Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567...

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init),
(gst_audio_reverb_class_init), (gst_audio_reverb_init),
(gst_audio_reverb_finalize), (gst_audio_reverb_set_property),
(gst_audio_reverb_get_property), (gst_audio_reverb_setup),
(gst_audio_reverb_stop), (gst_audio_reverb_transform_ip):
* gst/audiofx/audioreverb.h:
* tests/check/Makefile.am:
* tests/check/elements/audioreverb.c: (setup_reverb),
(cleanup_reverb), (GST_START_TEST), (audioreverb_suite):
Add an echo/reverb filter to the audiofx plugin, with configurable
echo delay, intensity and feedback. Fixes bug #567874.

14 files changed, 798 insertions, 3 deletions

diff --git a/ChangeLog b/ChangeLog
index e3f519d30..8533bdbdd 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,28 @@
 2009-01-19  Sebastian Dröge  <sebastian.droege@collabora.co.uk>
 
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* gst/audiofx/Makefile.am:
+	* gst/audiofx/audiofx.c: (plugin_init):
+	* gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init),
+	(gst_audio_reverb_class_init), (gst_audio_reverb_init),
+	(gst_audio_reverb_finalize), (gst_audio_reverb_set_property),
+	(gst_audio_reverb_get_property), (gst_audio_reverb_setup),
+	(gst_audio_reverb_stop), (gst_audio_reverb_transform_ip):
+	* gst/audiofx/audioreverb.h:
+	* tests/check/Makefile.am:
+	* tests/check/elements/audioreverb.c: (setup_reverb),
+	(cleanup_reverb), (GST_START_TEST), (audioreverb_suite):
+	Add an echo/reverb filter to the audiofx plugin, with configurable
+	echo delay, intensity and feedback. Fixes bug #567874.
+
+2009-01-19  Sebastian Dröge  <sebastian.droege@collabora.co.uk>
+
 	* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
 	(gst_spectrum_transform_ip):
 	* gst/spectrum/gstspectrum.h:
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am
index 8549ed3ea..1c03636b3 100644
--- a/docs/plugins/Makefile.am
+++ b/docs/plugins/Makefile.am
@@ -106,6 +106,7 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/alpha/gstalphacolor.h \
 	$(top_srcdir)/gst/apetag/gstapedemux.h \
 	$(top_srcdir)/gst/audiofx/audioamplify.h \
+	$(top_srcdir)/gst/audiofx/audioreverb.h \
 	$(top_srcdir)/gst/audiofx/audiodynamic.h \
 	$(top_srcdir)/gst/audiofx/audioinvert.h \
 	$(top_srcdir)/gst/audiofx/audiokaraoke.h \
diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml
index 8fedb7fa9..231017d7e 100644
--- a/docs/plugins/gst-plugins-good-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml
@@ -23,6 +23,7 @@
     <xi:include href="xml/element-audiowsincband.xml" />
     <xi:include href="xml/element-audiowsinclimit.xml" />
     <xi:include href="xml/element-audiofirfilter.xml" />
+    <xi:include href="xml/element-audioreverb.xml" />
     <xi:include href="xml/element-audiodynamic.xml" />
     <xi:include href="xml/element-audioinvert.xml" />
     <xi:include href="xml/element-audiopanorama.xml" />
diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt
index d2de0b053..03790c5bc 100644
--- a/docs/plugins/gst-plugins-good-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-good-plugins-sections.txt
@@ -117,6 +117,22 @@ gst_audio_iir_filter_get_type
 </SECTION>
 
 <SECTION>
+<FILE>element-audioreverb</FILE>
+<TITLE>audioreverb</TITLE>
+GstAudioReverb
+<SUBSECTION Standard>
+GstAudioReverbClass
+GstAudioReverbProcessFunc
+GST_AUDIO_REVERB
+GST_AUDIO_REVERB_CLASS
+GST_AUDIO_REVERB_GET_CLASS
+GST_IS_AUDIO_REVERB
+GST_IS_AUDIO_REVERB_CLASS
+GST_TYPE_AUDIO_REVERB
+gst_audio_reverb_get_type
+</SECTION>
+
+<SECTION>
 <FILE>element-audiodynamic</FILE>
 <TITLE>audiodynamic</TITLE>
 GstAudioDynamic
diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args
index 0064b6605..7a2603d58 100644
--- a/docs/plugins/gst-plugins-good-plugins.args
+++ b/docs/plugins/gst-plugins-good-plugins.args
@@ -19708,3 +19708,63 @@
 <DEFAULT></DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstAudioDelay::delay</NAME>
+<TYPE>guint64</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Delay</NICK>
+<BLURB>Delay in nanoseconds.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioDelay::feedback</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Feedback</NICK>
+<BLURB>Amount of feedback.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioDelay::intensity</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Intensity</NICK>
+<BLURB>Intensity of the echo.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioReverb::delay</NAME>
+<TYPE>guint64</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Delay</NICK>
+<BLURB>Delay of the echo in nanoseconds.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioReverb::feedback</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Feedback</NICK>
+<BLURB>Amount of feedback.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioReverb::intensity</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Intensity</NICK>
+<BLURB>Intensity of the echo.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
diff --git a/docs/plugins/gst-plugins-good-plugins.hierarchy b/docs/plugins/gst-plugins-good-plugins.hierarchy
index 55e1cdc29..e02a501e2 100644
--- a/docs/plugins/gst-plugins-good-plugins.hierarchy
+++ b/docs/plugins/gst-plugins-good-plugins.hierarchy
@@ -64,6 +64,7 @@ GObject
             GstAudioWSincLimit
             GstAudioWSincBand
             GstAudioFIRFilter
+          GstAudioReverb
           GstIirEqualizer
             GstIirEqualizerNBands
             GstIirEqualizer3Bands
@@ -221,6 +222,8 @@ GObject
     GstRegistry
     GstRingBuffer
   GstSignalObject
+  GstMixerTrack
+    GstMixerOptions
   GstCmmlTagStream
   GstCmmlTagHead
   GstCmmlTagClip
diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml
index 7ca2add6f..171d8b43f 100644
--- a/docs/plugins/inspect/plugin-audiofx.xml
+++ b/docs/plugins/inspect/plugin-audiofx.xml
@@ -199,6 +199,27 @@
       </pads>
     </element>
     <element>
+      <name>audioreverb</name>
+      <longname>Audio reverb</longname>
+      <class>Filter/Effect/Audio</class>
+      <description>Adds an echo or reverb effect to an audio stream</description>
+      <author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
       <name>audiowsincband</name>
       <longname>Band pass &amp; band reject filter</longname>
       <class>Filter/Effect/Audio</class>
diff --git a/docs/plugins/inspect/plugin-spectrum.xml b/docs/plugins/inspect/plugin-spectrum.xml
index deae6a447..e88d99013 100644
--- a/docs/plugins/inspect/plugin-spectrum.xml
+++ b/docs/plugins/inspect/plugin-spectrum.xml
@@ -14,7 +14,7 @@
       <longname>Spectrum analyzer</longname>
       <class>Filter/Analyzer/Audio</class>
       <description>Run an FFT on the audio signal, output spectrum data</description>
-      <author>Erik Walthinsen &lt;omega@cse.ogi.edu&gt;, Stefan Kost &lt;ensonic@users.sf.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
+      <author>Erik Walthinsen &lt;omega@cse.ogi.edu&gt;, Stefan Kost &lt;ensonic@users.sf.net&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
       <pads>
         <caps>
           <name>sink</name>
diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am
index 0ba4f1f34..f4f02be2c 100644
--- a/gst/audiofx/Makefile.am
+++ b/gst/audiofx/Makefile.am
@@ -16,7 +16,8 @@ libgstaudiofx_la_SOURCES = audiofx.c\
 	audiofxbasefirfilter.c \
 	audiowsincband.c \
 	audiowsinclimit.c \
-	audiofirfilter.c
+	audiofirfilter.c \
+	audioreverb.c
 
 # flags used to compile this plugin
 libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@@ -46,5 +47,6 @@ noinst_HEADERS = audiopanorama.h \
 	audiowsincband.h \
         audiowsinclimit.h \
 	audiofirfilter.h \
+	audioreverb.h \
 	math_compat.h
 
diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c
index 62b70761f..e23a638f1 100644
--- a/gst/audiofx/audiofx.c
+++ b/gst/audiofx/audiofx.c
@@ -36,6 +36,7 @@
 #include "audiowsincband.h"
 #include "audiowsinclimit.h"
 #include "audiofirfilter.h"
+#include "audioreverb.h"
 
 /* entry point to initialize the plug-in
  * initialize the plug-in itself
@@ -69,7 +70,9 @@ plugin_init (GstPlugin * plugin)
       gst_element_register (plugin, "audiowsincband", GST_RANK_NONE,
           GST_TYPE_AUDIO_WSINC_BAND) &&
       gst_element_register (plugin, "audiofirfilter", GST_RANK_NONE,
-          GST_TYPE_AUDIO_FIR_FILTER));
+          GST_TYPE_AUDIO_FIR_FILTER) &&
+      gst_element_register (plugin, "audioreverb", GST_RANK_NONE,
+          GST_TYPE_AUDIO_REVERB));
 }
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
diff --git a/gst/audiofx/audioreverb.c b/gst/audiofx/audioreverb.c
new file mode 100644
index 000000000..703a4ef0b
--- /dev/null
+++ b/gst/audiofx/audioreverb.c
@@ -0,0 +1,367 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-audioreverb
+ *
+ * <refsect2>
+ * audioreverb adds an echo or revert effect to an audio stream. The echo
+ * reverb, intensity and the percentage of feedback can be configured.
+ * <para>
+ * <programlisting>
+ * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioreverb reverb=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
+ * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioreverb reverb=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.12
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include "audioreverb.h"
+
+#define GST_CAT_DEFAULT gst_audio_reverb_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+  PROP_0,
+  PROP_DELAY,
+  PROP_INTENSITY,
+  PROP_FEEDBACK
+};
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-float,"                                              \
+    " width=(int) { 32, 64 }, "                                       \
+    " endianness=(int)BYTE_ORDER,"                                    \
+    " rate=(int)[1,MAX],"                                             \
+    " channels=(int)[1,MAX]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_reverb_debug, "audioreverb", 0, "audioreverb element");
+
+GST_BOILERPLATE_FULL (GstAudioReverb, gst_audio_reverb, GstAudioFilter,
+    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_reverb_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_audio_reverb_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_audio_reverb_finalize (GObject * object);
+
+static gboolean gst_audio_reverb_setup (GstAudioFilter * self,
+    GstRingBufferSpec * format);
+static gboolean gst_audio_reverb_stop (GstBaseTransform * base);
+static GstFlowReturn gst_audio_reverb_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+
+static void gst_audio_reverb_transform_float (GstAudioReverb * self,
+    gfloat * data, guint num_samples);
+static void gst_audio_reverb_transform_double (GstAudioReverb * self,
+    gdouble * data, guint num_samples);
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_reverb_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstCaps *caps;
+
+  gst_element_class_set_details_simple (element_class, "Audio reverb",
+      "Filter/Effect/Audio",
+      "Adds an echo or reverb effect to an audio stream",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+}
+
+static void
+gst_audio_reverb_class_init (GstAudioReverbClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
+  GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->set_property = gst_audio_reverb_set_property;
+  gobject_class->get_property = gst_audio_reverb_get_property;
+  gobject_class->finalize = gst_audio_reverb_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_DELAY,
+      g_param_spec_uint64 ("delay", "Delay",
+          "Delay of the echo in nanoseconds", 1, G_MAXUINT64,
+          1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+          | GST_PARAM_CONTROLLABLE));
+
+  g_object_class_install_property (gobject_class, PROP_INTENSITY,
+      g_param_spec_float ("intensity", "Intensity",
+          "Intensity of the echo", 0.0, 1.0,
+          0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+          | GST_PARAM_CONTROLLABLE));
+
+  g_object_class_install_property (gobject_class, PROP_FEEDBACK,
+      g_param_spec_float ("feedback", "Feedback",
+          "Amount of feedback", 0.0, 1.0,
+          0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+          | GST_PARAM_CONTROLLABLE));
+
+  audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_reverb_setup);
+  basetransform_class->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_reverb_transform_ip);
+  basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_reverb_stop);
+}
+
+static void
+gst_audio_reverb_init (GstAudioReverb * self, GstAudioReverbClass * klass)
+{
+  self->delay = 0;
+  self->intensity = 0.0;
+  self->feedback = 0.0;
+
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
+}
+
+static void
+gst_audio_reverb_finalize (GObject * object)
+{
+  GstAudioReverb *self = GST_AUDIO_REVERB (object);
+
+  g_free (self->buffer);
+  self->buffer = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_audio_reverb_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioReverb *self = GST_AUDIO_REVERB (object);
+
+  switch (prop_id) {
+    case PROP_DELAY:{
+      guint rate, width, channels;
+
+      GST_BASE_TRANSFORM_LOCK (self);
+      self->delay = g_value_get_uint64 (value);
+
+      rate = GST_AUDIO_FILTER (self)->format.rate;
+      width = GST_AUDIO_FILTER (self)->format.width / 8;
+      channels = GST_AUDIO_FILTER (self)->format.channels;
+
+      if (self->buffer && rate > 0) {
+        guint new_reverb =
+            MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
+        guint new_size = new_reverb * width * channels;
+
+        if (new_size > self->buffer_size) {
+          guint i;
+          guint8 *old_buffer = self->buffer;
+
+          self->buffer_size = new_size;
+          self->buffer = g_malloc0 (new_size);
+
+          for (i = 0; i < self->buffer_size_frames; i++) {
+            memcpy (&self->buffer[i * width * channels],
+                &old_buffer[((i +
+                            self->buffer_pos) % self->buffer_size_frames) *
+                    width * channels], channels * width);
+          }
+          self->buffer_size_frames = self->delay_frames = new_reverb;
+          self->buffer_pos = 0;
+        }
+      } else if (self->buffer) {
+        g_free (self->buffer);
+        self->buffer = NULL;
+      }
+
+      GST_BASE_TRANSFORM_UNLOCK (self);
+    }
+      break;
+    case PROP_INTENSITY:{
+      GST_BASE_TRANSFORM_LOCK (self);
+      self->intensity = g_value_get_float (value);
+      GST_BASE_TRANSFORM_UNLOCK (self);
+    }
+      break;
+    case PROP_FEEDBACK:{
+      GST_BASE_TRANSFORM_LOCK (self);
+      self->feedback = g_value_get_float (value);
+      GST_BASE_TRANSFORM_UNLOCK (self);
+    }
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_reverb_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioReverb *self = GST_AUDIO_REVERB (object);
+
+  switch (prop_id) {
+    case PROP_DELAY:
+      GST_BASE_TRANSFORM_LOCK (self);
+      g_value_set_uint64 (value, self->delay);
+      GST_BASE_TRANSFORM_UNLOCK (self);
+      break;
+    case PROP_INTENSITY:
+      GST_BASE_TRANSFORM_LOCK (self);
+      g_value_set_float (value, self->intensity);
+      GST_BASE_TRANSFORM_UNLOCK (self);
+      break;
+    case PROP_FEEDBACK:
+      GST_BASE_TRANSFORM_LOCK (self);
+      g_value_set_float (value, self->feedback);
+      GST_BASE_TRANSFORM_UNLOCK (self);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_reverb_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+{
+  GstAudioReverb *self = GST_AUDIO_REVERB (base);
+  gboolean ret = TRUE;
+
+  if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
+    self->process = (GstAudioReverbProcessFunc)
+        gst_audio_reverb_transform_float;
+  else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64)
+    self->process = (GstAudioReverbProcessFunc)
+        gst_audio_reverb_transform_double;
+  else
+    ret = FALSE;
+
+  g_free (self->buffer);
+  self->buffer = NULL;
+  self->buffer_pos = 0;
+  self->buffer_size = 0;
+  self->buffer_size_frames = 0;
+
+  return ret;
+}
+
+static gboolean
+gst_audio_reverb_stop (GstBaseTransform * base)
+{
+  GstAudioReverb *self = GST_AUDIO_REVERB (base);
+
+  g_free (self->buffer);
+  self->buffer = NULL;
+  self->buffer_pos = 0;
+  self->buffer_size = 0;
+  self->buffer_size_frames = 0;
+
+  return TRUE;
+}
+
+#define TRANSFORM_FUNC(name, type) \
+static void \
+gst_audio_reverb_transform_##name (GstAudioReverb * self, \
+    type * data, guint num_samples) \
+{ \
+  type *buffer = (type *) self->buffer; \
+  guint channels = GST_AUDIO_FILTER (self)->format.channels; \
+  guint rate = GST_AUDIO_FILTER (self)->format.rate; \
+  guint i, j; \
+  guint reverb_index = self->buffer_size_frames - self->delay_frames; \
+  gdouble reverb_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
+  \
+  if (reverb_off < 0.0) \
+    reverb_off = 0.0; \
+  \
+  num_samples /= channels; \
+  \
+  for (i = 0; i < num_samples; i++) { \
+    guint echo0_index = ((reverb_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
+    guint echo1_index = ((reverb_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
+    guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
+    for (j = 0; j < channels; j++) { \
+      gdouble in = data[i*channels + j]; \
+      gdouble echo0 = buffer[echo0_index + j]; \
+      gdouble echo1 = buffer[echo1_index + j]; \
+      gdouble echo = echo0 + (echo1-echo0)*reverb_off; \
+      type out = in + self->intensity * echo; \
+      \
+      data[i*channels + j] = out; \
+      \
+      buffer[rbout_index + j] = in + self->feedback * echo; \
+    } \
+    self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
+  } \
+}
+
+TRANSFORM_FUNC (float, gfloat);
+TRANSFORM_FUNC (double, gdouble);
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_reverb_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+  GstAudioReverb *self = GST_AUDIO_REVERB (base);
+  guint num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+    gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf));
+
+  if (self->buffer == NULL) {
+    guint width, rate, channels;
+
+    width = GST_AUDIO_FILTER (self)->format.width / 8;
+    rate = GST_AUDIO_FILTER (self)->format.rate;
+    channels = GST_AUDIO_FILTER (self)->format.channels;
+
+    self->delay_frames =
+        MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
+
+    self->buffer_size_frames = MAX (self->delay_frames, 1000);
+    self->buffer_size = self->buffer_size_frames * width * channels;
+    self->buffer = g_malloc0 (self->buffer_size);
+    self->buffer_pos = 0;
+  }
+
+  self->process (self, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}
diff --git a/gst/audiofx/audioreverb.h b/gst/audiofx/audioreverb.h
new file mode 100644
index 000000000..3ef5682e8
--- /dev/null
+++ b/gst/audiofx/audioreverb.h
@@ -0,0 +1,68 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_AUDIO_REVERB_H__
+#define __GST_AUDIO_REVERB_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AUDIO_REVERB            (gst_audio_reverb_get_type())
+#define GST_AUDIO_REVERB(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_REVERB,GstAudioReverb))
+#define GST_IS_AUDIO_REVERB(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_REVERB))
+#define GST_AUDIO_REVERB_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_REVERB,GstAudioReverbClass))
+#define GST_IS_AUDIO_REVERB_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_REVERB))
+#define GST_AUDIO_REVERB_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_REVERB,GstAudioReverbClass))
+typedef struct _GstAudioReverb GstAudioReverb;
+typedef struct _GstAudioReverbClass GstAudioReverbClass;
+
+typedef void (*GstAudioReverbProcessFunc) (GstAudioReverb *, guint8 *, guint);
+
+struct _GstAudioReverb
+{
+  GstAudioFilter audiofilter;
+
+  guint64 delay;
+  gfloat intensity;
+  gfloat feedback;
+
+  /* < private > */
+  GstAudioReverbProcessFunc process;
+  guint delay_frames;
+  guint8 *buffer;
+  guint buffer_pos;
+  guint buffer_size;
+  guint buffer_size_frames;
+};
+
+struct _GstAudioReverbClass
+{
+  GstAudioFilterClass parent;
+};
+
+GType gst_audio_reverb_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_REVERB_H__ */
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 5cc63dbb9..ee0afdc91 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -74,6 +74,7 @@ check_PROGRAMS = \
 	elements/audiocheblimit \
 	elements/audioiirfilter \
 	elements/audioamplify \
+	elements/audioreverb \
 	elements/audiodynamic \
 	elements/audiowsincband \
 	elements/audiowsinclimit \
diff --git a/tests/check/elements/audioreverb.c b/tests/check/elements/audioreverb.c
new file mode 100644
index 000000000..cafe8bbe9
--- /dev/null
+++ b/tests/check/elements/audioreverb.c
@@ -0,0 +1,229 @@
+/* GStreamer
+ *
+ * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+gboolean have_eos = FALSE;
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+GstPad *mysrcpad, *mysinkpad;
+
+#define REVERB_CAPS_STRING    \
+    "audio/x-raw-float, "               \
+    "channels = (int) 2, "              \
+    "rate = (int) 100000, "             \
+    "endianness = (int) BYTE_ORDER, "   \
+    "width = (int) 64"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-float, "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 1,  MAX ], "
+        "endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 }")
+    );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-float, "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 1,  MAX ], "
+        "endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 }")
+    );
+
+GstElement *
+setup_reverb ()
+{
+  GstElement *reverb;
+
+  GST_DEBUG ("setup_reverb");
+  reverb = gst_check_setup_element ("audioreverb");
+  mysrcpad = gst_check_setup_src_pad (reverb, &srctemplate, NULL);
+  mysinkpad = gst_check_setup_sink_pad (reverb, &sinktemplate, NULL);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  return reverb;
+}
+
+void
+cleanup_reverb (GstElement * reverb)
+{
+  GST_DEBUG ("cleanup_reverb");
+
+  g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (buffers);
+  buffers = NULL;
+
+  gst_pad_set_active (mysrcpad, FALSE);
+  gst_pad_set_active (mysinkpad, FALSE);
+  gst_check_teardown_src_pad (reverb);
+  gst_check_teardown_sink_pad (reverb);
+  gst_check_teardown_element (reverb);
+}
+
+GST_START_TEST (test_passthrough)
+{
+  GstElement *reverb;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gdouble in[] = { 1.0, -1.0, 0.0, 0.5, -0.5, 0.0 };
+  gdouble *res;
+
+  reverb = setup_reverb ();
+  g_object_set (G_OBJECT (reverb), "delay", 1, "intensity", 0.0, "feedback",
+      0.0, NULL);
+  fail_unless (gst_element_set_state (reverb,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (sizeof (in));
+  memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
+  caps = gst_caps_from_string (REVERB_CAPS_STRING);
+  gst_buffer_set_caps (inbuffer, caps);
+  gst_caps_unref (caps);
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  /* pushing gives away my reference ... */
+  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+  /* ... but it ends up being collected on the global buffer list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf",
+      in[0], in[1], in[2], in[3], in[4], in[5], res[0], res[1], res[2], res[3],
+      res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), in, sizeof (in)) == 0);
+
+  /* cleanup */
+  cleanup_reverb (reverb);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reverb)
+{
+  GstElement *reverb;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gdouble in[] = { 1.0, -1.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, };
+  gdouble out[] = { 1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 0.0, 0.0, 0.0 };
+  gdouble *res;
+
+  reverb = setup_reverb ();
+  g_object_set (G_OBJECT (reverb), "delay", 20000, "intensity", 1.0, "feedback",
+      0.0, NULL);
+  fail_unless (gst_element_set_state (reverb,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (sizeof (in));
+  memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
+  caps = gst_caps_from_string (REVERB_CAPS_STRING);
+  gst_buffer_set_caps (inbuffer, caps);
+  gst_caps_unref (caps);
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  /* pushing gives away my reference ... */
+  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+  /* ... but it ends up being collected on the global buffer list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf",
+      out[0], out[1], out[2], out[3], out[4], out[5], out[6], out[7], out[8],
+      out[9], res[0], res[1], res[2], res[3], res[4], res[5], res[6], res[7],
+      res[8], res[9]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, sizeof (out)) == 0);
+
+  /* cleanup */
+  cleanup_reverb (reverb);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_feedback)
+{
+  GstElement *reverb;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gdouble in[] = { 1.0, -1.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, };
+  gdouble out[] = { 1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 0.0, 1.0, -1.0 };
+  gdouble *res;
+
+  reverb = setup_reverb ();
+  g_object_set (G_OBJECT (reverb), "delay", 20000, "intensity", 1.0, "feedback",
+      1.0, NULL);
+  fail_unless (gst_element_set_state (reverb,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (sizeof (in));
+  memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
+  caps = gst_caps_from_string (REVERB_CAPS_STRING);
+  gst_buffer_set_caps (inbuffer, caps);
+  gst_caps_unref (caps);
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  /* pushing gives away my reference ... */
+  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+  /* ... but it ends up being collected on the global buffer list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf",
+      out[0], out[1], out[2], out[3], out[4], out[5], out[6], out[7], out[8],
+      out[9], res[0], res[1], res[2], res[3], res[4], res[5], res[6], res[7],
+      res[8], res[9]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, sizeof (out)) == 0);
+
+  /* cleanup */
+  cleanup_reverb (reverb);
+}
+
+GST_END_TEST;
+
+static Suite *
+audioreverb_suite (void)
+{
+  Suite *s = suite_create ("audioreverb");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_passthrough);
+  tcase_add_test (tc_chain, test_reverb);
+  tcase_add_test (tc_chain, test_feedback);
+
+  return s;
+}
+
+GST_CHECK_MAIN (audioreverb);