));
+    gst_structure_fixate_field_nearest_int (s, "rate", first_configured_rate);
+    gst_structure_fixate_field_nearest_int (s, "channels",
+        first_configured_channels);
+
+    gst_caps_unref (first_configured_caps);
+    gst_object_unref (first_configured_pad);
+  }
+
+  if (!gst_caps_is_fixed (caps))
+    caps = gst_caps_fixate (caps);
+
+  GST_INFO_OBJECT (agg, "Fixated src caps to %" GST_PTR_FORMAT, caps);
+
+  return caps;
+}
+
+/* Must be called with OBJECT_LOCK taken */
+static void
+gst_audio_aggregator_update_converters (GstAudioAggregator * aagg,
+    GstAudioInfo * new_info)
+{
+  GList *l;
+
+  for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
+    GstAudioAggregatorPad *aaggpad = l->data;
+
+    if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
+      GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
+          priv->converter_config_changed = TRUE;
+
+    /* If we currently were mixing a buffer, we need to convert it to the new
+     * format */
+    if (aaggpad->priv->buffer) {
+      GstBuffer *new_converted_buffer =
+          gst_audio_aggregator_convert_buffer (aagg, GST_PAD (aaggpad),
+          &aaggpad->info, new_info, aaggpad->priv->input_buffer);
+      gst_buffer_replace (&aaggpad->priv->buffer, new_converted_buffer);
+    }
+  }
+}
+
+/* We now have our final output caps, we can create the required converters */
+static gboolean
+gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
+  GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
+  GstAudioInfo info;
+
+  GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);
+
+  if (!gst_audio_info_from_caps (&info, caps)) {
+    GST_WARNING_OBJECT (aagg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
+    return FALSE;
+  }
+
+  GST_AUDIO_AGGREGATOR_LOCK (aagg);
+  GST_OBJECT_LOCK (aagg);
+
+  if (aaggclass->convert_buffer) {
+    gst_audio_aggregator_update_converters (aagg, &info);
+
+    if (aagg->priv->current_buffer
+        && !gst_audio_info_is_equal (&aagg->info, &info)) {
+      GstBuffer *converted =
+          gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info,
+          &info, aagg->priv->current_buffer);
+      gst_buffer_unref (aagg->priv->current_buffer);
+      aagg->priv->current_buffer = converted;
+    }
+  }
+
+  if (!gst_audio_info_is_equal (&info, &aagg->info)) {
+    GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
+    gst_caps_replace (&aagg->current_caps, caps);
+
+    memcpy (&aagg->info, &info, sizeof (info));
+  }
+
+  GST_OBJECT_UNLOCK (aagg);
+  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+
+  return
+      GST_AGGREGATOR_CLASS
+      (gst_audio_aggregator_parent_class)->negotiated_src_caps (agg, caps);
+}
+
+/* event handling */
+
+static gboolean
+gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event)
+{
+  gboolean result;
+
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
+  GST_DEBUG_OBJECT (agg->srcpad, "Got %s event on src pad",
+      GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_QOS:
+      /* QoS might be tricky */
+      gst_event_unref (event);
+      return FALSE;
+    case GST_EVENT_NAVIGATION:
+      /* navigation is rather pointless. */
+      gst_event_unref (event);
+      return FALSE;
+      break;
+    case GST_EVENT_SEEK:
+    {
+      GstSeekFlags flags;
+      gdouble rate;
+      GstSeekType start_type, stop_type;
+      gint64 start, stop;
+      GstFormat seek_format, dest_format;
+
+      /* parse the seek parameters */
+      gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
+          &start, &stop_type, &stop);
+
+      /* Check the seeking parameters before linking up */
+      if ((start_type != GST_SEEK_TYPE_NONE)
+          && (start_type != GST_SEEK_TYPE_SET)) {
+        result = FALSE;
+        GST_DEBUG_OBJECT (aagg,
+            "seeking failed, unhandled seek type for start: %d", start_type);
+        goto done;
+      }
+      if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) {
+        result = FALSE;
+        GST_DEBUG_OBJECT (aagg,
+            "seeking failed, unhandled seek type for end: %d", stop_type);
+        goto done;
+      }
+
+      GST_OBJECT_LOCK (agg);
+      dest_format = agg->segment.format;
+      GST_OBJECT_UNLOCK (agg);
+      if (seek_format != dest_format) {
+        result = FALSE;
+        GST_DEBUG_OBJECT (aagg,
+            "seeking failed, unhandled seek format: %s",
+            gst_format_get_name (seek_format));
+        goto done;
+      }
+    }
+      break;
+    default:
+      break;
+  }
+
+  return
+      GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_event (agg,
+      event);
+
+done:
+  return result;
+}
+
+
+static gboolean
+gst_audio_aggregator_sink_event (GstAggregator * agg,
+    GstAggregatorPad * aggpad, GstEvent * event)
+{
+  GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
+  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_SEGMENT:
+    {
+      const GstSegment *segment;
+      gst_event_parse_segment (event, &segment);
+
+      if (segment->format != GST_FORMAT_TIME) {
+        GST_ERROR_OBJECT (agg, "Segment of type %s are not supported,"
+            " only TIME segments are supported",
+            gst_format_get_name (segment->format));
+        gst_event_unref (event);
+        event = NULL;
+        res = FALSE;
+        break;
+      }
+
+      GST_OBJECT_LOCK (agg);
+      if (segment->rate != agg->segment.rate) {
+        GST_ERROR_OBJECT (aggpad,
+            "Got segment event with wrong rate %lf, expected %lf",
+            segment->rate, agg->segment.rate);
+        res = FALSE;
+        gst_event_unref (event);
+        event = NULL;
+      } else if (segment->rate < 0.0) {
+        GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet");
+        res = FALSE;
+        gst_event_unref (event);
+        event = NULL;
+      } else {
+        GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
+
+        GST_OBJECT_LOCK (pad);
+        pad->priv->new_segment = TRUE;
+        GST_OBJECT_UNLOCK (pad);
+      }
+      GST_OBJECT_UNLOCK (agg);
+
+      break;
+    }
+    case GST_EVENT_CAPS:
+    {
+      GstCaps *caps;
+
+      gst_event_parse_caps (event, &caps);
+      GST_INFO_OBJECT (aggpad, "Got caps %" GST_PTR_FORMAT, caps);
+      res = gst_audio_aggregator_sink_setcaps (aaggpad, agg, caps);
+      gst_event_unref (event);
+      event = NULL;
+      break;
+    }
+    default:
+      break;
+  }
+
+  if (event != NULL)
+    return
+        GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event
+        (agg, aggpad, event);
+
+  return res;
+}
+
+static gboolean
+gst_audio_aggregator_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_aggregator_sink_getcaps (GST_PAD (aggpad), agg, filter);
+      gst_query_set_caps_result (query, caps);
+      gst_caps_unref (caps);
+      res = TRUE;
+      break;
+    }
+    default:
+      res =
+          GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_query
+          (agg, aggpad, query);
+      break;
+  }
+
+  return res;
+}
+
+
+/* FIXME, the duration query should reflect how long you will produce
+ * data, that is the amount of stream time until you will emit EOS.
+ *
+ * For synchronized mixing this is always the max of all the durations
+ * of upstream since we emit EOS when all of them finished.
+ *
+ * We don't do synchronized mixing so this really depends on where the
+ * streams where punched in and what their relative offsets are against
+ * eachother which we can get from the first timestamps we see.
+ *
+ * When we add a new stream (or remove a stream) the duration might
+ * also become invalid again and we need to post a new DURATION
+ * message to notify this fact to the parent.
+ * For now we take the max of all the upstream elements so the simple
+ * cases work at least somewhat.
+ */
+static gboolean
+gst_audio_aggregator_query_duration (GstAudioAggregator * aagg,
+    GstQuery * query)
+{
+  gint64 max;
+  gboolean res;
+  GstFormat format;
+  GstIterator *it;
+  gboolean done;
+  GValue item = { 0, };
+
+  /* parse format */
+  gst_query_parse_duration (query, &format, NULL);
+
+  max = -1;
+  res = TRUE;
+  done = FALSE;
+
+  it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg));
+  while (!done) {
+    GstIteratorResult ires;
+
+    ires = gst_iterator_next (it, &item);
+    switch (ires) {
+      case GST_ITERATOR_DONE:
+        done = TRUE;
+        break;
+      case GST_ITERATOR_OK:
+      {
+        GstPad *pad = g_value_get_object (&item);
+        gint64 duration;
+
+        /* ask sink peer for duration */
+        res &= gst_pad_peer_query_duration (pad, format, &duration);
+        /* take max from all valid return values */
+        if (res) {
+          /* valid unknown length, stop searching */
+          if (duration == -1) {
+            max = duration;
+            done = TRUE;
+          }
+          /* else see if bigger than current max */
+          else if (duration > max)
+            max = duration;
+        }
+        g_value_reset (&item);
+        break;
+      }
+      case GST_ITERATOR_RESYNC:
+        max = -1;
+        res = TRUE;
+        gst_iterator_resync (it);
+        break;
+      default:
+        res = FALSE;
+        done = TRUE;
+        break;
+    }
+  }
+  g_value_unset (&item);
+  gst_iterator_free (it);
+
+  if (res) {
+    /* and store the max */
+    GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %"
+        GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
+    gst_query_set_duration (query, format, max);
+  }
+
+  return res;
+}
+
+
+static gboolean
+gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
+  gboolean res = FALSE;
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_DURATION:
+      res = gst_audio_aggregator_query_duration (aagg, query);
+      break;
+    case GST_QUERY_POSITION:
+    {
+      GstFormat format;
+
+      gst_query_parse_position (query, &format, NULL);
+
+      GST_OBJECT_LOCK (aagg);
+
+      switch (format) {
+        case GST_FORMAT_TIME:
+          gst_query_set_position (query, format,
+              gst_segment_to_stream_time (&agg->segment, GST_FORMAT_TIME,
+                  agg->segment.position));
+          res = TRUE;
+          break;
+        case GST_FORMAT_BYTES:
+          if (GST_AUDIO_INFO_BPF (&aagg->info)) {
+            gst_query_set_position (query, format, aagg->priv->offset *
+                GST_AUDIO_INFO_BPF (&aagg->info));
+            res = TRUE;
+          }
+          break;
+        case GST_FORMAT_DEFAULT:
+          gst_query_set_position (query, format, aagg->priv->offset);
+          res = TRUE;
+          break;
+        default:
+          break;
+      }
+
+      GST_OBJECT_UNLOCK (aagg);
+
+      break;
+    }
+    default:
+      res =
+          GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query
+          (agg, query);
+      break;
+  }
+
+  return res;
+}
+
+
+void
+gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg,
+    GstAudioAggregatorPad * pad, GstCaps * caps)
+{
+#ifndef G_DISABLE_ASSERT
+  gboolean valid;
+
+  GST_OBJECT_LOCK (pad);
+  valid = gst_audio_info_from_caps (&pad->info, caps);
+  g_assert (valid);
+  GST_OBJECT_UNLOCK (pad);
+#else
+  GST_OBJECT_LOCK (pad);
+  (void) gst_audio_info_from_caps (&pad->info, caps);
+  GST_OBJECT_UNLOCK (pad);
+#endif
+}
+
+/* Must hold object lock and aagg lock to call */
+
+static void
+gst_audio_aggregator_reset (GstAudioAggregator * aagg)
+{
+  GstAggregator *agg = GST_AGGREGATOR (aagg);
+
+  GST_AUDIO_AGGREGATOR_LOCK (aagg);
+  GST_OBJECT_LOCK (aagg);
+  agg->segment.position = -1;
+  aagg->priv->offset = -1;
+  gst_audio_info_init (&aagg->info);
+  gst_caps_replace (&aagg->current_caps, NULL);
+  gst_buffer_replace (&aagg->priv->current_buffer, NULL);
+  GST_OBJECT_UNLOCK (aagg);
+  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+}
+
+static gboolean
+gst_audio_aggregator_start (GstAggregator * agg)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
+
+  gst_audio_aggregator_reset (aagg);
+
+  return TRUE;
+}
+
+static gboolean
+gst_audio_aggregator_stop (GstAggregator * agg)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
+
+  gst_audio_aggregator_reset (aagg);
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_audio_aggregator_flush (GstAggregator * agg)
+{
+  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
+
+  GST_AUDIO_AGGREGATOR_LOCK (aagg);
+  GST_OBJECT_LOCK (aagg);
+  agg->segment.position = -1;
+  aagg->priv->offset = -1;
+  gst_buffer_replace (&aagg->priv->current_buffer, NULL);
+  GST_OBJECT_UNLOCK (aagg);
+  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+
+  return GST_FLOW_OK;
+}
+
+static GstBuffer *
+gst_audio_aggregator_do_clip (GstAggregator * agg,
+    GstAggregatorPad * bpad, GstBuffer * buffer)
+{
+  GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
+  gint rate, bpf;
+
+  rate = GST_AUDIO_INFO_RATE (&pad->info);
+  bpf = GST_AUDIO_INFO_BPF (&pad->info);
+
+  GST_OBJECT_LOCK (bpad);
+  buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
+  GST_OBJECT_UNLOCK (bpad);
+
+  return buffer;
+}
+
+/* Called with the object lock for both the element and pad held,
+ * as well as the aagg lock
+ *
+ * Replace the current buffer with input and update GstAudioAggregatorPadPrivate
+ * values.
+ */
+static gboolean
+gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
+    GstAudioAggregatorPad * pad)
+{
+  GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
+  GstClockTime start_time, end_time;
+  gboolean discont = FALSE;
+  guint64 start_offset, end_offset;
+  gint rate, bpf;
+
+  GstAggregator *agg = GST_AGGREGATOR (aagg);
+  GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
+
+  if (aaggclass->convert_buffer) {
+    rate = GST_AUDIO_INFO_RATE (&aagg->info);
+    bpf = GST_AUDIO_INFO_BPF (&aagg->info);
+  } else {
+    rate = GST_AUDIO_INFO_RATE (&pad->info);
+    bpf = GST_AUDIO_INFO_BPF (&pad->info);
+  }
+
+  pad->priv->position = 0;
+  pad->priv->size = gst_buffer_get_size (pad->priv->buffer) / bpf;
+
+  if (pad->priv->size == 0) {
+    if (!GST_BUFFER_DURATION_IS_VALID (pad->priv->buffer) ||
+        !GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_GAP)) {
+      GST_WARNING_OBJECT (pad, "Dropping 0-sized buffer missing either a"
+          " duration or a GAP flag: %" GST_PTR_FORMAT, pad->priv->buffer);
+      return FALSE;
+    }
+
+    pad->priv->size =
+        gst_util_uint64_scale (GST_BUFFER_DURATION (pad->priv->buffer), rate,
+        GST_SECOND);
+  }
+
+  if (!GST_BUFFER_PTS_IS_VALID (pad->priv->buffer)) {
+    if (pad->priv->output_offset == -1)
+      pad->priv->output_offset = aagg->priv->offset;
+    if (pad->priv->next_offset == -1)
+      pad->priv->next_offset = pad->priv->size;
+    else
+      pad->priv->next_offset += pad->priv->size;
+    goto done;
+  }
+
+  start_time = GST_BUFFER_PTS (pad->priv->buffer);
+  end_time =
+      start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
+      rate);
+
+  /* Clipping should've ensured this */
+  g_assert (start_time >= aggpad->segment.start);
+
+  start_offset =
+      gst_util_uint64_scale (start_time - aggpad->segment.start, rate,
+      GST_SECOND);
+  end_offset = start_offset + pad->priv->size;
+
+  if (GST_BUFFER_IS_DISCONT (pad->priv->buffer)
+      || GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_RESYNC)
+      || pad->priv->new_segment || pad->priv->next_offset == -1) {
+    discont = TRUE;
+    pad->priv->new_segment = FALSE;
+  } else {
+    guint64 diff, max_sample_diff;
+
+    /* Check discont, based on audiobasesink */
+    if (start_offset <= pad->priv->next_offset)
+      diff = pad->priv->next_offset - start_offset;
+    else
+      diff = start_offset - pad->priv->next_offset;
+
+    max_sample_diff =
+        gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate,
+        GST_SECOND);
+
+    /* Discont! */
+    if (G_UNLIKELY (diff >= max_sample_diff)) {
+      if (aagg->priv->discont_wait > 0) {
+        if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) {
+          pad->priv->discont_time = start_time;
+        } else if (start_time - pad->priv->discont_time >=
+            aagg->priv->discont_wait) {
+          discont = TRUE;
+          pad->priv->discont_time = GST_CLOCK_TIME_NONE;
+        }
+      } else {
+        discont = TRUE;
+      }
+    } else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) {
+      /* we have had a discont, but are now back on track! */
+      pad->priv->discont_time = GST_CLOCK_TIME_NONE;
+    }
+  }
+
+  if (discont) {
+    /* Have discont, need resync */
+    if (pad->priv->next_offset != -1)
+      GST_DEBUG_OBJECT (pad, "Have discont. Expected %"
+          G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
+          pad->priv->next_offset, start_offset);
+    pad->priv->output_offset = -1;
+    pad->priv->next_offset = end_offset;
+  } else {
+    pad->priv->next_offset += pad->priv->size;
+  }
+
+  if (pad->priv->output_offset == -1) {
+    GstClockTime start_running_time;
+    GstClockTime end_running_time;
+    GstClockTime segment_pos;
+    guint64 start_output_offset = -1;
+    guint64 end_output_offset = -1;
+
+    start_running_time =
+        gst_segment_to_running_time (&aggpad->segment,
+        GST_FORMAT_TIME, start_time);
+    end_running_time =
+        gst_segment_to_running_time (&aggpad->segment,
+        GST_FORMAT_TIME, end_time);
+
+    /* Convert to position in the output segment */
+    segment_pos =
+        gst_segment_position_from_running_time (&agg->segment, GST_FORMAT_TIME,
+        start_running_time);
+    if (GST_CLOCK_TIME_IS_VALID (segment_pos))
+      start_output_offset =
+          gst_util_uint64_scale (segment_pos - agg->segment.start, rate,
+          GST_SECOND);
+
+    segment_pos =
+        gst_segment_position_from_running_time (&agg->segment, GST_FORMAT_TIME,
+        end_running_time);
+    if (GST_CLOCK_TIME_IS_VALID (segment_pos))
+      end_output_offset =
+          gst_util_uint64_scale (segment_pos - agg->segment.start, rate,
+          GST_SECOND);
+
+    if (start_output_offset == -1 && end_output_offset == -1) {
+      /* Outside output segment, drop */
+      pad->priv->position = 0;
+      pad->priv->size = 0;
+      pad->priv->output_offset = -1;
+      GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
+      return FALSE;
+    }
+
+    /* Calculate end_output_offset if it was outside the output segment */
+    if (end_output_offset == -1)
+      end_output_offset = start_output_offset + pad->priv->size;
+
+    if (end_output_offset < aagg->priv->offset) {
+      pad->priv->position = 0;
+      pad->priv->size = 0;
+      pad->priv->output_offset = -1;
+      GST_DEBUG_OBJECT (pad,
+          "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
+          G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
+      return FALSE;
+    }
+
+    if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) {
+      guint diff;
+
+      if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
+        diff = pad->priv->size - end_output_offset + aagg->priv->offset;
+      } else if (start_output_offset == -1) {
+        start_output_offset = end_output_offset - pad->priv->size;
+
+        if (start_output_offset < aagg->priv->offset)
+          diff = aagg->priv->offset - start_output_offset;
+        else
+          diff = 0;
+      } else {
+        diff = aagg->priv->offset - start_output_offset;
+      }
+
+      pad->priv->position += diff;
+      if (pad->priv->position >= pad->priv->size) {
+        /* Empty buffer, drop */
+        pad->priv->position = 0;
+        pad->priv->size = 0;
+        pad->priv->output_offset = -1;
+        GST_DEBUG_OBJECT (pad,
+            "Buffer before segment or current position: %" G_GUINT64_FORMAT
+            " < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
+        return FALSE;
+      }
+    }
+
+    if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
+      pad->priv->output_offset = aagg->priv->offset;
+    else
+      pad->priv->output_offset = start_output_offset;
+
+    GST_DEBUG_OBJECT (pad,
+        "Buffer resynced: Pad offset %" G_GUINT64_FORMAT
+        ", current audio aggregator offset %" G_GINT64_FORMAT,
+        pad->priv->output_offset, aagg->priv->offset);
+  }
+
+done:
+
+  GST_LOG_OBJECT (pad,
+      "Queued new buffer at offset %" G_GUINT64_FORMAT,
+      pad->priv->output_offset);
+
+  return TRUE;
+}
+
+/* Called with pad object lock held */
+
+static gboolean
+gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
+    GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf,
+    guint blocksize)
+{
+  guint overlap;
+  guint out_start;
+  gboolean filled;
+  guint in_offset;
+  gboolean pad_changed = FALSE;
+
+  /* Overlap => mix */
+  if (aagg->priv->offset < pad->priv->output_offset)
+    out_start = pad->priv->output_offset - aagg->priv->offset;
+  else
+    out_start = 0;
+
+  overlap = pad->priv->size - pad->priv->position;
+  if (overlap > blocksize - out_start)
+    overlap = blocksize - out_start;
+
+  if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
+    /* skip gap buffer */
+    GST_LOG_OBJECT (pad, "skipping GAP buffer");
+    pad->priv->output_offset += pad->priv->size - pad->priv->position;
+    pad->priv->position = pad->priv->size;
+
+    gst_buffer_replace (&pad->priv->buffer, NULL);
+    gst_buffer_replace (&pad->priv->input_buffer, NULL);
+    return FALSE;
+  }
+
+  gst_buffer_ref (inbuf);
+  in_offset = pad->priv->position;
+  GST_OBJECT_UNLOCK (pad);
+  GST_OBJECT_UNLOCK (aagg);
+
+  filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg,
+      pad, inbuf, in_offset, outbuf, out_start, overlap);
+
+  GST_OBJECT_LOCK (aagg);
+  GST_OBJECT_LOCK (pad);
+
+  pad_changed = (inbuf != pad->priv->buffer);
+  gst_buffer_unref (inbuf);
+
+  if (filled)
+    GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP);
+
+  if (pad_changed)
+    return FALSE;
+
+  pad->priv->position += overlap;
+  pad->priv->output_offset += overlap;
+
+  if (pad->priv->position == pad->priv->size) {
+    /* Buffer done, drop it */
+    gst_buffer_replace (&pad->priv->buffer, NULL);
+    gst_buffer_replace (&pad->priv->input_buffer, NULL);
+    GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next");
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+static GstBuffer *
+gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
+    guint num_frames)
+{
+  GstAllocator *allocator;
+  GstAllocationParams params;
+  GstBuffer *outbuf;
+  GstMapInfo outmap;
+
+  gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, &params);
+
+  GST_DEBUG ("Creating output buffer with size %d",
+      num_frames * GST_AUDIO_INFO_BPF (&aagg->info));
+
+  outbuf = gst_buffer_new_allocate (allocator, num_frames *
+      GST_AUDIO_INFO_BPF (&aagg->info), &params);
+
+  if (allocator)
+    gst_object_unref (allocator);
+
+  gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
+  gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
+  gst_buffer_unmap (outbuf, &outmap);
+
+  return outbuf;
+}
+
+static gboolean
+sync_pad_values (GstElement * aagg, GstPad * pad, gpointer user_data)
+{
+  GstAudioAggregatorPad *aapad = GST_AUDIO_AGGREGATOR_PAD (pad);
+  GstAggregatorPad *bpad = GST_AGGREGATOR_PAD_CAST (pad);
+  GstClockTime timestamp, stream_time;
+
+  if (aapad->priv->buffer == NULL)
+    return TRUE;
+
+  timestamp = GST_BUFFER_PTS (aapad->priv->buffer);
+  GST_OBJECT_LOCK (bpad);
+  stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
+      timestamp);
+  GST_OBJECT_UNLOCK (bpad);
+
+  /* sync object properties on stream time */
+  /* TODO: Ideally we would want to do that on every sample */
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (GST_OBJECT_CAST (pad), stream_time);
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
+{
+  /* Calculate the current output offset/timestamp and offset_end/timestamp_end.
+   * Allocate a silence buffer for this and store it.
+   *
+   * For all pads:
+   * 1) Once per input buffer (cached)
+   *   1) Check discont (flag and timestamp with tolerance)
+   *   2) If discont or new, resync. That means:
+   *     1) Drop all start data of the buffer that comes before
+   *        the current position/offset.
+   *     2) Calculate the offset (output segment!) that the first
+   *        frame of the input buffer corresponds to. Base this on
+   *        the running time.
+   *
+   * 2) If the current pad's offset/offset_end overlaps with the output
+   *    offset/offset_end, mix it at the appropiate position in the output
+   *    buffer and advance the pad's position. Remember if this pad needs
+   *    a new buffer to advance behind the output offset_end.
+   *
+   * If we had no pad with a buffer, go EOS.
+   *
+   * If we had at least one pad that did not advance behind output
+   * offset_end, let aggregate be called again for the current
+   * output offset/offset_end.
+   */
+  GstElement *element;
+  GstAudioAggregator *aagg;
+  GList *iter;
+  GstFlowReturn ret;
+  GstBuffer *outbuf = NULL;
+  gint64 next_offset;
+  gint64 next_timestamp;
+  gint rate, bpf;
+  gboolean dropped = FALSE;
+  gboolean is_eos = TRUE;
+  gboolean is_done = TRUE;
+  guint blocksize;
+
+  element = GST_ELEMENT (agg);
+  aagg = GST_AUDIO_AGGREGATOR (agg);
+
+  /* Sync pad properties to the stream time */
+  gst_element_foreach_sink_pad (element, sync_pad_values, NULL);
+
+  GST_AUDIO_AGGREGATOR_LOCK (aagg);
+  GST_OBJECT_LOCK (agg);
+
+  /* Update position from the segment start/stop if needed */
+  if (agg->segment.position == -1) {
+    if (agg->segment.rate > 0.0)
+      agg->segment.position = agg->segment.start;
+    else
+      agg->segment.position = agg->segment.stop;
+  }
+
+  if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
+    if (timeout) {
+      GST_DEBUG_OBJECT (aagg,
+          "Got timeout before receiving any caps, don't output anything");
+
+      /* Advance position */
+      if (agg->segment.rate > 0.0)
+        agg->segment.position += aagg->priv->output_buffer_duration;
+      else if (agg->segment.position > aagg->priv->output_buffer_duration)
+        agg->segment.position -= aagg->priv->output_buffer_duration;
+      else
+        agg->segment.position = 0;
+
+      GST_OBJECT_UNLOCK (agg);
+      GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+      return GST_AGGREGATOR_FLOW_NEED_DATA;
+    } else {
+      GST_OBJECT_UNLOCK (agg);
+      goto not_negotiated;
+    }
+  }
+
+  rate = GST_AUDIO_INFO_RATE (&aagg->info);
+  bpf = GST_AUDIO_INFO_BPF (&aagg->info);
+
+  if (aagg->priv->offset == -1) {
+    aagg->priv->offset =
+        gst_util_uint64_scale (agg->segment.position - agg->segment.start, rate,
+        GST_SECOND);
+    GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT,
+        aagg->priv->offset);
+  }
+
+  blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
+      rate, GST_SECOND);
+  blocksize = MAX (1, blocksize);
+
+  /* FIXME: Reverse mixing does not work at all yet */
+  if (agg->segment.rate > 0.0) {
+    next_offset = aagg->priv->offset + blocksize;
+  } else {
+    next_offset = aagg->priv->offset - blocksize;
+  }
+
+  /* Use the sample counter, which will never accumulate rounding errors */
+  next_timestamp =
+      agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
+      rate);
+
+  if (aagg->priv->current_buffer == NULL) {
+    GST_OBJECT_UNLOCK (agg);
+    aagg->priv->current_buffer =
+        GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
+        blocksize);
+    /* Be careful, some things could have changed ? */
+    GST_OBJECT_LOCK (agg);
+    GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
+  }
+  outbuf = aagg->priv->current_buffer;
+
+  GST_LOG_OBJECT (agg,
+      "Starting to mix %u samples for offset %" G_GINT64_FORMAT
+      " with timestamp %" GST_TIME_FORMAT, blocksize,
+      aagg->priv->offset, GST_TIME_ARGS (agg->segment.position));
+
+  for (iter = element->sinkpads; iter; iter = iter->next) {
+    GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
+    GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
+    gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);
+
+    if (!pad_eos)
+      is_eos = FALSE;
+
+    pad->priv->input_buffer = gst_aggregator_pad_peek_buffer (aggpad);
+
+    GST_OBJECT_LOCK (pad);
+    if (!pad->priv->input_buffer) {
+      if (timeout) {
+        if (pad->priv->output_offset < next_offset) {
+          gint64 diff = next_offset - pad->priv->output_offset;
+          GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT
+              " frames (%" GST_TIME_FORMAT ")", diff,
+              GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
+                      GST_AUDIO_INFO_RATE (&aagg->info))));
+        }
+      } else if (!pad_eos) {
+        is_done = FALSE;
+      }
+      GST_OBJECT_UNLOCK (pad);
+      continue;
+    }
+
+    /* New buffer? */
+    if (!pad->priv->buffer) {
+      if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
+        pad->priv->buffer =
+            gst_audio_aggregator_convert_buffer
+            (aagg, GST_PAD (pad), &pad->info, &aagg->info,
+            pad->priv->input_buffer);
+      else
+        pad->priv->buffer = gst_buffer_ref (pad->priv->input_buffer);
+
+      if (!gst_audio_aggregator_fill_buffer (aagg, pad)) {
+        gst_buffer_replace (&pad->priv->buffer, NULL);
+        gst_buffer_replace (&pad->priv->input_buffer, NULL);
+        pad->priv->buffer = NULL;
+        dropped = TRUE;
+        GST_OBJECT_UNLOCK (pad);
+
+        gst_aggregator_pad_drop_buffer (aggpad);
+        continue;
+      }
+    } else {
+      gst_buffer_unref (pad->priv->input_buffer);
+    }
+
+    if (!pad->priv->buffer && !dropped && pad_eos) {
+      GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
+      GST_OBJECT_UNLOCK (pad);
+      continue;
+    }
+
+    g_assert (pad->priv->buffer);
+
+    /* This pad is lagging behind, we need to update the offset
+     * and maybe drop the current buffer */
+    if (pad->priv->output_offset < aagg->priv->offset) {
+      gint64 diff = aagg->priv->offset - pad->priv->output_offset;
+      gint64 odiff = diff;
+
+      if (pad->priv->position + diff > pad->priv->size)
+        diff = pad->priv->size - pad->priv->position;
+      pad->priv->position += diff;
+      pad->priv->output_offset += diff;
+
+      if (pad->priv->position == pad->priv->size) {
+        GST_DEBUG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
+            ", dropping %" GST_PTR_FORMAT,
+            GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
+                    GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
+        /* Buffer done, drop it */
+        gst_buffer_replace (&pad->priv->buffer, NULL);
+        gst_buffer_replace (&pad->priv->input_buffer, NULL);
+        dropped = TRUE;
+        GST_OBJECT_UNLOCK (pad);
+        gst_aggregator_pad_drop_buffer (aggpad);
+        continue;
+      }
+    }
+
+    g_assert (pad->priv->buffer);
+
+    if (pad->priv->output_offset >= aagg->priv->offset
+        && pad->priv->output_offset < aagg->priv->offset + blocksize) {
+      gboolean drop_buf;
+
+      GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
+      drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
+          outbuf, blocksize);
+      if (pad->priv->output_offset >= next_offset) {
+        GST_LOG_OBJECT (pad,
+            "Pad is at or after current offset: %" G_GUINT64_FORMAT " >= %"
+            G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
+      } else {
+        is_done = FALSE;
+      }
+      if (drop_buf) {
+        GST_OBJECT_UNLOCK (pad);
+        gst_aggregator_pad_drop_buffer (aggpad);
+        continue;
+      }
+    }
+
+    GST_OBJECT_UNLOCK (pad);
+  }
+  GST_OBJECT_UNLOCK (agg);
+
+  if (dropped) {
+    /* We dropped a buffer, retry */
+    GST_LOG_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
+    GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+    return GST_AGGREGATOR_FLOW_NEED_DATA;
+  }
+
+  if (!is_done && !is_eos) {
+    /* Get more buffers */
+    GST_LOG_OBJECT (aagg,
+        "We're not done yet for the current offset, waiting for more data");
+    GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+    return GST_AGGREGATOR_FLOW_NEED_DATA;
+  }
+
+  if (is_eos) {
+    gint64 max_offset = 0;
+
+    GST_DEBUG_OBJECT (aagg, "We're EOS");
+
+    GST_OBJECT_LOCK (agg);
+    for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
+      GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
+
+      max_offset = MAX ((gint64) max_offset, (gint64) pad->priv->output_offset);
+    }
+    GST_OBJECT_UNLOCK (agg);
+
+    /* This means EOS or nothing mixed in at all */
+    if (aagg->priv->offset == max_offset) {
+      gst_buffer_replace (&aagg->priv->current_buffer, NULL);
+      GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+      return GST_FLOW_EOS;
+    }
+
+    if (max_offset <= next_offset) {
+      GST_DEBUG_OBJECT (aagg,
+          "Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
+          G_GINT64_FORMAT, max_offset, next_offset);
+      next_offset = max_offset;
+      next_timestamp =
+          agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
+          rate);
+
+      if (next_offset > aagg->priv->offset)
+        gst_buffer_resize (outbuf, 0, (next_offset - aagg->priv->offset) * bpf);
+    }
+  }
+
+  /* set timestamps on the output buffer */
+  GST_OBJECT_LOCK (agg);
+  if (agg->segment.rate > 0.0) {
+    GST_BUFFER_PTS (outbuf) = agg->segment.position;
+    GST_BUFFER_OFFSET (outbuf) = aagg->priv->offset;
+    GST_BUFFER_OFFSET_END (outbuf) = next_offset;
+    GST_BUFFER_DURATION (outbuf) = next_timestamp - agg->segment.position;
+  } else {
+    GST_BUFFER_PTS (outbuf) = next_timestamp;
+    GST_BUFFER_OFFSET (outbuf) = next_offset;
+    GST_BUFFER_OFFSET_END (outbuf) = aagg->priv->offset;
+    GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp;
+  }
+
+  GST_OBJECT_UNLOCK (agg);
+
+  /* send it out */
+  GST_LOG_OBJECT (aagg,
+      "pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
+      G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
+      GST_BUFFER_OFFSET (outbuf));
+
+  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+
+  ret = gst_aggregator_finish_buffer (agg, outbuf);
+  aagg->priv->current_buffer = NULL;
+
+  GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret));
+
+  GST_AUDIO_AGGREGATOR_LOCK (aagg);
+  GST_OBJECT_LOCK (agg);
+  aagg->priv->offset = next_offset;
+  agg->segment.position = next_timestamp;
+
+  /* If there was a timeout and there was a gap in data in out of the streams,
+   * then it's a very good time to for a resync with the timestamps.
+   */
+  if (timeout) {
+    for (iter = element->sinkpads; iter; iter = iter->next) {
+      GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
+
+      GST_OBJECT_LOCK (pad);
+      if (pad->priv->output_offset < aagg->priv->offset)
+        pad->priv->output_offset = -1;
+      GST_OBJECT_UNLOCK (pad);
+    }
+  }
+  GST_OBJECT_UNLOCK (agg);
+  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+
+  return ret;
+  /* ERRORS */
+not_negotiated:
+  {
+    GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
+    GST_ELEMENT_ERROR (aagg, STREAM, FORMAT, (NULL),
+        ("Unknown data received, not negotiated"));
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+}
diff --git a/gst-libs/gst/audio/gstaudioaggregator.h b/gst-libs/gst/audio/gstaudioaggregator.h
new file mode 100644
index 000000000..b32630ee6
--- /dev/null
+++ b/gst-libs/gst/audio/gstaudioaggregator.h
@@ -0,0 +1,228 @@
+/* GStreamer
+ * Copyright (C) 2014 Collabora
+ *   Author: Olivier Crete <olivier.crete@collabora.com>
+ *
+ * gstaudioaggregator.h:
+ *
+ * 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_AGGREGATOR_H__
+#define __GST_AUDIO_AGGREGATOR_H__
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "The Base library from gst-plugins-bad is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstaggregator.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+/*******************************
+ * GstAudioAggregator Structs  *
+ *******************************/
+
+typedef struct _GstAudioAggregator GstAudioAggregator;
+typedef struct _GstAudioAggregatorPrivate GstAudioAggregatorPrivate;
+typedef struct _GstAudioAggregatorClass GstAudioAggregatorClass;
+
+
+/************************
+ * GstAudioAggregatorPad API *
+ ***********************/
+
+#define GST_TYPE_AUDIO_AGGREGATOR_PAD            (gst_audio_aggregator_pad_get_type())
+#define GST_AUDIO_AGGREGATOR_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPad))
+#define GST_AUDIO_AGGREGATOR_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass))
+#define GST_AUDIO_AGGREGATOR_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass))
+#define GST_IS_AUDIO_AGGREGATOR_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD))
+#define GST_IS_AUDIO_AGGREGATOR_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD))
+
+/****************************
+ * GstAudioAggregatorPad Structs *
+ ***************************/
+
+typedef struct _GstAudioAggregatorPad GstAudioAggregatorPad;
+typedef struct _GstAudioAggregatorPadClass GstAudioAggregatorPadClass;
+typedef struct _GstAudioAggregatorPadPrivate GstAudioAggregatorPadPrivate;
+
+/**
+ * GstAudioAggregatorPad:
+ * @parent: The parent #GstAggregatorPad
+ * @info: The audio info for this pad set from the incoming caps
+ *
+ * The default implementation of GstPad used with #GstAudioAggregator
+ */
+struct _GstAudioAggregatorPad
+{
+  GstAggregatorPad                  parent;
+
+  GstAudioInfo                      info;
+
+  /*< private >*/
+  GstAudioAggregatorPadPrivate   *  priv;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorPadClass:
+ *
+ */
+struct _GstAudioAggregatorPadClass
+  {
+  GstAggregatorPadClass   parent_class;
+
+  /*< private >*/
+  gpointer      _gst_reserved[GST_PADDING_LARGE];
+};
+
+GST_EXPORT
+GType gst_audio_aggregator_pad_get_type           (void);
+
+#define GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD            (gst_audio_aggregator_convert_pad_get_type())
+#define GST_AUDIO_AGGREGATOR_CONVERT_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, GstAudioAggregatorConvertPad))
+#define GST_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, GstAudioAggregatorConvertPadClass))
+#define GST_AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, GstAudioAggregatorConvertPadClass))
+#define GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD))
+#define GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD))
+
+/****************************
+ * GstAudioAggregatorPad Structs *
+ ***************************/
+
+typedef struct _GstAudioAggregatorConvertPad GstAudioAggregatorConvertPad;
+typedef struct _GstAudioAggregatorConvertPadClass GstAudioAggregatorConvertPadClass;
+typedef struct _GstAudioAggregatorConvertPadPrivate GstAudioAggregatorConvertPadPrivate;
+
+/**
+ * GstAudioAggregatorConvertPad:
+ * @parent: The parent #GstAudioAggregatorPad
+ *
+ * An implementation of GstPad that can be used with #GstAudioAggregator.
+ *
+ * See #GstAudioAggregator for more details.
+ */
+struct _GstAudioAggregatorConvertPad
+{
+  GstAudioAggregatorPad                  parent;
+
+  /*< private >*/
+  GstAudioAggregatorConvertPadPrivate   *  priv;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorConvertPadClass:
+ *
+ */
+struct _GstAudioAggregatorConvertPadClass
+{
+  GstAudioAggregatorPadClass   parent_class;
+
+  /*< private >*/
+  gpointer      _gst_reserved[GST_PADDING];
+};
+
+GST_EXPORT
+GType gst_audio_aggregator_convert_pad_get_type           (void);
+
+/**************************
+ * GstAudioAggregator API *
+ **************************/
+
+#define GST_TYPE_AUDIO_AGGREGATOR            (gst_audio_aggregator_get_type())
+#define GST_AUDIO_AGGREGATOR(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregator))
+#define GST_AUDIO_AGGREGATOR_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass))
+#define GST_AUDIO_AGGREGATOR_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass))
+#define GST_IS_AUDIO_AGGREGATOR(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR))
+#define GST_IS_AUDIO_AGGREGATOR_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR))
+
+/**
+ * GstAudioAggregator:
+ * @parent: The parent #GstAggregator
+ * @info: The information parsed from the current caps
+ * @current_caps: The caps set by the subclass
+ *
+ * GstAudioAggregator object
+ */
+struct _GstAudioAggregator
+{
+  GstAggregator            parent;
+
+  /* All member are read only for subclasses, must hold OBJECT lock  */
+  GstAudioInfo    info;
+
+  GstCaps *current_caps;
+
+  /*< private >*/
+  GstAudioAggregatorPrivate *priv;
+
+  gpointer                 _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorClass:
+ * @create_output_buffer: Create a new output buffer contains num_frames frames.
+ * @aggregate_one_buffer: Aggregates one input buffer to the output
+ *  buffer.  The in_offset and out_offset are in "frames", which is
+ *  the size of a sample times the number of channels. Returns TRUE if
+ *  any non-silence was added to the buffer
+ * @convert_buffer: Convert a buffer from one format to another. The pad
+ *  is either a sinkpad, when converting an input buffer, or the source pad,
+ *  when converting the output buffer after a downstream format change is
+ *  requested.
+ */
+struct _GstAudioAggregatorClass {
+  GstAggregatorClass   parent_class;
+
+  GstBuffer * (* create_output_buffer) (GstAudioAggregator * aagg,
+      guint num_frames);
+  gboolean (* aggregate_one_buffer) (GstAudioAggregator * aagg,
+      GstAudioAggregatorPad * pad, GstBuffer * inbuf, guint in_offset,
+      GstBuffer * outbuf, guint out_offset, guint num_frames);
+  GstBuffer * (* convert_buffer) (GstAudioAggregator *aagg,
+                                  GstPad * pad,
+                                  GstAudioInfo *in_info,
+                                  GstAudioInfo *out_info,
+                                  GstBuffer * buffer);
+
+  /*< private >*/
+  gpointer          _gst_reserved[GST_PADDING_LARGE];
+};
+
+/*************************
+ * GstAggregator methods *
+ ************************/
+
+GST_EXPORT
+GType gst_audio_aggregator_get_type(void);
+
+GST_EXPORT
+void  gst_audio_aggregator_set_sink_caps (GstAudioAggregator    * aagg,
+                                          GstAudioAggregatorPad * pad,
+                                          GstCaps               * caps);
+
+GST_EXPORT
+void gst_audio_aggregator_class_perform_conversion (GstAudioAggregatorClass * klass);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_AGGREGATOR_H__ */
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,
+)
diff --git a/tests/check/elements/audiointerleave.c b/tests/check/elements/audiointerleave.c
new file mode 100644
index 000000000..71348f459
--- /dev/null
+++ b/tests/check/elements/audiointerleave.c
@@ -0,0 +1,1128 @@
+/* GStreamer unit tests for the audiointerleave element
+ * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
+ * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#ifdef HAVE_VALGRIND
+# include <valgrind/valgrind.h>
+#endif
+
+#include <gst/check/gstcheck.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/audio-enumtypes.h>
+
+#include <gst/check/gstharness.h>
+
+static void
+gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element,
+    GstCaps * caps, GstFormat format, const gchar * stream_id)
+{
+  GstSegment segment;
+
+  gst_segment_init (&segment, format);
+
+  fail_unless (gst_pad_push_event (srcpad,
+          gst_event_new_stream_start (stream_id)));
+  if (caps)
+    fail_unless (gst_pad_push_event (srcpad, gst_event_new_caps (caps)));
+  fail_unless (gst_pad_push_event (srcpad, gst_event_new_segment (&segment)));
+}
+
+GST_START_TEST (test_create_and_unref)
+{
+  GstElement *interleave;
+
+  interleave = gst_element_factory_make ("audiointerleave", NULL);
+  fail_unless (interleave != NULL);
+
+  gst_element_set_state (interleave, GST_STATE_NULL);
+  gst_object_unref (interleave);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_request_pads)
+{
+  GstElement *interleave;
+  GstPad *pad1, *pad2;
+
+  interleave = gst_element_factory_make ("audiointerleave", NULL);
+  fail_unless (interleave != NULL);
+
+  pad1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (pad1 != NULL);
+  fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink_0");
+
+  pad2 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (pad2 != NULL);
+  fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink_1");
+
+  gst_element_release_request_pad (interleave, pad2);
+  gst_object_unref (pad2);
+  gst_element_release_request_pad (interleave, pad1);
+  gst_object_unref (pad1);
+
+  gst_element_set_state (interleave, GST_STATE_NULL);
+  gst_object_unref (interleave);
+}
+
+GST_END_TEST;
+
+static GstPad **mysrcpads, *mysinkpad;
+static GstBus *bus;
+static GstElement *interleave;
+static GMutex data_mutex;
+static GCond data_cond;
+static gint have_data;
+static gfloat input[2];
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (F32) ", "
+        "channels = (int) 2, layout = (string) {interleaved, non-interleaved}, rate = (int) 48000"));
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (F32) ", "
+        "channels = (int) 1, layout = (string) interleaved, rate = (int) 48000"));
+
+#define CAPS_48khz \
+        "audio/x-raw, " \
+        "format = (string) " GST_AUDIO_NE (F32) ", " \
+        "channels = (int) 1, layout = (string) non-interleaved," \
+        "rate = (int) 48000"
+
+static GstFlowReturn
+interleave_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+  GstMapInfo map;
+  gfloat *outdata;
+  gint i;
+
+  fail_unless (GST_IS_BUFFER (buffer));
+  fail_unless (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP));
+  gst_buffer_map (buffer, &map, GST_MAP_READ);
+  outdata = (gfloat *) map.data;
+  fail_unless (outdata != NULL);
+
+#ifdef HAVE_VALGRIND
+  if (!(RUNNING_ON_VALGRIND))
+#endif
+    for (i = 0; i < map.size / sizeof (float); i += 2) {
+      fail_unless_equals_float (outdata[i], input[0]);
+      fail_unless_equals_float (outdata[i + 1], input[1]);
+    }
+
+  g_mutex_lock (&data_mutex);
+  have_data += map.size;
+  g_cond_signal (&data_cond);
+  g_mutex_unlock (&data_mutex);
+
+  gst_buffer_unmap (buffer, &map);
+  gst_buffer_unref (buffer);
+
+
+  return GST_FLOW_OK;
+}
+
+GST_START_TEST (test_audiointerleave_2ch)
+{
+  GstElement *queue;
+  GstPad *sink0, *sink1, *src, *tmp;
+  GstCaps *caps;
+  gint i;
+  GstBuffer *inbuf;
+  gfloat *indata;
+  GstMapInfo map;
+
+  mysrcpads = g_new0 (GstPad *, 2);
+
+  have_data = 0;
+
+  interleave = gst_element_factory_make ("audiointerleave", NULL);
+  fail_unless (interleave != NULL);
+
+  g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
+
+  queue = gst_element_factory_make ("queue", "queue");
+  fail_unless (queue != NULL);
+
+  sink0 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sink0 != NULL);
+  fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
+
+  sink1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sink1 != NULL);
+  fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
+
+  mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
+  fail_unless (mysrcpads[0] != NULL);
+
+  caps = gst_caps_from_string (CAPS_48khz);
+  gst_pad_set_active (mysrcpads[0], TRUE);
+  gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
+      GST_FORMAT_TIME, "0");
+  gst_pad_use_fixed_caps (mysrcpads[0]);
+
+  mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
+  fail_unless (mysrcpads[1] != NULL);
+
+  gst_pad_set_active (mysrcpads[1], TRUE);
+  gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
+      GST_FORMAT_TIME, "1");
+  gst_pad_use_fixed_caps (mysrcpads[1]);
+
+  tmp = gst_element_get_static_pad (queue, "sink");
+  fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  tmp = gst_element_get_static_pad (queue, "src");
+  fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
+
+  mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
+  fail_unless (mysinkpad != NULL);
+  gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  src = gst_element_get_static_pad (interleave, "src");
+  fail_unless (src != NULL);
+  fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
+  gst_object_unref (src);
+
+  bus = gst_bus_new ();
+  gst_element_set_bus (interleave, bus);
+
+  fail_unless (gst_element_set_state (interleave,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+  fail_unless (gst_element_set_state (queue,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+
+  input[0] = -1.0;
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  //GST_BUFFER_PTS (inbuf) = 0;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = -1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
+
+  input[1] = 1.0;
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  //GST_BUFFER_PTS (inbuf) = 0;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = 1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
+
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  //GST_BUFFER_PTS (inbuf) = GST_SECOND;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = -1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
+
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  //GST_BUFFER_PTS (inbuf) = GST_SECOND;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = 1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
+
+  g_mutex_lock (&data_mutex);
+  while (have_data < 48000 * 2 * 2 * sizeof (float))
+    g_cond_wait (&data_cond, &data_mutex);
+  g_mutex_unlock (&data_mutex);
+
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_state (interleave, GST_STATE_NULL);
+  gst_element_set_state (queue, GST_STATE_NULL);
+
+  gst_object_unref (mysrcpads[0]);
+  gst_object_unref (mysrcpads[1]);
+  gst_object_unref (mysinkpad);
+
+  gst_element_release_request_pad (interleave, sink0);
+  gst_object_unref (sink0);
+  gst_element_release_request_pad (interleave, sink1);
+  gst_object_unref (sink1);
+
+  gst_object_unref (interleave);
+  gst_object_unref (queue);
+  gst_object_unref (bus);
+  gst_caps_unref (caps);
+
+  g_free (mysrcpads);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_audiointerleave_2ch_1eos)
+{
+  GstElement *queue;
+  GstPad *sink0, *sink1, *src, *tmp;
+  GstCaps *caps;
+  gint i;
+  GstBuffer *inbuf;
+  gfloat *indata;
+  GstMapInfo map;
+
+  mysrcpads = g_new0 (GstPad *, 2);
+
+  have_data = 0;
+
+  interleave = gst_element_factory_make ("audiointerleave", NULL);
+  fail_unless (interleave != NULL);
+
+  g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
+
+  queue = gst_element_factory_make ("queue", "queue");
+  fail_unless (queue != NULL);
+
+  sink0 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sink0 != NULL);
+  fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
+
+  sink1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sink1 != NULL);
+  fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
+
+  mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
+  fail_unless (mysrcpads[0] != NULL);
+
+  caps = gst_caps_from_string (CAPS_48khz);
+  gst_pad_set_active (mysrcpads[0], TRUE);
+  gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
+      GST_FORMAT_TIME, "0");
+  gst_pad_use_fixed_caps (mysrcpads[0]);
+
+  mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
+  fail_unless (mysrcpads[1] != NULL);
+
+  gst_pad_set_active (mysrcpads[1], TRUE);
+  gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
+      GST_FORMAT_TIME, "1");
+  gst_pad_use_fixed_caps (mysrcpads[1]);
+
+  tmp = gst_element_get_static_pad (queue, "sink");
+  fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  tmp = gst_element_get_static_pad (queue, "src");
+  fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
+
+  mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
+  fail_unless (mysinkpad != NULL);
+  gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  src = gst_element_get_static_pad (interleave, "src");
+  fail_unless (src != NULL);
+  fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
+  gst_object_unref (src);
+
+  bus = gst_bus_new ();
+  gst_element_set_bus (interleave, bus);
+
+  fail_unless (gst_element_set_state (interleave,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+  fail_unless (gst_element_set_state (queue,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+
+  input[0] = -1.0;
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  GST_BUFFER_PTS (inbuf) = 0;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = -1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
+
+  input[1] = 1.0;
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  GST_BUFFER_PTS (inbuf) = 0;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = 1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
+
+  g_mutex_lock (&data_mutex);
+  /* 48000 samples per buffer * 2 sources * 2 buffers */
+  while (have_data != 48000 * 2 * sizeof (float))
+    g_cond_wait (&data_cond, &data_mutex);
+  g_mutex_unlock (&data_mutex);
+
+  input[0] = 0.0;
+  gst_pad_push_event (mysrcpads[0], gst_event_new_eos ());
+
+  input[1] = 1.0;
+  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
+  GST_BUFFER_PTS (inbuf) = GST_SECOND;
+  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
+  indata = (gfloat *) map.data;
+  for (i = 0; i < 48000; i++)
+    indata[i] = 1.0;
+  gst_buffer_unmap (inbuf, &map);
+  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
+
+  g_mutex_lock (&data_mutex);
+  /* 48000 samples per buffer * 2 sources * 2 buffers */
+  while (have_data != 48000 * 2 * 2 * sizeof (float))
+    g_cond_wait (&data_cond, &data_mutex);
+  g_mutex_unlock (&data_mutex);
+
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_state (interleave, GST_STATE_NULL);
+  gst_element_set_state (queue, GST_STATE_NULL);
+
+  gst_object_unref (mysrcpads[0]);
+  gst_object_unref (mysrcpads[1]);
+  gst_object_unref (mysinkpad);
+
+  gst_element_release_request_pad (interleave, sink0);
+  gst_object_unref (sink0);
+  gst_element_release_request_pad (interleave, sink1);
+  gst_object_unref (sink1);
+
+  gst_object_unref (interleave);
+  gst_object_unref (queue);
+  gst_object_unref (bus);
+  gst_caps_unref (caps);
+
+  g_free (mysrcpads);
+}
+
+GST_END_TEST;
+
+static void
+src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
+    gboolean interleaved, gpointer user_data)
+{
+  gint n = GPOINTER_TO_INT (user_data);
+  gfloat *data;
+  gint i, num_samples;
+  GstCaps *caps;
+  guint64 mask;
+  GstAudioChannelPosition pos;
+  GstMapInfo map;
+
+  fail_unless (gst_buffer_is_writable (buffer));
+
+  switch (n) {
+    case 0:
+    case 1:
+    case 2:
+      pos = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+      break;
+    case 3:
+      pos = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+      break;
+    default:
+      pos = GST_AUDIO_CHANNEL_POSITION_INVALID;
+      break;
+  }
+
+  mask = G_GUINT64_CONSTANT (1) << pos;
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
+      "channels", G_TYPE_INT, 1,
+      "layout", G_TYPE_STRING, interleaved ? "interleaved" : "non-interleaved",
+      "channel-mask", GST_TYPE_BITMASK, mask, "rate", G_TYPE_INT, 48000, NULL);
+
+  gst_pad_set_caps (pad, caps);
+  gst_caps_unref (caps);
+
+  fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE));
+  fail_unless (map.size % sizeof (gfloat) == 0);
+
+  fail_unless (map.size > 480);
+
+  num_samples = map.size / sizeof (gfloat);
+  data = (gfloat *) map.data;
+
+  for (i = 0; i < num_samples; i++)
+    data[i] = (n % 2 == 0) ? -1.0 : 1.0;
+
+  gst_buffer_unmap (buffer, &map);
+}
+
+static void
+src_handoff_float32_audiointerleaved (GstElement * element, GstBuffer * buffer,
+    GstPad * pad, gpointer user_data)
+{
+  src_handoff_float32 (element, buffer, pad, TRUE, user_data);
+}
+
+static void
+src_handoff_float32_non_audiointerleaved (GstElement * element,
+    GstBuffer * buffer, GstPad * pad, gpointer user_data)
+{
+  src_handoff_float32 (element, buffer, pad, FALSE, user_data);
+}
+
+static void
+sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
+    gpointer user_data)
+{
+  gint i;
+  GstMapInfo map;
+  gfloat *data;
+  GstCaps *caps, *ccaps;
+  gint n = GPOINTER_TO_INT (user_data);
+  guint64 mask;
+
+  fail_unless (GST_IS_BUFFER (buffer));
+  gst_buffer_map (buffer, &map, GST_MAP_READ);
+  data = (gfloat *) map.data;
+
+  /* Give a little leeway for rounding errors */
+  fail_unless (gst_util_uint64_scale (map.size, GST_SECOND,
+          48000 * 2 * sizeof (gfloat)) <= GST_BUFFER_DURATION (buffer) + 1 ||
+      gst_util_uint64_scale (map.size, GST_SECOND,
+          48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1);
+
+  if (n == 0 || n == 3) {
+    GstAudioChannelPosition pos[2] =
+        { GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
+    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
+  } else if (n == 1) {
+    GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
+    };
+    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
+  } else if (n == 2) {
+    GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+      GST_AUDIO_CHANNEL_POSITION_REAR_CENTER
+    };
+    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
+  } else {
+    g_assert_not_reached ();
+  }
+
+  if (pad) {
+    caps = gst_caps_new_simple ("audio/x-raw",
+        "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
+        "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
+        "layout", G_TYPE_STRING, "interleaved",
+        "channel-mask", GST_TYPE_BITMASK, mask, NULL);
+
+    ccaps = gst_pad_get_current_caps (pad);
+    fail_unless (gst_caps_is_equal (caps, ccaps));
+    gst_caps_unref (ccaps);
+    gst_caps_unref (caps);
+  }
+#ifdef HAVE_VALGRIND
+  if (!(RUNNING_ON_VALGRIND))
+#endif
+    for (i = 0; i < map.size / sizeof (float); i += 2) {
+      fail_unless_equals_float (data[i], -1.0);
+      if (n != 3)
+        fail_unless_equals_float (data[i + 1], 1.0);
+    }
+  have_data += map.size;
+
+  gst_buffer_unmap (buffer, &map);
+
+}
+
+static void
+test_audiointerleave_2ch_pipeline (gboolean interleaved)
+{
+  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
+  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
+  GstMessage *msg;
+  void *src_handoff_float32 =
+      interleaved ? &src_handoff_float32_audiointerleaved :
+      &src_handoff_float32_non_audiointerleaved;
+
+  have_data = 0;
+
+  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
+  fail_unless (pipeline != NULL);
+
+  src1 = gst_element_factory_make ("fakesrc", "src1");
+  fail_unless (src1 != NULL);
+  g_object_set (src1, "num-buffers", 4, NULL);
+  g_object_set (src1, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src1, "signal-handoffs", TRUE, NULL);
+  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
+      GINT_TO_POINTER (0));
+  gst_bin_add (GST_BIN (pipeline), src1);
+
+  src2 = gst_element_factory_make ("fakesrc", "src2");
+  fail_unless (src2 != NULL);
+  g_object_set (src2, "num-buffers", 4, NULL);
+  g_object_set (src2, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src2, "signal-handoffs", TRUE, NULL);
+  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
+      GINT_TO_POINTER (1));
+  gst_bin_add (GST_BIN (pipeline), src2);
+
+  queue = gst_element_factory_make ("queue", "queue");
+  fail_unless (queue != NULL);
+  gst_bin_add (GST_BIN (pipeline), queue);
+
+  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
+  fail_unless (interleave != NULL);
+  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
+
+  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad0 != NULL);
+  tmp = gst_element_get_static_pad (src1, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad1 != NULL);
+  tmp = gst_element_get_static_pad (src2, "src");
+  tmp2 = gst_element_get_static_pad (queue, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+  tmp = gst_element_get_static_pad (queue, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sink = gst_element_factory_make ("fakesink", "sink");
+  fail_unless (sink != NULL);
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
+      GINT_TO_POINTER (0));
+  gst_bin_add (GST_BIN (pipeline), sink);
+  tmp = gst_element_get_static_pad (interleave, "src");
+  tmp2 = gst_element_get_static_pad (sink, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
+  gst_message_unref (msg);
+
+  /* 48000 samples per buffer * 2 sources * 4 buffers */
+  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_element_release_request_pad (interleave, sinkpad0);
+  gst_object_unref (sinkpad0);
+  gst_element_release_request_pad (interleave, sinkpad1);
+  gst_object_unref (sinkpad1);
+  gst_object_unref (interleave);
+  gst_object_unref (pipeline);
+}
+
+GST_START_TEST (test_audiointerleave_2ch_pipeline_audiointerleaved)
+{
+  test_audiointerleave_2ch_pipeline (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_audiointerleave_2ch_pipeline_non_audiointerleaved)
+{
+  test_audiointerleave_2ch_pipeline (FALSE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
+{
+  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
+  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
+  GstMessage *msg;
+
+  have_data = 0;
+
+  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
+  fail_unless (pipeline != NULL);
+
+  src1 = gst_element_factory_make ("fakesrc", "src1");
+  fail_unless (src1 != NULL);
+  g_object_set (src1, "num-buffers", 4, NULL);
+  g_object_set (src1, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src1, "signal-handoffs", TRUE, NULL);
+  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src1, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
+  gst_bin_add (GST_BIN (pipeline), src1);
+
+  src2 = gst_element_factory_make ("fakesrc", "src2");
+  fail_unless (src2 != NULL);
+  g_object_set (src2, "num-buffers", 4, NULL);
+  g_object_set (src2, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src2, "signal-handoffs", TRUE, NULL);
+  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src2, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (3));
+  gst_bin_add (GST_BIN (pipeline), src2);
+
+  queue = gst_element_factory_make ("queue", "queue");
+  fail_unless (queue != NULL);
+  gst_bin_add (GST_BIN (pipeline), queue);
+
+  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
+  fail_unless (interleave != NULL);
+  g_object_set (interleave, "channel-positions-from-input", TRUE, NULL);
+  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
+
+  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad0 != NULL);
+  tmp = gst_element_get_static_pad (src1, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad1 != NULL);
+  tmp = gst_element_get_static_pad (src2, "src");
+  tmp2 = gst_element_get_static_pad (queue, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+  tmp = gst_element_get_static_pad (queue, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sink = gst_element_factory_make ("fakesink", "sink");
+  fail_unless (sink != NULL);
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
+      GINT_TO_POINTER (1));
+  gst_bin_add (GST_BIN (pipeline), sink);
+  tmp = gst_element_get_static_pad (interleave, "src");
+  tmp2 = gst_element_get_static_pad (sink, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
+  gst_message_unref (msg);
+
+  /* 48000 samples per buffer * 2 sources * 4 buffers */
+  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_element_release_request_pad (interleave, sinkpad0);
+  gst_object_unref (sinkpad0);
+  gst_element_release_request_pad (interleave, sinkpad1);
+  gst_object_unref (sinkpad1);
+  gst_object_unref (interleave);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
+{
+  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
+  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
+  GstMessage *msg;
+  GValueArray *arr;
+  GValue val = { 0, };
+
+  have_data = 0;
+
+  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
+  fail_unless (pipeline != NULL);
+
+  src1 = gst_element_factory_make ("fakesrc", "src1");
+  fail_unless (src1 != NULL);
+  g_object_set (src1, "num-buffers", 4, NULL);
+  g_object_set (src1, "signal-handoffs", TRUE, NULL);
+  g_object_set (src1, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src1, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
+  gst_bin_add (GST_BIN (pipeline), src1);
+
+  src2 = gst_element_factory_make ("fakesrc", "src2");
+  fail_unless (src2 != NULL);
+  g_object_set (src2, "num-buffers", 4, NULL);
+  g_object_set (src2, "signal-handoffs", TRUE, NULL);
+  g_object_set (src2, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src2, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
+  gst_bin_add (GST_BIN (pipeline), src2);
+
+  queue = gst_element_factory_make ("queue", "queue");
+  fail_unless (queue != NULL);
+  gst_bin_add (GST_BIN (pipeline), queue);
+
+  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
+  fail_unless (interleave != NULL);
+  g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
+  arr = g_value_array_new (2);
+
+  g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
+  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER);
+  g_value_array_append (arr, &val);
+  g_value_reset (&val);
+  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER);
+  g_value_array_append (arr, &val);
+  g_value_unset (&val);
+
+  g_object_set (interleave, "channel-positions", arr, NULL);
+  g_value_array_free (arr);
+  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
+
+  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad0 != NULL);
+  tmp = gst_element_get_static_pad (src1, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad1 != NULL);
+  tmp = gst_element_get_static_pad (src2, "src");
+  tmp2 = gst_element_get_static_pad (queue, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+  tmp = gst_element_get_static_pad (queue, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sink = gst_element_factory_make ("fakesink", "sink");
+  fail_unless (sink != NULL);
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
+      GINT_TO_POINTER (2));
+  gst_bin_add (GST_BIN (pipeline), sink);
+  tmp = gst_element_get_static_pad (interleave, "src");
+  tmp2 = gst_element_get_static_pad (sink, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
+  gst_message_unref (msg);
+
+  /* 48000 samples per buffer * 2 sources * 4 buffers */
+  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_element_release_request_pad (interleave, sinkpad0);
+  gst_object_unref (sinkpad0);
+  gst_element_release_request_pad (interleave, sinkpad1);
+  gst_object_unref (sinkpad1);
+  gst_object_unref (interleave);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
+{
+  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
+  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
+  GstMessage *msg;
+
+  have_data = 0;
+
+  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
+  fail_unless (pipeline != NULL);
+
+  src1 = gst_element_factory_make ("fakesrc", "src1");
+  fail_unless (src1 != NULL);
+  g_object_set (src1, "num-buffers", 4, NULL);
+  g_object_set (src1, "signal-handoffs", TRUE, NULL);
+  g_object_set (src1, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src1, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
+  gst_bin_add (GST_BIN (pipeline), src1);
+
+  src2 = gst_element_factory_make ("fakesrc", "src2");
+  fail_unless (src2 != NULL);
+  g_object_set (src2, "num-buffers", 4, NULL);
+  g_object_set (src2, "signal-handoffs", TRUE, NULL);
+  g_object_set (src2, "sizetype", 2,
+      "sizemax", (int) 48000 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (src2, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
+  gst_bin_add (GST_BIN (pipeline), src2);
+
+  queue = gst_element_factory_make ("queue", "queue");
+  fail_unless (queue != NULL);
+  gst_bin_add (GST_BIN (pipeline), queue);
+
+  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
+  fail_unless (interleave != NULL);
+  g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
+  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
+
+  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad0 != NULL);
+  tmp = gst_element_get_static_pad (src1, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
+  fail_unless (sinkpad1 != NULL);
+  tmp = gst_element_get_static_pad (src2, "src");
+  tmp2 = gst_element_get_static_pad (queue, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+  tmp = gst_element_get_static_pad (queue, "src");
+  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+
+  sink = gst_element_factory_make ("fakesink", "sink");
+  fail_unless (sink != NULL);
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
+      GINT_TO_POINTER (0));
+  gst_bin_add (GST_BIN (pipeline), sink);
+  tmp = gst_element_get_static_pad (interleave, "src");
+  tmp2 = gst_element_get_static_pad (sink, "sink");
+  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
+  gst_object_unref (tmp);
+  gst_object_unref (tmp2);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
+  gst_message_unref (msg);
+
+  /* 48000 samples per buffer * 2 sources * 4 buffers */
+  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_element_release_request_pad (interleave, sinkpad0);
+  gst_object_unref (sinkpad0);
+  gst_element_release_request_pad (interleave, sinkpad1);
+  gst_object_unref (sinkpad1);
+  gst_object_unref (interleave);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+static void
+forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type)
+{
+  GstEvent *e;
+
+  e = gst_harness_pull_event (hsrc);
+  fail_unless (GST_EVENT_TYPE (e) == type);
+  gst_harness_push_event (h, e);
+}
+
+GST_START_TEST (test_audiointerleave_2ch_smallbuf)
+{
+  GstElement *audiointerleave;
+  GstHarness *hsrc;
+  GstHarness *h;
+  GstHarness *h2;
+  GstBuffer *buffer;
+  gint i;
+  GstEvent *ev;
+  GstCaps *ecaps, *caps;
+
+  audiointerleave = gst_element_factory_make ("audiointerleave", NULL);
+
+  g_object_set (audiointerleave, "latency", GST_SECOND / 2,
+      "output-buffer-duration", GST_SECOND / 4, NULL);
+
+  h = gst_harness_new_with_element (audiointerleave, "sink_0", "src");
+  gst_harness_use_testclock (h);
+
+  h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL);
+  gst_harness_set_src_caps_str (h2, "audio/x-raw, "
+      "format=" GST_AUDIO_NE (F32) ", channels=(int)1,"
+      " layout=interleaved, rate=48000, channel-mask=(bitmask)8");
+
+  hsrc = gst_harness_new ("fakesrc");
+  gst_harness_use_testclock (hsrc);
+  g_object_set (hsrc->element,
+      "is-live", TRUE,
+      "sync", TRUE,
+      "signal-handoffs", TRUE,
+      "format", GST_FORMAT_TIME,
+      "sizetype", 2,
+      "sizemax", (int) 480 * sizeof (gfloat),
+      "datarate", (int) 48000 * sizeof (gfloat), NULL);
+  g_signal_connect (hsrc->element, "handoff",
+      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
+  gst_harness_play (hsrc);
+
+  gst_harness_crank_single_clock_wait (hsrc);
+  forward_check_event (h, hsrc, GST_EVENT_STREAM_START);
+  forward_check_event (h, hsrc, GST_EVENT_CAPS);
+  forward_check_event (h, hsrc, GST_EVENT_SEGMENT);
+  gst_harness_push (h, gst_harness_pull (hsrc));        /* buffer */
+
+  for (i = 0; i < 24; i++) {
+    gst_harness_crank_single_clock_wait (hsrc);
+    forward_check_event (h, hsrc, GST_EVENT_CAPS);
+    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
+  }
+
+  gst_harness_crank_single_clock_wait (h);
+
+
+  gst_event_unref (gst_harness_pull_event (h)); /* stream-start */
+  ev = gst_harness_pull_event (h);      /* caps */
+  fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev));
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
+      "channels", G_TYPE_INT, 2,
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK,
+      (guint64) 0x9, NULL);
+
+  gst_event_parse_caps (ev, &ecaps);
+  gst_check_caps_equal (ecaps, caps);
+  gst_caps_unref (caps);
+  gst_event_unref (ev);
+
+  /* eat the caps processing */
+  gst_harness_crank_single_clock_wait (h);
+  for (i = 0; i < 23; i++)
+    gst_harness_crank_single_clock_wait (h);
+  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
+          (h->element)), 750 * GST_MSECOND);
+
+  buffer = gst_harness_pull (h);
+  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
+  gst_buffer_unref (buffer);
+  fail_unless_equals_int (gst_harness_buffers_received (h), 1);
+
+  for (i = 0; i < 50; i++) {
+    gst_harness_crank_single_clock_wait (hsrc);
+    forward_check_event (h, hsrc, GST_EVENT_CAPS);
+    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
+  }
+  for (i = 0; i < 25; i++)
+    gst_harness_crank_single_clock_wait (h);
+  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
+          (h->element)), 1000 * GST_MSECOND);
+  buffer = gst_harness_pull (h);
+  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
+  gst_buffer_unref (buffer);
+  fail_unless_equals_int (gst_harness_buffers_received (h), 2);
+
+  for (i = 0; i < 25; i++) {
+    gst_harness_crank_single_clock_wait (hsrc);
+    forward_check_event (h, hsrc, GST_EVENT_CAPS);
+    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
+  }
+  for (i = 0; i < 25; i++)
+    gst_harness_crank_single_clock_wait (h);
+  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
+          (h->element)), 1250 * GST_MSECOND);
+  buffer = gst_harness_pull (h);
+  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
+  gst_buffer_unref (buffer);
+  fail_unless_equals_int (gst_harness_buffers_received (h), 3);
+
+  gst_harness_push_event (h, gst_event_new_eos ());
+
+  for (i = 0; i < 25; i++)
+    gst_harness_crank_single_clock_wait (h);
+  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
+          (h->element)), 1500 * GST_MSECOND);
+  buffer = gst_harness_pull (h);
+  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
+  gst_buffer_unref (buffer);
+
+  fail_unless_equals_int (gst_harness_buffers_received (h), 4);
+
+  gst_harness_teardown (h2);
+  gst_harness_teardown (h);
+  gst_harness_teardown (hsrc);
+  gst_object_unref (audiointerleave);
+}
+
+GST_END_TEST;
+
+static Suite *
+audiointerleave_suite (void)
+{
+  Suite *s = suite_create ("audiointerleave");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_set_timeout (tc_chain, 180);
+  tcase_add_test (tc_chain, test_create_and_unref);
+  tcase_add_test (tc_chain, test_request_pads);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch_1eos);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_audiointerleaved);
+  tcase_add_test (tc_chain,
+      test_audiointerleave_2ch_pipeline_non_audiointerleaved);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos);
+  tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf);
+
+  return s;
+}
+
+GST_CHECK_MAIN (audiointerleave);
diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c
new file mode 100644
index 000000000..4a8a8233b
--- /dev/null
+++ b/tests/check/elements/audiomixer.c
@@ -0,0 +1,1894 @@
+/* GStreamer
+ *
+ * unit test for audiomixer
+ *
+ * Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
+ * Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include <config.h>
+#endif
+
+#ifdef HAVE_VALGRIND
+# include <valgrind/valgrind.h>
+#endif
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstconsistencychecker.h>
+#include <gst/audio/audio.h>
+#include <gst/base/gstbasesrc.h>
+#include <gst/controller/gstdirectcontrolbinding.h>
+#include <gst/controller/gstinterpolationcontrolsource.h>
+
+static GMainLoop *main_loop;
+
+/* fixtures */
+
+static void
+test_setup (void)
+{
+  main_loop = g_main_loop_new (NULL, FALSE);
+}
+
+static void
+test_teardown (void)
+{
+  g_main_loop_unref (main_loop);
+  main_loop = NULL;
+}
+
+
+/* some test helpers */
+
+static GstElement *
+setup_pipeline (GstElement * audiomixer, gint num_srcs, GstElement * capsfilter)
+{
+  GstElement *pipeline, *src, *sink;
+  gint i;
+
+  pipeline = gst_pipeline_new ("pipeline");
+  if (!audiomixer) {
+    audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  }
+
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (pipeline), audiomixer, sink, NULL);
+
+  if (capsfilter) {
+    gst_bin_add (GST_BIN (pipeline), capsfilter);
+    gst_element_link_many (audiomixer, capsfilter, sink, NULL);
+  } else {
+    gst_element_link (audiomixer, sink);
+  }
+
+  for (i = 0; i < num_srcs; i++) {
+    src = gst_element_factory_make ("audiotestsrc", NULL);
+    g_object_set (src, "wave", 4, NULL);        /* silence */
+    gst_bin_add (GST_BIN (pipeline), src);
+    gst_element_link (src, audiomixer);
+  }
+  return pipeline;
+}
+
+static GstCaps *
+get_element_sink_pad_caps (GstElement * pipeline, const gchar * element_name)
+{
+  GstElement *sink;
+  GstCaps *caps;
+  GstPad *pad;
+
+  sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
+  pad = gst_element_get_static_pad (sink, "sink");
+  caps = gst_pad_get_current_caps (pad);
+  gst_object_unref (pad);
+  gst_object_unref (sink);
+
+  return caps;
+}
+
+static void
+set_state_and_wait (GstElement * pipeline, GstState state)
+{
+  GstStateChangeReturn state_res;
+
+  /* prepare paused/playing */
+  state_res = gst_element_set_state (pipeline, state);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* wait for preroll */
+  state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+}
+
+static gboolean
+set_playing (GstElement * element)
+{
+  GstStateChangeReturn state_res;
+
+  state_res = gst_element_set_state (element, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  return FALSE;
+}
+
+static void
+play_and_wait (GstElement * pipeline)
+{
+  GstStateChangeReturn state_res;
+
+  g_idle_add ((GSourceFunc) set_playing, pipeline);
+
+  GST_INFO ("running main loop");
+  g_main_loop_run (main_loop);
+
+  state_res = gst_element_set_state (pipeline, GST_STATE_NULL);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+}
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+      GST_MESSAGE_SRC (message), message);
+
+  switch (message->type) {
+    case GST_MESSAGE_EOS:
+      g_main_loop_quit (main_loop);
+      break;
+    case GST_MESSAGE_WARNING:{
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_warning (message, &gerror, &debug);
+      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      break;
+    }
+    case GST_MESSAGE_ERROR:{
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_error (message, &gerror, &debug);
+      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      g_main_loop_quit (main_loop);
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+static GstBuffer *
+new_buffer (gsize num_bytes, gint data, GstClockTime ts, GstClockTime dur,
+    GstBufferFlags flags)
+{
+  GstMapInfo map;
+  GstBuffer *buffer = gst_buffer_new_and_alloc (num_bytes);
+
+  gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+  memset (map.data, data, map.size);
+  gst_buffer_unmap (buffer, &map);
+  GST_BUFFER_TIMESTAMP (buffer) = ts;
+  GST_BUFFER_DURATION (buffer) = dur;
+  if (flags)
+    GST_BUFFER_FLAG_SET (buffer, flags);
+  GST_DEBUG ("created buffer %p", buffer);
+  return buffer;
+}
+
+/* make sure downstream gets a CAPS event before buffers are sent */
+GST_START_TEST (test_caps)
+{
+  GstElement *pipeline;
+  GstCaps *caps;
+
+  /* build pipeline */
+  pipeline = setup_pipeline (NULL, 1, NULL);
+
+  /* prepare playing */
+  set_state_and_wait (pipeline, GST_STATE_PAUSED);
+
+  /* check caps on fakesink */
+  caps = get_element_sink_pad_caps (pipeline, "sink");
+  fail_unless (caps != NULL);
+  gst_caps_unref (caps);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+/* check that caps set on the property are honoured */
+GST_START_TEST (test_filter_caps)
+{
+  GstElement *pipeline, *audiomixer, *capsfilter;
+  GstCaps *filter_caps, *caps;
+
+  filter_caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
+      "channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL);
+
+  capsfilter = gst_element_factory_make ("capsfilter", NULL);
+
+  /* build pipeline */
+  audiomixer = gst_element_factory_make ("audiomixer", NULL);
+  g_object_set (capsfilter, "caps", filter_caps, NULL);
+  pipeline = setup_pipeline (audiomixer, 1, capsfilter);
+
+  /* prepare playing */
+  set_state_and_wait (pipeline, GST_STATE_PAUSED);
+
+  /* check caps on fakesink */
+  caps = get_element_sink_pad_caps (pipeline, "sink");
+  fail_unless (caps != NULL);
+  GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps);
+  fail_unless (gst_caps_is_equal_fixed (caps, filter_caps));
+  gst_caps_unref (caps);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+
+  gst_caps_unref (filter_caps);
+}
+
+GST_END_TEST;
+
+static GstFormat format = GST_FORMAT_UNDEFINED;
+static gint64 position = -1;
+
+static void
+test_event_message_received (GstBus * bus, GstMessage * message,
+    GstPipeline * bin)
+{
+  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+      GST_MESSAGE_SRC (message), message);
+
+  switch (message->type) {
+    case GST_MESSAGE_SEGMENT_DONE:
+      gst_message_parse_segment_done (message, &format, &position);
+      GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position);
+      g_main_loop_quit (main_loop);
+      break;
+    default:
+      g_assert_not_reached ();
+      break;
+  }
+}
+
+
+GST_START_TEST (test_event)
+{
+  GstElement *bin, *src1, *src2, *audiomixer, *sink;
+  GstBus *bus;
+  GstEvent *seek_event;
+  gboolean res;
+  GstPad *srcpad, *sinkpad;
+  GstStreamConsistency *chk_1, *chk_2, *chk_3;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  src1 = gst_element_factory_make ("audiotestsrc", "src1");
+  g_object_set (src1, "wave", 4, NULL); /* silence */
+  src2 = gst_element_factory_make ("audiotestsrc", "src2");
+  g_object_set (src2, "wave", 4, NULL); /* silence */
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
+
+  res = gst_element_link (src1, audiomixer);
+  fail_unless (res == TRUE, NULL);
+  res = gst_element_link (src2, audiomixer);
+  fail_unless (res == TRUE, NULL);
+  res = gst_element_link (audiomixer, sink);
+  fail_unless (res == TRUE, NULL);
+
+  srcpad = gst_element_get_static_pad (audiomixer, "src");
+  chk_3 = gst_consistency_checker_new (srcpad);
+  gst_object_unref (srcpad);
+
+  /* create consistency checkers for the pads */
+  srcpad = gst_element_get_static_pad (src1, "src");
+  chk_1 = gst_consistency_checker_new (srcpad);
+  sinkpad = gst_pad_get_peer (srcpad);
+  gst_consistency_checker_add_pad (chk_3, sinkpad);
+  gst_object_unref (sinkpad);
+  gst_object_unref (srcpad);
+
+  srcpad = gst_element_get_static_pad (src2, "src");
+  chk_2 = gst_consistency_checker_new (srcpad);
+  sinkpad = gst_pad_get_peer (srcpad);
+  gst_consistency_checker_add_pad (chk_3, sinkpad);
+  gst_object_unref (sinkpad);
+  gst_object_unref (srcpad);
+
+  seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
+      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
+      GST_SEEK_TYPE_SET, (GstClockTime) 0,
+      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
+
+  format = GST_FORMAT_UNDEFINED;
+  position = -1;
+
+  g_signal_connect (bus, "message::segment-done",
+      (GCallback) test_event_message_received, bin);
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  GST_INFO ("starting test");
+
+  /* prepare playing */
+  set_state_and_wait (bin, GST_STATE_PAUSED);
+
+  res = gst_element_send_event (bin, seek_event);
+  fail_unless (res == TRUE, NULL);
+
+  /* run pipeline */
+  play_and_wait (bin);
+
+  ck_assert_int_eq (position, 2 * GST_SECOND);
+
+  /* cleanup */
+  gst_consistency_checker_free (chk_1);
+  gst_consistency_checker_free (chk_2);
+  gst_consistency_checker_free (chk_3);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+static guint play_count = 0;
+static GstEvent *play_seek_event = NULL;
+
+static void
+test_play_twice_message_received (GstBus * bus, GstMessage * message,
+    GstElement * bin)
+{
+  gboolean res;
+  GstStateChangeReturn state_res;
+
+  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+      GST_MESSAGE_SRC (message), message);
+
+  switch (message->type) {
+    case GST_MESSAGE_SEGMENT_DONE:
+      play_count++;
+      if (play_count == 1) {
+        state_res = gst_element_set_state (bin, GST_STATE_READY);
+        ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+        /* prepare playing again */
+        set_state_and_wait (bin, GST_STATE_PAUSED);
+
+        res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+        fail_unless (res == TRUE, NULL);
+
+        state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
+        ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+      } else {
+        g_main_loop_quit (main_loop);
+      }
+      break;
+    default:
+      g_assert_not_reached ();
+      break;
+  }
+}
+
+
+GST_START_TEST (test_play_twice)
+{
+  GstElement *bin, *audiomixer;
+  GstBus *bus;
+  gboolean res;
+  GstPad *srcpad;
+  GstStreamConsistency *consist;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  bin = setup_pipeline (audiomixer, 2, NULL);
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  srcpad = gst_element_get_static_pad (audiomixer, "src");
+  consist = gst_consistency_checker_new (srcpad);
+  gst_object_unref (srcpad);
+
+  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
+      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
+      GST_SEEK_TYPE_SET, (GstClockTime) 0,
+      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
+
+  play_count = 0;
+
+  g_signal_connect (bus, "message::segment-done",
+      (GCallback) test_play_twice_message_received, bin);
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  GST_INFO ("starting test");
+
+  /* prepare playing */
+  set_state_and_wait (bin, GST_STATE_PAUSED);
+
+  res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+  fail_unless (res == TRUE, NULL);
+
+  GST_INFO ("seeked");
+
+  /* run pipeline */
+  play_and_wait (bin);
+
+  ck_assert_int_eq (play_count, 2);
+
+  /* cleanup */
+  gst_consistency_checker_free (consist);
+  gst_event_unref (play_seek_event);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_twice_then_add_and_play_again)
+{
+  GstElement *bin, *src, *audiomixer;
+  GstBus *bus;
+  gboolean res;
+  GstStateChangeReturn state_res;
+  gint i;
+  GstPad *srcpad;
+  GstStreamConsistency *consist;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  bin = setup_pipeline (audiomixer, 2, NULL);
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  srcpad = gst_element_get_static_pad (audiomixer, "src");
+  consist = gst_consistency_checker_new (srcpad);
+  gst_object_unref (srcpad);
+
+  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
+      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
+      GST_SEEK_TYPE_SET, (GstClockTime) 0,
+      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
+
+  g_signal_connect (bus, "message::segment-done",
+      (GCallback) test_play_twice_message_received, bin);
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  /* run it twice */
+  for (i = 0; i < 2; i++) {
+    play_count = 0;
+
+    GST_INFO ("starting test-loop %d", i);
+
+    /* prepare playing */
+    set_state_and_wait (bin, GST_STATE_PAUSED);
+
+    res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+    fail_unless (res == TRUE, NULL);
+
+    GST_INFO ("seeked");
+
+    /* run pipeline */
+    play_and_wait (bin);
+
+    ck_assert_int_eq (play_count, 2);
+
+    /* plug another source */
+    if (i == 0) {
+      src = gst_element_factory_make ("audiotestsrc", NULL);
+      g_object_set (src, "wave", 4, NULL);      /* silence */
+      gst_bin_add (GST_BIN (bin), src);
+
+      res = gst_element_link (src, audiomixer);
+      fail_unless (res == TRUE, NULL);
+    }
+
+    gst_consistency_checker_reset (consist);
+  }
+
+  state_res = gst_element_set_state (bin, GST_STATE_NULL);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* cleanup */
+  gst_event_unref (play_seek_event);
+  gst_consistency_checker_free (consist);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+
+static GstElement *
+test_live_seeking_try_audiosrc (const gchar * factory_name)
+{
+  GstElement *src;
+  GstStateChangeReturn state_res;
+
+  if (!(src = gst_element_factory_make (factory_name, NULL))) {
+    GST_INFO ("can't make '%s', skipping", factory_name);
+    return NULL;
+  }
+
+  /* Test that the audio source can get to ready, else skip */
+  state_res = gst_element_set_state (src, GST_STATE_READY);
+  gst_element_set_state (src, GST_STATE_NULL);
+
+  if (state_res == GST_STATE_CHANGE_FAILURE) {
+    GST_INFO_OBJECT (src, "can't go to ready, skipping");
+    gst_object_unref (src);
+    return NULL;
+  }
+
+  return src;
+}
+
+/* test failing seeks on live-sources */
+GST_START_TEST (test_live_seeking)
+{
+  GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink;
+  GstCaps *caps;
+  GstBus *bus;
+  gboolean res;
+  GstPad *srcpad;
+  GstPad *sinkpad;
+  gint i;
+  GstStreamConsistency *consist;
+  /* don't use autoaudiosrc, as then we can't set anything here */
+  const gchar *audio_src_factories[] = {
+    "alsasrc",
+    "pulseaudiosrc"
+  };
+
+  GST_INFO ("preparing test");
+  play_seek_event = NULL;
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  for (i = 0; (i < G_N_ELEMENTS (audio_src_factories) && src1 == NULL); i++) {
+    src1 = test_live_seeking_try_audiosrc (audio_src_factories[i]);
+  }
+  if (!src1) {
+    /* normal audiosources behave differently than audiotestsrc */
+    GST_WARNING ("no real audiosrc found, using audiotestsrc is-live");
+    src1 = gst_element_factory_make ("audiotestsrc", "src1");
+    g_object_set (src1, "wave", 4, "is-live", TRUE, NULL);      /* silence */
+  } else {
+    /* live sources ignore seeks, force eos after 2 sec (4 buffers half second
+     * each)
+     */
+    g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL);
+  }
+
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  cf = gst_element_factory_make ("capsfilter", "capsfilter");
+  sink = gst_element_factory_make ("fakesink", "sink");
+
+  gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL);
+  res = gst_element_link_many (src1, cf, audiomixer, sink, NULL);
+  fail_unless (res == TRUE, NULL);
+
+  /* get the caps for the livesrc, we'll reuse this for the non-live source */
+  set_state_and_wait (bin, GST_STATE_PLAYING);
+
+  sinkpad = gst_element_get_static_pad (sink, "sink");
+  fail_unless (sinkpad != NULL);
+  caps = gst_pad_get_current_caps (sinkpad);
+  fail_unless (caps != NULL);
+  gst_object_unref (sinkpad);
+
+  gst_element_set_state (bin, GST_STATE_NULL);
+
+  g_object_set (cf, "caps", caps, NULL);
+
+  src2 = gst_element_factory_make ("audiotestsrc", "src2");
+  g_object_set (src2, "wave", 4, NULL); /* silence */
+  gst_bin_add (GST_BIN (bin), src2);
+
+  res = gst_element_link_filtered (src2, audiomixer, caps);
+  fail_unless (res == TRUE, NULL);
+
+  gst_caps_unref (caps);
+
+  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
+      GST_SEEK_FLAG_FLUSH,
+      GST_SEEK_TYPE_SET, (GstClockTime) 0,
+      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
+
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  srcpad = gst_element_get_static_pad (audiomixer, "src");
+  consist = gst_consistency_checker_new (srcpad);
+  gst_object_unref (srcpad);
+
+  GST_INFO ("starting test");
+
+  /* run it twice */
+  for (i = 0; i < 2; i++) {
+
+    GST_INFO ("starting test-loop %d", i);
+
+    /* prepare playing */
+    set_state_and_wait (bin, GST_STATE_PAUSED);
+
+    res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+    fail_unless (res == TRUE, NULL);
+
+    GST_INFO ("seeked");
+
+    /* run pipeline */
+    play_and_wait (bin);
+
+    gst_consistency_checker_reset (consist);
+  }
+
+  /* cleanup */
+  GST_INFO ("cleaning up");
+  gst_consistency_checker_free (consist);
+  if (play_seek_event)
+    gst_event_unref (play_seek_event);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+/* check if adding pads work as expected */
+GST_START_TEST (test_add_pad)
+{
+  GstElement *bin, *src1, *src2, *audiomixer, *sink;
+  GstBus *bus;
+  GstPad *srcpad;
+  gboolean res;
+  GstStateChangeReturn state_res;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  src1 = gst_element_factory_make ("audiotestsrc", "src1");
+  g_object_set (src1, "num-buffers", 4, "wave", /* silence */ 4, NULL);
+  src2 = gst_element_factory_make ("audiotestsrc", "src2");
+  /* one buffer less, we connect with 1 buffer of delay */
+  g_object_set (src2, "num-buffers", 3, "wave", /* silence */ 4, NULL);
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL);
+
+  res = gst_element_link (src1, audiomixer);
+  fail_unless (res == TRUE, NULL);
+  res = gst_element_link (audiomixer, sink);
+  fail_unless (res == TRUE, NULL);
+
+  srcpad = gst_element_get_static_pad (audiomixer, "src");
+  gst_object_unref (srcpad);
+
+  g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
+      bin);
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  GST_INFO ("starting test");
+
+  /* prepare playing */
+  set_state_and_wait (bin, GST_STATE_PAUSED);
+
+  /* add other element */
+  gst_bin_add_many (GST_BIN (bin), src2, NULL);
+
+  /* now link the second element */
+  res = gst_element_link (src2, audiomixer);
+  fail_unless (res == TRUE, NULL);
+
+  /* set to PAUSED as well */
+  state_res = gst_element_set_state (src2, GST_STATE_PAUSED);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* now play all */
+  play_and_wait (bin);
+
+  /* cleanup */
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+/* check if removing pads work as expected */
+GST_START_TEST (test_remove_pad)
+{
+  GstElement *bin, *src, *audiomixer, *sink;
+  GstBus *bus;
+  GstPad *pad, *srcpad;
+  gboolean res;
+  GstStateChangeReturn state_res;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  src = gst_element_factory_make ("audiotestsrc", "src");
+  g_object_set (src, "num-buffers", 4, "wave", 4, NULL);
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL);
+
+  res = gst_element_link (src, audiomixer);
+  fail_unless (res == TRUE, NULL);
+  res = gst_element_link (audiomixer, sink);
+  fail_unless (res == TRUE, NULL);
+
+  /* create an unconnected sinkpad in audiomixer */
+  pad = gst_element_get_request_pad (audiomixer, "sink_%u");
+  fail_if (pad == NULL, NULL);
+
+  srcpad = gst_element_get_static_pad (audiomixer, "src");
+  gst_object_unref (srcpad);
+
+  g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
+      bin);
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  GST_INFO ("starting test");
+
+  /* prepare playing, this will not preroll as audiomixer is waiting
+   * on the unconnected sinkpad. */
+  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* wait for completion for one second, will return ASYNC */
+  state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND);
+  ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC);
+
+  /* get rid of the pad now, audiomixer should stop waiting on it and
+   * continue the preroll */
+  gst_element_release_request_pad (audiomixer, pad);
+  gst_object_unref (pad);
+
+  /* wait for completion, should work now */
+  state_res =
+      gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+      GST_CLOCK_TIME_NONE);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* now play all */
+  play_and_wait (bin);
+
+  /* cleanup */
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (G_OBJECT (bus));
+  gst_object_unref (G_OBJECT (bin));
+}
+
+GST_END_TEST;
+
+
+static GstBuffer *handoff_buffer = NULL;
+
+static void
+handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
+    gpointer user_data)
+{
+  GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT
+      " -- %p PTS is %" GST_TIME_FORMAT " END is %" GST_TIME_FORMAT,
+      gst_buffer_get_size (buffer), buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
+
+  gst_buffer_replace (&handoff_buffer, buffer);
+}
+
+/* check if clipping works as expected */
+GST_START_TEST (test_clip)
+{
+  GstSegment segment;
+  GstElement *bin, *audiomixer, *sink;
+  GstBus *bus;
+  GstPad *sinkpad;
+  gboolean res;
+  GstStateChangeReturn state_res;
+  GstFlowReturn ret;
+  GstEvent *event;
+  GstBuffer *buffer;
+  GstCaps *caps;
+  GstQuery *drain = gst_query_new_drain ();
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  /* just an audiomixer and a fakesink */
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  g_object_set (audiomixer, "output-buffer-duration", 50 * GST_MSECOND, NULL);
+  sink = gst_element_factory_make ("fakesink", "sink");
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
+  gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
+
+  res = gst_element_link (audiomixer, sink);
+  fail_unless (res == TRUE, NULL);
+
+  /* set to playing */
+  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* create an unconnected sinkpad in audiomixer, should also automatically activate
+   * the pad */
+  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
+  fail_if (sinkpad == NULL, NULL);
+
+  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
+
+  gst_pad_set_caps (sinkpad, caps);
+  gst_caps_unref (caps);
+
+  /* send segment to audiomixer */
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  segment.start = GST_SECOND;
+  segment.stop = 2 * GST_SECOND;
+  segment.time = 0;
+  event = gst_event_new_segment (&segment);
+  gst_pad_send_event (sinkpad, event);
+
+  /* should be clipped and ok */
+  buffer = new_buffer (44100, 0, 0, 250 * GST_MSECOND, 0);
+  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
+      buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+  /* The aggregation is done in a dedicated thread, so we can't
+   * know when it is actually going to happen, so we use a DRAIN query
+   * to wait for it to complete.
+   */
+  gst_pad_query (sinkpad, drain);
+  fail_unless (handoff_buffer == NULL);
+
+  /* should be partially clipped */
+  buffer = new_buffer (44100, 0, 900 * GST_MSECOND, 250 * GST_MSECOND,
+      GST_BUFFER_FLAG_DISCONT);
+  GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
+      GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+  gst_pad_query (sinkpad, drain);
+
+  fail_unless (handoff_buffer != NULL);
+  ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
+      GST_BUFFER_DURATION (handoff_buffer), 150 * GST_MSECOND);
+  gst_buffer_replace (&handoff_buffer, NULL);
+
+  /* should not be clipped */
+  buffer = new_buffer (44100, 0, 1150 * GST_MSECOND, 250 * GST_MSECOND, 0);
+  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
+      buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+  gst_pad_query (sinkpad, drain);
+  fail_unless (handoff_buffer != NULL);
+  ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
+      GST_BUFFER_DURATION (handoff_buffer), 400 * GST_MSECOND);
+  gst_buffer_replace (&handoff_buffer, NULL);
+  fail_unless (handoff_buffer == NULL);
+
+  /* should be clipped and ok */
+  buffer = new_buffer (44100, 0, 2 * GST_SECOND, 250 * GST_MSECOND,
+      GST_BUFFER_FLAG_DISCONT);
+  GST_DEBUG ("pushing buffer %p PTS is %" GST_TIME_FORMAT
+      " END is %" GST_TIME_FORMAT,
+      buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+  gst_pad_query (sinkpad, drain);
+  fail_unless (handoff_buffer == NULL);
+
+  gst_element_release_request_pad (audiomixer, sinkpad);
+  gst_object_unref (sinkpad);
+  gst_element_set_state (bin, GST_STATE_NULL);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+  gst_query_unref (drain);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_duration_is_max)
+{
+  GstElement *bin, *src[3], *audiomixer, *sink;
+  GstStateChangeReturn state_res;
+  GstFormat format = GST_FORMAT_TIME;
+  gboolean res;
+  gint64 duration;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+
+  /* 3 sources, an audiomixer and a fakesink */
+  src[0] = gst_element_factory_make ("audiotestsrc", NULL);
+  src[1] = gst_element_factory_make ("audiotestsrc", NULL);
+  src[2] = gst_element_factory_make ("audiotestsrc", NULL);
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
+      NULL);
+
+  gst_element_link (src[0], audiomixer);
+  gst_element_link (src[1], audiomixer);
+  gst_element_link (src[2], audiomixer);
+  gst_element_link (audiomixer, sink);
+
+  /* irks, duration is reset on basesrc */
+  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
+  fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
+
+  /* set durations on src */
+  GST_BASE_SRC (src[0])->segment.duration = 1000;
+  GST_BASE_SRC (src[1])->segment.duration = 3000;
+  GST_BASE_SRC (src[2])->segment.duration = 2000;
+
+  /* set to playing */
+  set_state_and_wait (bin, GST_STATE_PLAYING);
+
+  res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
+  fail_unless (res, NULL);
+
+  ck_assert_int_eq (duration, 3000);
+
+  gst_element_set_state (bin, GST_STATE_NULL);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_duration_unknown_overrides)
+{
+  GstElement *bin, *src[3], *audiomixer, *sink;
+  GstStateChangeReturn state_res;
+  GstFormat format = GST_FORMAT_TIME;
+  gboolean res;
+  gint64 duration;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+
+  /* 3 sources, an audiomixer and a fakesink */
+  src[0] = gst_element_factory_make ("audiotestsrc", NULL);
+  src[1] = gst_element_factory_make ("audiotestsrc", NULL);
+  src[2] = gst_element_factory_make ("audiotestsrc", NULL);
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
+      NULL);
+
+  gst_element_link (src[0], audiomixer);
+  gst_element_link (src[1], audiomixer);
+  gst_element_link (src[2], audiomixer);
+  gst_element_link (audiomixer, sink);
+
+  /* irks, duration is reset on basesrc */
+  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
+  fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
+
+  /* set durations on src */
+  GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE;
+  GST_BASE_SRC (src[1])->segment.duration = 3000;
+  GST_BASE_SRC (src[2])->segment.duration = 2000;
+
+  /* set to playing */
+  set_state_and_wait (bin, GST_STATE_PLAYING);
+
+  res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
+  fail_unless (res, NULL);
+
+  ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE);
+
+  gst_element_set_state (bin, GST_STATE_NULL);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+
+static gboolean looped = FALSE;
+
+static void
+loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin)
+{
+  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+      GST_MESSAGE_SRC (message), message);
+
+  if (looped) {
+    g_main_loop_quit (main_loop);
+  } else {
+    GstEvent *seek_event;
+    gboolean res;
+
+    seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
+        GST_SEEK_FLAG_SEGMENT,
+        GST_SEEK_TYPE_SET, (GstClockTime) 0,
+        GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
+
+    res = gst_element_send_event (bin, seek_event);
+    fail_unless (res == TRUE, NULL);
+    looped = TRUE;
+  }
+}
+
+GST_START_TEST (test_loop)
+{
+  GstElement *bin;
+  GstBus *bus;
+  GstEvent *seek_event;
+  gboolean res;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = setup_pipeline (NULL, 2, NULL);
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
+      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
+      GST_SEEK_TYPE_SET, (GstClockTime) 0,
+      GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
+
+  g_signal_connect (bus, "message::segment-done",
+      (GCallback) loop_segment_done, bin);
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  GST_INFO ("starting test");
+
+  /* prepare playing */
+  set_state_and_wait (bin, GST_STATE_PAUSED);
+
+  res = gst_element_send_event (bin, seek_event);
+  fail_unless (res == TRUE, NULL);
+
+  /* run pipeline */
+  play_and_wait (bin);
+
+  fail_unless (looped);
+
+  /* cleanup */
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_flush_start_flush_stop)
+{
+  GstPadTemplate *sink_template;
+  GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src;
+  GstElement *pipeline, *src1, *src2, *audiomixer, *sink;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  pipeline = gst_pipeline_new ("pipeline");
+  src1 = gst_element_factory_make ("audiotestsrc", "src1");
+  g_object_set (src1, "wave", 4, NULL); /* silence */
+  src2 = gst_element_factory_make ("audiotestsrc", "src2");
+  g_object_set (src2, "wave", 4, NULL); /* silence */
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL);
+
+  sink_template =
+      gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer),
+      "sink_%u");
+  fail_unless (GST_IS_PAD_TEMPLATE (sink_template));
+  sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
+  srcpad1 = gst_element_get_static_pad (src1, "src");
+  gst_pad_link (srcpad1, sinkpad1);
+
+  sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
+  tmppad = gst_element_get_static_pad (src2, "src");
+  gst_pad_link (tmppad, sinkpad2);
+  gst_object_unref (tmppad);
+
+  gst_element_link (audiomixer, sink);
+
+  /* prepare playing */
+  set_state_and_wait (pipeline, GST_STATE_PLAYING);
+
+  audiomixer_src = gst_element_get_static_pad (audiomixer, "src");
+  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
+  gst_pad_send_event (sinkpad1, gst_event_new_flush_start ());
+  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
+  fail_unless (GST_PAD_IS_FLUSHING (sinkpad1));
+  /* Hold the streamlock to make sure the flush stop is not between
+     the attempted push of a segment event and of the following buffer. */
+  GST_PAD_STREAM_LOCK (srcpad1);
+  gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE));
+  GST_PAD_STREAM_UNLOCK (srcpad1);
+  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
+  fail_if (GST_PAD_IS_FLUSHING (sinkpad1));
+  gst_object_unref (audiomixer_src);
+
+  gst_element_release_request_pad (audiomixer, sinkpad1);
+  gst_object_unref (sinkpad1);
+  gst_element_release_request_pad (audiomixer, sinkpad2);
+  gst_object_unref (sinkpad2);
+  gst_object_unref (srcpad1);
+
+  /* cleanup */
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+static void
+handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer,
+    GstPad * pad, gpointer user_data)
+{
+  GList **received_buffers = user_data;
+
+  GST_DEBUG ("got buffer %p", buffer);
+  *received_buffers =
+      g_list_append (*received_buffers, gst_buffer_ref (buffer));
+}
+
+typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2);
+typedef void (*CheckBuffersFunction) (GList * buffers);
+
+static void
+run_sync_test (SendBuffersFunction send_buffers,
+    CheckBuffersFunction check_buffers)
+{
+  GstSegment segment;
+  GstElement *bin, *audiomixer, *queue1, *queue2, *sink;
+  GstBus *bus;
+  GstPad *sinkpad1, *sinkpad2;
+  GstPad *queue1_sinkpad, *queue2_sinkpad;
+  GstPad *pad;
+  gboolean res;
+  GstStateChangeReturn state_res;
+  GstEvent *event;
+  GstCaps *caps;
+  GList *received_buffers = NULL;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  /* just an audiomixer and a fakesink */
+  queue1 = gst_element_factory_make ("queue", "queue1");
+  queue2 = gst_element_factory_make ("queue", "queue2");
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL);
+  sink = gst_element_factory_make ("fakesink", "sink");
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,
+      &received_buffers);
+  gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL);
+
+  res = gst_element_link (audiomixer, sink);
+  fail_unless (res == TRUE, NULL);
+
+  /* set to paused */
+  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* create an unconnected sinkpad in audiomixer, should also automatically activate
+   * the pad */
+  sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u");
+  fail_if (sinkpad1 == NULL, NULL);
+
+  queue1_sinkpad = gst_element_get_static_pad (queue1, "sink");
+  pad = gst_element_get_static_pad (queue1, "src");
+  fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK);
+  gst_object_unref (pad);
+
+  sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u");
+  fail_if (sinkpad2 == NULL, NULL);
+
+  queue2_sinkpad = gst_element_get_static_pad (queue2, "sink");
+  pad = gst_element_get_static_pad (queue2, "src");
+  fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK);
+  gst_object_unref (pad);
+
+  gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test"));
+  gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test"));
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
+
+  gst_pad_set_caps (queue1_sinkpad, caps);
+  gst_pad_set_caps (queue2_sinkpad, caps);
+  gst_caps_unref (caps);
+
+  /* send segment to audiomixer */
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  event = gst_event_new_segment (&segment);
+  gst_pad_send_event (queue1_sinkpad, gst_event_ref (event));
+  gst_pad_send_event (queue2_sinkpad, event);
+
+  /* Push buffers */
+  send_buffers (queue1_sinkpad, queue2_sinkpad);
+
+  /* Set PLAYING */
+  g_idle_add ((GSourceFunc) set_playing, bin);
+
+  /* Collect buffers and messages */
+  g_main_loop_run (main_loop);
+
+  /* Here we get once we got EOS, for errors we failed */
+
+  check_buffers (received_buffers);
+
+  g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref);
+
+  gst_element_release_request_pad (audiomixer, sinkpad1);
+  gst_object_unref (sinkpad1);
+  gst_object_unref (queue1_sinkpad);
+  gst_element_release_request_pad (audiomixer, sinkpad2);
+  gst_object_unref (sinkpad2);
+  gst_object_unref (queue2_sinkpad);
+  gst_element_set_state (bin, GST_STATE_NULL);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+static void
+send_buffers_sync (GstPad * pad1, GstPad * pad2)
+{
+  GstBuffer *buffer;
+  GstFlowReturn ret;
+
+  buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad1, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (2000, 1, 2 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad1, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  gst_pad_send_event (pad1, gst_event_new_eos ());
+
+  buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad2, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad2, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  gst_pad_send_event (pad2, gst_event_new_eos ());
+}
+
+static void
+check_buffers_sync (GList * received_buffers)
+{
+  GstBuffer *buffer;
+  GList *l;
+  gint i;
+  GstMapInfo map;
+
+  /* Should have 8 * 0.5s buffers */
+  fail_unless_equals_int (g_list_length (received_buffers), 8);
+  for (i = 0, l = received_buffers; l; l = l->next, i++) {
+    buffer = l->data;
+
+    gst_buffer_map (buffer, &map, GST_MAP_READ);
+
+    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
+      fail_unless (map.data[0] == 0);
+      fail_unless (map.data[map.size - 1] == 0);
+    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 0);
+      fail_unless (map.data[map.size - 1] == 0);
+    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 1);
+      fail_unless (map.data[map.size - 1] == 1);
+    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 1);
+      fail_unless (map.data[map.size - 1] == 1);
+    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 3);
+      fail_unless (map.data[map.size - 1] == 3);
+    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 3);
+      fail_unless (map.data[map.size - 1] == 3);
+    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 2);
+      fail_unless (map.data[map.size - 1] == 2);
+    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 2);
+      fail_unless (map.data[map.size - 1] == 2);
+    } else {
+      g_assert_not_reached ();
+    }
+
+    gst_buffer_unmap (buffer, &map);
+
+  }
+}
+
+GST_START_TEST (test_sync)
+{
+  run_sync_test (send_buffers_sync, check_buffers_sync);
+}
+
+GST_END_TEST;
+
+static void
+send_buffers_sync_discont (GstPad * pad1, GstPad * pad2)
+{
+  GstBuffer *buffer;
+  GstFlowReturn ret;
+
+  buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad1, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (2000, 1, 3 * GST_SECOND, 1 * GST_SECOND,
+      GST_BUFFER_FLAG_DISCONT);
+  ret = gst_pad_chain (pad1, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  gst_pad_send_event (pad1, gst_event_new_eos ());
+
+  buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad2, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad2, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  gst_pad_send_event (pad2, gst_event_new_eos ());
+}
+
+static void
+check_buffers_sync_discont (GList * received_buffers)
+{
+  GstBuffer *buffer;
+  GList *l;
+  gint i;
+  GstMapInfo map;
+
+  /* Should have 8 * 0.5s buffers */
+  fail_unless_equals_int (g_list_length (received_buffers), 8);
+  for (i = 0, l = received_buffers; l; l = l->next, i++) {
+    buffer = l->data;
+
+    gst_buffer_map (buffer, &map, GST_MAP_READ);
+
+    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
+      fail_unless (map.data[0] == 0);
+      fail_unless (map.data[map.size - 1] == 0);
+    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 0);
+      fail_unless (map.data[map.size - 1] == 0);
+    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 1);
+      fail_unless (map.data[map.size - 1] == 1);
+    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 1);
+      fail_unless (map.data[map.size - 1] == 1);
+    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 2);
+      fail_unless (map.data[map.size - 1] == 2);
+    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 2);
+      fail_unless (map.data[map.size - 1] == 2);
+    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 3);
+      fail_unless (map.data[map.size - 1] == 3);
+    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 3);
+      fail_unless (map.data[map.size - 1] == 3);
+    } else {
+      g_assert_not_reached ();
+    }
+
+    gst_buffer_unmap (buffer, &map);
+
+  }
+}
+
+GST_START_TEST (test_sync_discont)
+{
+  run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont);
+}
+
+GST_END_TEST;
+
+static void
+send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2)
+{
+  GstBuffer *buffer;
+  GstFlowReturn ret;
+
+  buffer = new_buffer (2000, 1, 750 * GST_MSECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad1, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (2000, 1, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad1, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  gst_pad_send_event (pad1, gst_event_new_eos ());
+
+  buffer = new_buffer (2000, 2, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad2, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (2000, 2, 2750 * GST_MSECOND, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (pad2, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  gst_pad_send_event (pad2, gst_event_new_eos ());
+}
+
+static void
+check_buffers_sync_unaligned (GList * received_buffers)
+{
+  GstBuffer *buffer;
+  GList *l;
+  gint i;
+  GstMapInfo map;
+
+  /* Should have 8 * 0.5s buffers */
+  fail_unless_equals_int (g_list_length (received_buffers), 8);
+  for (i = 0, l = received_buffers; l; l = l->next, i++) {
+    buffer = l->data;
+
+    gst_buffer_map (buffer, &map, GST_MAP_READ);
+
+    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
+      fail_unless (map.data[0] == 0);
+      fail_unless (map.data[map.size - 1] == 0);
+    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 0);
+      fail_unless (map.data[499] == 0);
+      fail_unless (map.data[500] == 1);
+      fail_unless (map.data[map.size - 1] == 1);
+    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 1);
+      fail_unless (map.data[map.size - 1] == 1);
+    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 1);
+      fail_unless (map.data[499] == 1);
+      fail_unless (map.data[500] == 3);
+      fail_unless (map.data[map.size - 1] == 3);
+    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 3);
+      fail_unless (map.data[499] == 3);
+      fail_unless (map.data[500] == 3);
+      fail_unless (map.data[map.size - 1] == 3);
+    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
+      fail_unless (map.data[0] == 3);
+      fail_unless (map.data[499] == 3);
+      fail_unless (map.data[500] == 2);
+      fail_unless (map.data[map.size - 1] == 2);
+    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
+      fail_unless (map.data[0] == 2);
+      fail_unless (map.data[499] == 2);
+      fail_unless (map.data[500] == 2);
+      fail_unless (map.data[map.size - 1] == 2);
+    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
+      fail_unless (map.size == 500);
+      fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND);
+      fail_unless (map.data[0] == 2);
+      fail_unless (map.data[499] == 2);
+    } else {
+      g_assert_not_reached ();
+    }
+
+    gst_buffer_unmap (buffer, &map);
+
+  }
+}
+
+GST_START_TEST (test_sync_unaligned)
+{
+  run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_segment_base_handling)
+{
+  GstElement *pipeline, *sink, *mix, *src1, *src2;
+  GstPad *srcpad, *sinkpad;
+  GstClockTime end_time;
+  GstSample *last_sample = NULL;
+  GstSample *sample;
+  GstBuffer *buf;
+  GstCaps *caps;
+
+  caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100,
+      "channels", G_TYPE_INT, 2, NULL);
+
+  pipeline = gst_pipeline_new ("pipeline");
+  mix = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("appsink", "sink");
+  g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
+  gst_caps_unref (caps);
+  /* 50 buffers of 1/10 sec = 5 sec */
+  src1 = gst_element_factory_make ("audiotestsrc", "src1");
+  g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
+  src2 = gst_element_factory_make ("audiotestsrc", "src2");
+  g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
+  gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL);
+  fail_unless (gst_element_link (mix, sink));
+
+  srcpad = gst_element_get_static_pad (src1, "src");
+  sinkpad = gst_element_get_request_pad (mix, "sink_1");
+  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
+  gst_object_unref (sinkpad);
+  gst_object_unref (srcpad);
+
+  srcpad = gst_element_get_static_pad (src2, "src");
+  sinkpad = gst_element_get_request_pad (mix, "sink_2");
+  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
+  /* set a pad offset of another 5 seconds */
+  gst_pad_set_offset (sinkpad, 5 * GST_SECOND);
+  gst_object_unref (sinkpad);
+  gst_object_unref (srcpad);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  do {
+    g_signal_emit_by_name (sink, "pull-sample", &sample);
+    if (sample == NULL)
+      break;
+    if (last_sample)
+      gst_sample_unref (last_sample);
+    last_sample = sample;
+  } while (TRUE);
+
+  buf = gst_sample_get_buffer (last_sample);
+  end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+  fail_unless_equals_int64 (end_time, 10 * GST_SECOND);
+  gst_sample_unref (last_sample);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+static void
+set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
+    GstClockTime end, gdouble end_value)
+{
+  GstControlSource *cs;
+  GstTimedValueControlSource *tvcs;
+
+  cs = gst_interpolation_control_source_new ();
+  fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
+          gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
+              "volume", cs)));
+
+  /* set volume interpolation mode */
+  g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
+
+  tvcs = (GstTimedValueControlSource *) cs;
+  fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
+  fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
+  gst_object_unref (cs);
+}
+
+GST_START_TEST (test_sinkpad_property_controller)
+{
+  GstBus *bus;
+  GstMessage *msg;
+  GstElement *pipeline, *sink, *mix, *src1;
+  GstPad *srcpad, *sinkpad;
+  GError *error = NULL;
+  gchar *debug;
+
+  pipeline = gst_pipeline_new ("pipeline");
+  mix = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  src1 = gst_element_factory_make ("audiotestsrc", "src1");
+  g_object_set (src1, "num-buffers", 100, NULL);
+  gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
+  fail_unless (gst_element_link (mix, sink));
+
+  srcpad = gst_element_get_static_pad (src1, "src");
+  sinkpad = gst_element_get_request_pad (mix, "sink_0");
+  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
+  set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
+  gst_object_unref (sinkpad);
+  gst_object_unref (srcpad);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+  msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
+      GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_ERROR:
+      gst_message_parse_error (msg, &error, &debug);
+      g_printerr ("ERROR from element %s: %s\n",
+          GST_OBJECT_NAME (msg->src), error->message);
+      g_printerr ("Debug info: %s\n", debug);
+      g_error_free (error);
+      g_free (debug);
+      break;
+    case GST_MESSAGE_EOS:
+      break;
+    default:
+      g_assert_not_reached ();
+  }
+  gst_message_unref (msg);
+  g_object_unref (bus);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
+static void
+change_src_caps (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
+    GstElement * capsfilter)
+{
+  GstCaps *caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
+
+  g_object_set (capsfilter, "caps", caps, NULL);
+  g_signal_connect (fakesink, "handoff", (GCallback) handoff_buffer_cb, NULL);
+  g_signal_handlers_disconnect_by_func (fakesink, change_src_caps, capsfilter);
+}
+
+/* In this test, we create an input buffer with a duration of 2 seconds,
+ * and require the audiomixer to output 1 second long buffers.
+ * The input buffer will thus be mixed twice, and the audiomixer will
+ * output two buffers.
+ *
+ * After audiomixer has output a first buffer, we change its output format
+ * from S8 to S32. As our sample rate stays the same at 10 fps, and we use
+ * mono, the first buffer should be 10 bytes long, and the second 40.
+ *
+ * The input buffer is made up of 15 0-valued bytes, and 5 1-valued bytes.
+ * We verify that the second buffer contains 5 0-valued integers, and
+ * 5 1 << 24 valued integers.
+ */
+GST_START_TEST (test_change_output_caps)
+{
+  GstSegment segment;
+  GstElement *bin, *audiomixer, *capsfilter, *sink;
+  GstBus *bus;
+  GstPad *sinkpad;
+  gboolean res;
+  GstStateChangeReturn state_res;
+  GstFlowReturn ret;
+  GstEvent *event;
+  GstBuffer *buffer;
+  GstCaps *caps;
+  GstQuery *drain = gst_query_new_drain ();
+  GstMapInfo inmap;
+  GstMapInfo outmap;
+  gsize i;
+
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  g_object_set (audiomixer, "output-buffer-duration", GST_SECOND, NULL);
+  capsfilter = gst_element_factory_make ("capsfilter", NULL);
+  sink = gst_element_factory_make ("fakesink", "sink");
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", (GCallback) change_src_caps, capsfilter);
+  gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
+
+  res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
+  fail_unless (res == TRUE, NULL);
+
+  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
+  fail_if (sinkpad == NULL, NULL);
+
+  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, "S8",
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
+
+  gst_pad_set_caps (sinkpad, caps);
+  g_object_set (capsfilter, "caps", caps, NULL);
+  gst_caps_unref (caps);
+
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  segment.start = 0;
+  segment.stop = 2 * GST_SECOND;
+  segment.time = 0;
+  event = gst_event_new_segment (&segment);
+  gst_pad_send_event (sinkpad, event);
+
+  gst_buffer_replace (&handoff_buffer, NULL);
+
+  buffer = new_buffer (20, 0, 0, 2 * GST_SECOND, 0);
+  gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
+  memset (inmap.data + 15, 1, 5);
+  gst_buffer_unmap (buffer, &inmap);
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+  gst_pad_query (sinkpad, drain);
+  fail_unless (handoff_buffer != NULL);
+  fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 40);
+
+  gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
+  for (i = 0; i < 10; i++) {
+    guint32 sample;
+
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+    sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
+#else
+    sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
+#endif
+
+    if (i < 5) {
+      fail_unless_equals_int (sample, 0);
+    } else {
+      fail_unless_equals_int (sample, 1 << 24);
+    }
+  }
+  gst_buffer_unmap (handoff_buffer, &outmap);
+
+  gst_element_release_request_pad (audiomixer, sinkpad);
+  gst_object_unref (sinkpad);
+  gst_element_set_state (bin, GST_STATE_NULL);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+  gst_query_unref (drain);
+}
+
+GST_END_TEST;
+
+static Suite *
+audiomixer_suite (void)
+{
+  Suite *s = suite_create ("audiomixer");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_caps);
+  tcase_add_test (tc_chain, test_filter_caps);
+  tcase_add_test (tc_chain, test_event);
+  tcase_add_test (tc_chain, test_play_twice);
+  tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again);
+  tcase_add_test (tc_chain, test_live_seeking);
+  tcase_add_test (tc_chain, test_add_pad);
+  tcase_add_test (tc_chain, test_remove_pad);
+  tcase_add_test (tc_chain, test_clip);
+  tcase_add_test (tc_chain, test_duration_is_max);
+  tcase_add_test (tc_chain, test_duration_unknown_overrides);
+  tcase_add_test (tc_chain, test_loop);
+  tcase_add_test (tc_chain, test_flush_start_flush_stop);
+  tcase_add_test (tc_chain, test_sync);
+  tcase_add_test (tc_chain, test_sync_discont);
+  tcase_add_test (tc_chain, test_sync_unaligned);
+  tcase_add_test (tc_chain, test_segment_base_handling);
+  tcase_add_test (tc_chain, test_sinkpad_property_controller);
+  tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
+  tcase_add_test (tc_chain, test_change_output_caps);
+
+  /* Use a longer timeout */
+#ifdef HAVE_VALGRIND
+  if (RUNNING_ON_VALGRIND) {
+    tcase_set_timeout (tc_chain, 5 * 60);
+  } else
+#endif
+  {
+    /* this is shorter than the default 60 seconds?! (tpm) */
+    /* tcase_set_timeout (tc_chain, 6); */
+  }
+
+  return s;
+}
+
+GST_CHECK_MAIN (audiomixer);