audioaggregator: remove, moved to -base

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

6 files changed, 3 insertions, 2238 deletions

diff --git a/docs/libs/gst-plugins-bad-libs.types b/docs/libs/gst-plugins-bad-libs.types
index 7b4851c42..ba43ffdbc 100644
--- a/docs/libs/gst-plugins-bad-libs.types
+++ b/docs/libs/gst-plugins-bad-libs.types
@@ -1,6 +1,5 @@
 #include <gst/gst.h>
 
-#include <gst/audio/gstaudioaggregator.h>
 #include <gst/video/gstvideoaggregator.h>
 #include <gst/codecparsers/gsth264parser.h>
 #include <gst/codecparsers/gstmpegvideoparser.h>
@@ -9,9 +8,6 @@
 #include <gst/player/player.h>
 #include <gst/webrtc/webrtc.h>
 
-gst_audio_aggregator_get_type
-gst_audio_aggregator_pad_get_type
-
 gst_video_aggregator_get_type
 gst_video_aggregator_pad_get_type
 
diff --git a/docs/plugins/gst-plugins-bad-plugins.hierarchy b/docs/plugins/gst-plugins-bad-plugins.hierarchy
index 63c911ea3..9455b4786 100644
--- a/docs/plugins/gst-plugins-bad-plugins.hierarchy
+++ b/docs/plugins/gst-plugins-bad-plugins.hierarchy
@@ -17,10 +17,6 @@ GObject
       GstControlSource
       GstElement
         GstAggregator
-          GstAudioAggregator
-            GstAudioInterleave
-            GstAudioMixer
-              GstLiveAdder
           GstMXFMux
           GstVideoAggregator
             GstCompositor
@@ -327,9 +323,6 @@ GObject
       GstGLContext
       GstPad
         GstAggregatorPad
-          GstAudioAggregatorPad
-            GstAudioInterleavePad
-            GstAudioMixerPad
           GstMXFMuxPad
           GstVideoAggregatorPad
             GstCompositorPad
diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am
index ca9e3f7e4..89d8ce743 100644
--- a/gst-libs/gst/audio/Makefile.am
+++ b/gst-libs/gst/audio/Makefile.am
@@ -4,7 +4,6 @@ lib_LTLIBRARIES = libgstbadaudio-@GST_API_VERSION@.la
 CLEANFILES =
 
 libgstbadaudio_@GST_API_VERSION@_la_SOURCES = \
-	gstaudioaggregator.c \
 	gstnonstreamaudiodecoder.c
 
 nodist_libgstbadaudio_@GST_API_VERSION@_la_SOURCES = $(BUILT_SOURCES)
@@ -24,4 +23,4 @@ libgstbadaudio_@GST_API_VERSION@_la_LIBADD = \
 libgstbadaudio_@GST_API_VERSION@_la_LDFLAGS = $(GST_LIB_LDFLAGS) $(GST_ALL_LDFLAGS) $(GST_LT_LDFLAGS)
 
 libgstaudio_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/audio
-libgstaudio_@GST_API_VERSION@include_HEADERS = gstaudioaggregator.h gstnonstreamaudiodecoder.h
+libgstaudio_@GST_API_VERSION@include_HEADERS = gstnonstreamaudiodecoder.h
diff --git a/gst-libs/gst/audio/gstaudioaggregator.c b/gst-libs/gst/audio/gstaudioaggregator.c
deleted file mode 100644
index fa9911b31..000000000
--- a/gst-libs/gst/audio/gstaudioaggregator.c
+++ /dev/null
@@ -1,1995 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2001 Thomas <thomas@apestaart.org>
- *               2005,2006 Wim Taymans <wim@fluendo.com>
- *                    2013 Sebastian Dröge <sebastian@centricular.com>
- *                    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
deleted file mode 100644
index b32630ee6..000000000
--- a/gst-libs/gst/audio/gstaudioaggregator.h
+++ /dev/null
@@ -1,228 +0,0 @@
-/* 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-libs/gst/audio/meson.build b/gst-libs/gst/audio/meson.build
index ac4871903..e32bdf604 100644
--- a/gst-libs/gst/audio/meson.build
+++ b/gst-libs/gst/audio/meson.build
@@ -1,5 +1,5 @@
-badaudio_sources = ['gstaudioaggregator.c', 'gstnonstreamaudiodecoder.c']
-badaudio_headers = ['gstaudioaggregator.h', 'gstnonstreamaudiodecoder.h']
+badaudio_sources = ['gstnonstreamaudiodecoder.c']
+badaudio_headers = ['gstnonstreamaudiodecoder.h']
 install_headers(badaudio_headers, subdir : 'gstreamer-1.0/gst/audio')