gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme...

Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes #397162
* 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/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify

12 files changed, 1065 insertions, 3 deletions

diff --git a/ChangeLog b/ChangeLog
index c04ba488..0f8a4038 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,39 @@
+2007-01-24  Sebastian Dröge  <slomo@circular-chaos.org>
+
+	reviewed by: Stefan Kost  <ensonic@users.sf.net>
+
+	* gst/audiofx/Makefile.am:
+	* gst/audiofx/audioamplify.c:
+	(gst_audio_amplify_clipping_method_get_type),
+	(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
+	(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
+	(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
+	(gst_audio_amplify_set_caps),
+	(gst_audio_amplify_transform_int_clip),
+	(gst_audio_amplify_transform_int_wrap_negative),
+	(gst_audio_amplify_transform_int_wrap_positive),
+	(gst_audio_amplify_transform_float_clip),
+	(gst_audio_amplify_transform_float_wrap_negative),
+	(gst_audio_amplify_transform_float_wrap_positive),
+	(gst_audio_amplify_transform_ip):
+	* gst/audiofx/audioamplify.h:
+	* gst/audiofx/audiofx.c: (plugin_init):
+	Add new element "audioamplify". This allows scaling of raw audio
+	samples, similar to the "volume" element, but provides different modes
+	for clipping and allows unlimited amplification. It's mainly targeted
+	for creative sound design and not as a replacement of the "volume"
+	element. Fixes #397162
+	* 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/inspect/plugin-audiofx.xml:
+	Add docs for audioamplify and integrate them into the build system
+	* tests/check/Makefile.am:
+	* tests/check/elements/audioamplify.c: (setup_amplify),
+	(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
+	Add fairly extensive unit test suite for audioamplify
+
 2007-01-24  Wim Taymans  <wim@fluendo.com>
 
 	* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am
index e7746e30..2a758478 100644
--- a/docs/plugins/Makefile.am
+++ b/docs/plugins/Makefile.am
@@ -77,6 +77,7 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/apetag/gstapedemux.h \
 	$(top_srcdir)/gst/audiofx/audiopanorama.h \
 	$(top_srcdir)/gst/audiofx/audioinvert.h \
+	$(top_srcdir)/gst/audiofx/audioamplify.h \
 	$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
 	$(top_srcdir)/gst/autodetect/gstautovideosink.h \
 	$(top_srcdir)/gst/avi/gstavidemux.h \
diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml
index 5f7af731..39b97578 100644
--- a/docs/plugins/gst-plugins-good-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml
@@ -16,6 +16,7 @@
     <xi:include href="xml/element-apev2mux.xml" />
     <xi:include href="xml/element-audiopanorama.xml" />
     <xi:include href="xml/element-audioinvert.xml" />
+    <xi:include href="xml/element-audioamplify.xml" />
     <xi:include href="xml/element-autoaudiosink.xml" />
     <xi:include href="xml/element-autovideosink.xml" />
     <xi:include href="xml/element-avidemux.xml" />
diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt
index 9da8dacf..bb8d1331 100644
--- a/docs/plugins/gst-plugins-good-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-good-plugins-sections.txt
@@ -48,6 +48,16 @@ GST_AUDIO_INVERT_CLASS
 </SECTION>
 
 <SECTION>
+<FILE>element-audioamplify</FILE>
+GstAudioAmplify
+<TITLE>audioamplify</TITLE>
+<SUBSECTION Standard>
+GstAudioAmplifyClass
+GST_AUDIO_AMPLIFY
+GST_AUDIO_AMPLIFY_CLASS
+</SECTION>
+
+<SECTION>
 <FILE>element-autoaudiosink</FILE>
 GstAutoAudioSink
 <TITLE>autoaudiosink</TITLE>
diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args
index 19bc7f5d..19537ee2 100644
--- a/docs/plugins/gst-plugins-good-plugins.args
+++ b/docs/plugins/gst-plugins-good-plugins.args
@@ -16648,3 +16648,23 @@
 <DEFAULT>0</DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstAudioAmplify::amplification</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Amplification</NICK>
+<BLURB>Factor of amplification.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioAmplify::clipping-method</NAME>
+<TYPE>GstAudioPanoramaClippingMethod</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Clipping method</NICK>
+<BLURB>Selects how to handle values higher than the maximum.</BLURB>
+<DEFAULT>Normal Clipping (default)</DEFAULT>
+</ARG>
+
diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml
index 54b4926c..f860bd2b 100644
--- a/docs/plugins/inspect/plugin-audiofx.xml
+++ b/docs/plugins/inspect/plugin-audiofx.xml
@@ -10,6 +10,13 @@
   <origin>http://gstreamer.net/</origin>
   <elements>
     <element>
+      <name>audioamplify</name>
+      <longname>AudioAmplify</longname>
+      <class>Filter/Effect/Audio</class>
+      <description>Amplifies an audio stream by a given factor</description>
+      <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
+    </element>
+    <element>
       <name>audioinvert</name>
       <longname>AudioInvert</longname>
       <class>Filter/Effect/Audio</class>
diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am
index 0b8be7a9..b3ac0b1a 100644
--- a/gst/audiofx/Makefile.am
+++ b/gst/audiofx/Makefile.am
@@ -5,7 +5,8 @@ plugin_LTLIBRARIES = libgstaudiofx.la
 # sources used to compile this plug-in
 libgstaudiofx_la_SOURCES = audiofx.c\
 	audiopanorama.c \
-	audioinvert.c
+	audioinvert.c \
+	audioamplify.c
 
 # flags used to compile this plugin
 libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@@ -18,4 +19,5 @@ libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 
 # headers we need but don't want installed
 noinst_HEADERS = audiopanorama.h \
-	audioinvert.h
+	audioinvert.h \
+	audioamplify.h
diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c
new file mode 100644
index 00000000..076aa431
--- /dev/null
+++ b/gst/audiofx/audioamplify.c
@@ -0,0 +1,451 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
+ *
+ * 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-audioamplify
+ * @short_description: Amplifies an audio stream with selectable clipping mode
+ *
+ * <refsect2>
+ * Amplifies an audio stream by a given factor and allows the selection of different clipping modes.
+ * The difference between the clipping modes is best evaluated by testing.
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink
+ * gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/controller/gstcontroller.h>
+
+#include "audioamplify.h"
+
+#define GST_CAT_DEFAULT gst_audio_amplify_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioAmplify",
+    "Filter/Effect/Audio",
+    "Amplifies an audio stream by a given factor",
+    "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_AMPLIFICATION,
+  PROP_CLIPPING_METHOD
+};
+
+enum
+{
+  METHOD_CLIP = 0,
+  METHOD_WRAP_NEGATIVE,
+  METHOD_WRAP_POSITIVE,
+  NUM_METHODS
+};
+
+#define GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD (gst_audio_amplify_clipping_method_get_type ())
+static GType
+gst_audio_amplify_clipping_method_get_type (void)
+{
+  static GType gtype = 0;
+
+  if (gtype == 0) {
+    static const GEnumValue values[] = {
+      {METHOD_CLIP, "Normal Clipping (default)", "clip"},
+      {METHOD_WRAP_NEGATIVE,
+            "Push overdriven values back from the opposite side",
+          "wrap-negative"},
+      {METHOD_WRAP_POSITIVE, "Push overdriven values back from the same side",
+          "wrap-positive"},
+      {0, NULL, NULL}
+    };
+
+    gtype = g_enum_register_static ("GstAudioPanoramaClippingMethod", values);
+  }
+  return gtype;
+}
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-float, "
+        "rate = (int) [ 1, MAX ], "
+        "channels = (int) [ 1, MAX ], "
+        "endianness = (int) BYTE_ORDER, " "width = (int) 32; "
+        "audio/x-raw-int, "
+        "rate = (int) [ 1, MAX ], "
+        "channels = (int) [ 1, MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
+    );
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-float, "
+        "rate = (int) [ 1, MAX ], "
+        "channels = (int) [ 1, MAX], "
+        "endianness = (int) BYTE_ORDER, " "width = (int) 32; "
+        "audio/x-raw-int, "
+        "rate = (int) [ 1, MAX ], "
+        "channels = (int) [ 1, MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
+    );
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0, "audioamplify element");
+
+GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstBaseTransform,
+    GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
+
+static void gst_audio_amplify_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_audio_amplify_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_amplify_set_caps (GstBaseTransform * base,
+    GstCaps * incaps, GstCaps * outcaps);
+static GstFlowReturn gst_audio_amplify_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+
+static void gst_audio_amplify_transform_int_clip (GstAudioAmplify * filter,
+    gint16 * data, guint num_samples);
+static void gst_audio_amplify_transform_int_wrap_negative (GstAudioAmplify *
+    filter, gint16 * data, guint num_samples);
+static void gst_audio_amplify_transform_int_wrap_positive (GstAudioAmplify *
+    filter, gint16 * data, guint num_samples);
+static void gst_audio_amplify_transform_float_clip (GstAudioAmplify * filter,
+    gfloat * data, guint num_samples);
+static void gst_audio_amplify_transform_float_wrap_negative (GstAudioAmplify *
+    filter, gfloat * data, guint num_samples);
+static void gst_audio_amplify_transform_float_wrap_positive (GstAudioAmplify *
+    filter, gfloat * data, guint num_samples);
+
+/* table of processing functions: [format][clipping_method] */
+static GstAudioAmplifyProcessFunc processing_functions[2][3] = {
+  {
+        (GstAudioAmplifyProcessFunc) gst_audio_amplify_transform_int_clip,
+        (GstAudioAmplifyProcessFunc)
+        gst_audio_amplify_transform_int_wrap_negative,
+      (GstAudioAmplifyProcessFunc)
+        gst_audio_amplify_transform_int_wrap_positive},
+  {
+        (GstAudioAmplifyProcessFunc) gst_audio_amplify_transform_float_clip,
+        (GstAudioAmplifyProcessFunc)
+        gst_audio_amplify_transform_float_wrap_negative,
+      (GstAudioAmplifyProcessFunc)
+        gst_audio_amplify_transform_float_wrap_positive}
+};
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_amplify_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_set_details (element_class, &element_details);
+}
+
+static void
+gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gobject_class->set_property = gst_audio_amplify_set_property;
+  gobject_class->get_property = gst_audio_amplify_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_AMPLIFICATION,
+      g_param_spec_float ("amplification", "Amplification",
+          "Factor of amplification", 0.0, G_MAXFLOAT,
+          1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+  /**
+   * GstAudioAmplify:clipping-method
+   *
+   * Clipping method: clip mode set values higher than the maximum to the
+   * maximum. The wrap-negative mode pushes those values back from the
+   * opposite side, wrap-positive pushes them back from the same side.
+   *
+   **/
+  g_object_class_install_property (gobject_class, PROP_CLIPPING_METHOD,
+      g_param_spec_enum ("clipping-method", "Clipping method",
+          "Selects how to handle values higher than the maximum",
+          GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD, METHOD_CLIP,
+          G_PARAM_READWRITE));
+
+  GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
+      GST_DEBUG_FUNCPTR (gst_audio_amplify_set_caps);
+  GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_amplify_transform_ip);
+}
+
+static void
+gst_audio_amplify_init (GstAudioAmplify * filter, GstAudioAmplifyClass * klass)
+{
+  filter->amplification = 1.0;
+  filter->clipping_method = METHOD_CLIP;
+  filter->width = 0;
+  filter->format_float = FALSE;
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+}
+
+static gboolean
+gst_audio_amplify_set_process_function (GstAudioAmplify * filter)
+{
+  gint format_index, method_index;
+
+  /* set processing function */
+
+  format_index = (filter->format_float) ? 1 : 0;
+
+  method_index = filter->clipping_method;
+  if (method_index >= NUM_METHODS || method_index < 0)
+    method_index = METHOD_CLIP;
+
+  filter->process = processing_functions[format_index][method_index];
+  return TRUE;
+}
+
+static void
+gst_audio_amplify_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
+
+  switch (prop_id) {
+    case PROP_AMPLIFICATION:
+      filter->amplification = g_value_get_float (value);
+      gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
+          filter->amplification == 1.0);
+      break;
+    case PROP_CLIPPING_METHOD:
+      filter->clipping_method = g_value_get_enum (value);
+      gst_audio_amplify_set_process_function (filter);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_amplify_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
+
+  switch (prop_id) {
+    case PROP_AMPLIFICATION:
+      g_value_set_float (value, filter->amplification);
+      break;
+    case PROP_CLIPPING_METHOD:
+      g_value_set_enum (value, filter->clipping_method);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+/* GstBaseTransform vmethod implementations */
+
+static gboolean
+gst_audio_amplify_set_caps (GstBaseTransform * base, GstCaps * incaps,
+    GstCaps * outcaps)
+{
+  GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
+  const GstStructure *structure;
+  gboolean ret;
+  gint width;
+  const gchar *fmt;
+
+  /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
+
+  structure = gst_caps_get_structure (incaps, 0);
+
+  ret = gst_structure_get_int (structure, "width", &width);
+  if (!ret)
+    goto no_width;
+  filter->width = width / 8;
+
+
+  fmt = gst_structure_get_name (structure);
+  if (!strcmp (fmt, "audio/x-raw-int"))
+    filter->format_float = FALSE;
+  else
+    filter->format_float = TRUE;
+
+  GST_DEBUG ("try to process %s input", fmt);
+  ret = gst_audio_amplify_set_process_function (filter);
+  if (!ret)
+    GST_WARNING ("can't process input");
+
+  return TRUE;
+
+no_width:
+  GST_DEBUG ("no width in caps");
+  return FALSE;
+}
+
+static void
+gst_audio_amplify_transform_int_clip (GstAudioAmplify * filter,
+    gint16 * data, guint num_samples)
+{
+  gint i;
+  glong val;
+
+  for (i = 0; i < num_samples; i++) {
+    val = (*data) * filter->amplification;
+    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
+  }
+}
+
+static void
+gst_audio_amplify_transform_int_wrap_negative (GstAudioAmplify * filter,
+    gint16 * data, guint num_samples)
+{
+  gint i;
+  glong val;
+
+  for (i = 0; i < num_samples; i++) {
+    val = (*data) * filter->amplification;
+    if (val > G_MAXINT16)
+      val = ((val - G_MININT16) & 0xffff) + G_MININT16;
+    else if (val < G_MININT16)
+      val = ((val - G_MAXINT16) & 0xffff) + G_MAXINT16;
+    *data++ = val;
+  }
+}
+
+static void
+gst_audio_amplify_transform_int_wrap_positive (GstAudioAmplify * filter,
+    gint16 * data, guint num_samples)
+{
+  gint i;
+  glong val;
+
+  for (i = 0; i < num_samples; i++) {
+    val = (*data) * filter->amplification;
+    while (val > G_MAXINT16 || val < G_MININT16) {
+      if (val > G_MAXINT16)
+        val = G_MAXINT16 - (val - G_MAXINT16);
+      else if (val < G_MININT16)
+        val = G_MININT16 - (val - G_MININT16);
+    }
+    *data++ = val;
+  }
+}
+
+static void
+gst_audio_amplify_transform_float_clip (GstAudioAmplify * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i;
+  gfloat val;
+
+  for (i = 0; i < num_samples; i++) {
+    val = (*data) * filter->amplification;
+    if (val > 1.0)
+      val = 1.0;
+    else if (val < -1.0)
+      val = -1.0;
+
+    *data++ = val;
+  }
+}
+
+static void
+gst_audio_amplify_transform_float_wrap_negative (GstAudioAmplify * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i;
+  gfloat val;
+
+  for (i = 0; i < num_samples; i++) {
+    val = (*data) * filter->amplification;
+    while (val > 1.0 || val < -1.0) {
+      if (val > 1.0)
+        val = -1.0 + (val - 1.0);
+      else if (val < -1.0)
+        val = 1.0 + (val + 1.0);
+    }
+    *data++ = val;
+  }
+}
+
+static void
+gst_audio_amplify_transform_float_wrap_positive (GstAudioAmplify * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i;
+  gfloat val;
+
+  for (i = 0; i < num_samples; i++) {
+    val = (*data) * filter->amplification;
+    while (val > 1.0 || val < -1.0) {
+      if (val > 1.0)
+        val = 1.0 - (val - 1.0);
+      else if (val < -1.0)
+        val = -1.0 - (val + 1.0);
+    }
+    *data++ = val;
+  }
+}
+
+/* this function does the actual processing
+ */
+static GstFlowReturn
+gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+  GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
+  guint num_samples = GST_BUFFER_SIZE (buf) / filter->width;
+
+  if (!gst_buffer_is_writable (buf))
+    return GST_FLOW_OK;
+
+  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}
diff --git a/gst/audiofx/audioamplify.h b/gst/audiofx/audioamplify.h
new file mode 100644
index 00000000..5b8de443
--- /dev/null
+++ b/gst/audiofx/audioamplify.h
@@ -0,0 +1,61 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
+ *
+ * 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_AMPLIFY_H__
+#define __GST_AUDIO_AMPLIFY_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_AMPLIFY            (gst_audio_amplify_get_type())
+#define GST_AUDIO_AMPLIFY(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplify))
+#define GST_IS_AUDIO_AMPLIFY(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AMPLIFY))
+#define GST_AUDIO_AMPLIFY_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplifyClass))
+#define GST_IS_AUDIO_AMPLIFY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_AMPLIFY))
+#define GST_AUDIO_AMPLIFY_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplifyClass))
+typedef struct _GstAudioAmplify GstAudioAmplify;
+typedef struct _GstAudioAmplifyClass GstAudioAmplifyClass;
+
+typedef void (*GstAudioAmplifyProcessFunc) (GstAudioAmplify *, guint8 *, guint);
+
+struct _GstAudioAmplify
+{
+  GstBaseTransform element;
+
+  gfloat amplification;
+
+  /* < private > */
+  GstAudioAmplifyProcessFunc process;
+  gint clipping_method;
+  gint width;
+  gboolean format_float;
+};
+
+struct _GstAudioAmplifyClass
+{
+  GstBaseTransformClass parent;
+};
+
+GType gst_audio_amplify_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_AMPLIFY_H__ */
diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c
index 541aa143..1ee1da66 100644
--- a/gst/audiofx/audiofx.c
+++ b/gst/audiofx/audiofx.c
@@ -27,6 +27,7 @@
 
 #include "audiopanorama.h"
 #include "audioinvert.h"
+#include "audioamplify.h"
 
 /* entry point to initialize the plug-in
  * initialize the plug-in itself
@@ -42,7 +43,9 @@ plugin_init (GstPlugin * plugin)
   return (gst_element_register (plugin, "audiopanorama", GST_RANK_NONE,
           GST_TYPE_AUDIO_PANORAMA) &&
       gst_element_register (plugin, "audioinvert", GST_RANK_NONE,
-          GST_TYPE_AUDIO_INVERT));
+          GST_TYPE_AUDIO_INVERT) &&
+      gst_element_register (plugin, "audioamplify", GST_RANK_NONE,
+          GST_TYPE_AUDIO_AMPLIFY));
 }
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 9b6fac23..53121d87 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -35,6 +35,7 @@ check_PROGRAMS = \
 	$(check_annodex) \
 	elements/audiopanorama \
 	elements/audioinvert \
+	elements/audioamplify \
 	elements/avimux \
 	elements/level \
 	elements/matroskamux \
diff --git a/tests/check/elements/audioamplify.c b/tests/check/elements/audioamplify.c
new file mode 100644
index 00000000..a1899c3a
--- /dev/null
+++ b/tests/check/elements/audioamplify.c
@@ -0,0 +1,469 @@
+/* GStreamer
+ *
+ * unit test for audioamplify
+ *
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * Greatly based on the audiopanorama unit test
+ * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
+ *
+ * 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 <unistd.h>
+
+#include <gst/base/gstbasetransform.h>
+#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 AMPLIFY_CAPS_STRING    \
+    "audio/x-raw-int, "                 \
+    "channels = (int) 1, "              \
+    "rate = (int) 44100, "              \
+    "endianness = (int) BYTE_ORDER, "   \
+    "width = (int) 16, "                \
+    "depth = (int) 16, "                \
+    "signed = (bool) TRUE"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "channels = (int) 1, "
+        "rate = (int) [ 1,  MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")
+    );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "channels = (int) 1, "
+        "rate = (int) [ 1,  MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")
+    );
+
+GstElement *
+setup_amplify ()
+{
+  GstElement *amplify;
+
+  GST_DEBUG ("setup_amplify");
+  amplify = gst_check_setup_element ("audioamplify");
+  mysrcpad = gst_check_setup_src_pad (amplify, &srctemplate, NULL);
+  mysinkpad = gst_check_setup_sink_pad (amplify, &sinktemplate, NULL);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  return amplify;
+}
+
+void
+cleanup_amplify (GstElement * amplify)
+{
+  GST_DEBUG ("cleanup_amplify");
+
+  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 (amplify);
+  gst_check_teardown_sink_pad (amplify);
+  gst_check_teardown_element (amplify);
+}
+
+GST_START_TEST (test_passthrough)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      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, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_zero)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { 0, 0, 0, 0, 0, 0 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 0.0, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_050_clip)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { 12288, -8192, 128, -64, 0, -12288 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 0.5, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_200_clip)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { G_MAXINT16, -32768, 512, -256, 0, G_MININT16 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 2.0, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_050_wrap_negative)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { 12288, -8192, 128, -64, 0, -12288 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 0.5, NULL);
+  g_object_set (G_OBJECT (amplify), "clipping-method", 1, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_200_wrap_negative)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { -16384, -32768, 512, -256, 0, 16384 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 2.0, NULL);
+  g_object_set (G_OBJECT (amplify), "clipping-method", 1, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_050_wrap_positive)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { 12288, -8192, 128, -64, 0, -12288 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 0.5, NULL);
+  g_object_set (G_OBJECT (amplify), "clipping-method", 2, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_200_wrap_positive)
+{
+  GstElement *amplify;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+  gint16 out[6] = { 16382, -32768, 512, -256, 0, -16384 };
+  gint16 *res;
+
+  amplify = setup_amplify ();
+  g_object_set (G_OBJECT (amplify), "amplification", 2.0, NULL);
+  g_object_set (G_OBJECT (amplify), "clipping-method", 2, NULL);
+  fail_unless (gst_element_set_state (amplify,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  inbuffer = gst_buffer_new_and_alloc (12);
+  memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+  fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+  caps = gst_caps_from_string (AMPLIFY_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);
+  /* ... and puts a new buffer on the global list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+  res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+  GST_INFO
+      ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+      out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2],
+      res[3], res[4], res[5]);
+  fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0);
+
+  /* cleanup */
+  cleanup_amplify (amplify);
+}
+
+GST_END_TEST;
+
+Suite *
+amplify_suite (void)
+{
+  Suite *s = suite_create ("amplify");
+  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_zero);
+  tcase_add_test (tc_chain, test_050_clip);
+  tcase_add_test (tc_chain, test_200_clip);
+  tcase_add_test (tc_chain, test_050_wrap_negative);
+  tcase_add_test (tc_chain, test_200_wrap_negative);
+  tcase_add_test (tc_chain, test_050_wrap_positive);
+  tcase_add_test (tc_chain, test_200_wrap_positive);
+  return s;
+}
+
+int
+main (int argc, char **argv)
+{
+  int nf;
+
+  Suite *s = amplify_suite ();
+  SRunner *sr = srunner_create (s);
+
+  gst_check_init (&argc, &argv);
+
+  srunner_run_all (sr, CK_NORMAL);
+  nf = srunner_ntests_failed (sr);
+  srunner_free (sr);
+
+  return nf;
+}