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

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

gst-libs/gst/audio/gstaudioaggregator.h

+/* 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__ */

gst/audiomixer/Makefile.am

+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
+

gst/audiomixer/gstaudiointerleave.h

+/* 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__ */

gst/audiomixer/gstaudiomixer.c

+/* 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;
+  }
+}
+