Merge CAPS branch

Original commit message from CVS:
Merge CAPS branch

6 files changed, 639 insertions, 70 deletions

diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am
index 4f95f4ef..d99cbcc6 100644
--- a/gst-libs/gst/audio/Makefile.am
+++ b/gst-libs/gst/audio/Makefile.am
@@ -1,12 +1,21 @@
 librarydir = $(libdir)/gstreamer-@GST_MAJORMINOR@
 
-library_LTLIBRARIES = libgstaudio.la
+library_LTLIBRARIES = libgstaudio.la libgstaudiofilter.la libgstaudiofilterexample.la
 
 libgstaudio_la_SOURCES = audio.c audioclock.c
 
 libgstaudioincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/audio
-libgstaudioinclude_HEADERS = audio.h audioclock.h
+libgstaudioinclude_HEADERS = audio.h audioclock.h gstaudiofilter.h
 
 libgstaudio_la_LIBADD =
 libgstaudio_la_CFLAGS = $(GST_CFLAGS)
 libgstaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+libgstaudiofilter_la_SOURCES = gstaudiofilter.c gstaudiofilter.h
+libgstaudiofilter_la_CFLAGS = $(GST_CFLAGS)
+libgstaudiofilter_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+libgstaudiofilterexample_la_SOURCES = gstaudiofilterexample.c
+libgstaudiofilterexample_la_CFLAGS = $(GST_CFLAGS)
+libgstaudiofilterexample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c
index f5ddaf8b..bec298e9 100644
--- a/gst-libs/gst/audio/audio.c
+++ b/gst-libs/gst/audio/audio.c
@@ -35,22 +35,23 @@ gst_audio_frame_byte_size (GstPad* pad)
 
   int width = 0;
   int channels = 0;
-
-  GstCaps *caps = NULL;
+  GstCaps *caps;
+  GstStructure *structure;
 
   /* get caps of pad */
   caps = GST_PAD_CAPS (pad);
 
-  if (caps == NULL)
-  {
+  if (caps == NULL) {
     /* ERROR: could not get caps of pad */
     g_warning ("gstaudio: could not get caps of pad %s:%s\n", 
 	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
     return 0;
   }
 
-  gst_caps_get_int (caps, "width",    &width);
-  gst_caps_get_int (caps, "channels", &channels);
+  structure = gst_caps_get_structure (caps, 0);
+
+  gst_structure_get_int (structure, "width",    &width);
+  gst_structure_get_int (structure, "channels", &channels);
   return (width / 8) * channels; 
 }
 
@@ -83,6 +84,7 @@ gst_audio_frame_rate (GstPad *pad)
 {
   GstCaps *caps = NULL;
   gint rate;
+  GstStructure *structure;
 
   /* get caps of pad */
   caps = GST_PAD_CAPS (pad);
@@ -94,7 +96,8 @@ gst_audio_frame_rate (GstPad *pad)
     return 0;
   }
   else {
-    gst_caps_get_int (caps, "rate", &rate);
+    structure = gst_caps_get_structure (caps, 0);
+    gst_structure_get_int  (structure, "rate", &rate);
     return rate;
   }
 }
@@ -115,6 +118,7 @@ gst_audio_length (GstPad* pad, GstBuffer* buf)
   double length;
 
   GstCaps *caps = NULL;
+  GstStructure *structure;
 
   g_assert (GST_IS_BUFFER (buf));
   /* get caps of pad */
@@ -128,10 +132,11 @@ gst_audio_length (GstPad* pad, GstBuffer* buf)
   }
   else
   {
+    structure = gst_caps_get_structure (caps, 0);
     bytes = GST_BUFFER_SIZE (buf);
-    gst_caps_get_int (caps, "width",    &width);
-    gst_caps_get_int (caps, "channels", &channels);
-    gst_caps_get_int (caps, "rate",     &rate);
+    gst_structure_get_int  (structure, "width",    &width);
+    gst_structure_get_int  (structure, "channels", &channels);
+    gst_structure_get_int  (structure, "rate",     &rate);
 
     g_assert (bytes != 0);
     g_assert (width != 0);
@@ -152,6 +157,7 @@ gst_audio_highest_sample_value (GstPad* pad)
   gboolean is_signed = FALSE;
   gint width = 0;
   GstCaps *caps = NULL;
+  GstStructure *structure;
   
   caps = GST_PAD_CAPS (pad);
   if (caps == NULL)
@@ -160,8 +166,9 @@ gst_audio_highest_sample_value (GstPad* pad)
 	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
   }
   
-  gst_caps_get_int (caps, "width", &width);
-  gst_caps_get_boolean (caps, "signed", &is_signed);
+  structure = gst_caps_get_structure (caps, 0);
+  gst_structure_get_int  (structure, "width", &width);
+  gst_structure_get_boolean  (structure, "signed", &is_signed);
   
   if (is_signed) --width;
   /* example : 16 bit, signed : samples between -32768 and 32767 */
diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
index 0958f654..73a4043f 100644
--- a/gst-libs/gst/audio/audio.h
+++ b/gst-libs/gst/audio/audio.h
@@ -50,66 +50,43 @@ G_BEGIN_DECLS
 
 #define GST_AUDIO_DEF_RATE 44100
 
-#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
-  gst_props_new (\
-    "rate",       GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    "channels",   GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    "endianness", GST_PROPS_LIST (\
-                    GST_PROPS_INT (G_LITTLE_ENDIAN),\
-                    GST_PROPS_INT (G_BIG_ENDIAN)\
-                  ),\
-    "width",      GST_PROPS_LIST (\
-                    GST_PROPS_INT (8),\
-                    GST_PROPS_INT (16),\
-                    GST_PROPS_INT (32)\
-                  ),\
-    "depth",      GST_PROPS_INT_RANGE (1, 32),\
-    "signed",     GST_PROPS_LIST (\
-                    GST_PROPS_BOOLEAN (TRUE),\
-                    GST_PROPS_BOOLEAN (FALSE)\
-                  ),\
-    "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    NULL)
+#define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
+  "audio/x-raw-int, " \
+  "rate = (int) [ 1, MAX ], " \
+  "channels = (int) [ 1, MAX ], " \
+  "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+  "width = (int) { 8, 16, 32 }, " \
+  "depth = (int) [ 1, 32 ], " \
+  "signed = (boolean) { true, false }, " \
+  "buffer-frames = (int) [ 1, MAX ]"
+
 
 /* "standard" int audio is native order, 16 bit stereo. */
-#define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_PROPS \
-  gst_props_new (\
-    "rate",       GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    "channels",   GST_PROPS_INT (2),\
-    "endianness", GST_PROPS_INT (G_BYTE_ORDER),\
-    "width",      GST_PROPS_INT (16),\
-    "depth",      GST_PROPS_INT (16),\
-    "signed",     GST_PROPS_LIST (\
-                    GST_PROPS_BOOLEAN (TRUE),\
-                    GST_PROPS_BOOLEAN (FALSE)\
-                  ),\
-    "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    NULL)
-
-#define GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS \
-  gst_props_new (\
-    "rate",          GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    "channels",      GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    "endianness",    GST_PROPS_LIST (\
-                       GST_PROPS_INT (G_LITTLE_ENDIAN),\
-                       GST_PROPS_INT (G_BIG_ENDIAN)\
-                     ),\
-    "width",         GST_PROPS_LIST (\
-                       GST_PROPS_INT (32),\
-                       GST_PROPS_INT (64)\
-                     ),\
-    "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    NULL)
+#define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS \
+  "audio/x-raw-int, " \
+  "rate = (int) [ 1, MAX ], " \
+  "channels = (int) 2, " \
+  "endianness = (int) BYTE_ORDER, " \
+  "width = (int) 16, " \
+  "depth = (int) 16, " \
+  "signed = (boolean) true, " \
+  "buffer-frames = (int) [ 1, MAX]"
+
+#define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
+  "audio/x-raw-float, " \
+  "rate = (int) [ 1, MAX ], " \
+  "channels = (int) [ 1, MAX ], " \
+  "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " \
+  "width = (int) { 32, 64 }, " \
+  "buffer-frames = (int) [ 1, MAX]"
 
 /* "standard" float audio is native order, 32 bit mono. */
-#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS \
-  gst_props_new (\
-    "rate",          GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    "channels",      GST_PROPS_INT (1),\
-    "endianness",    GST_PROPS_INT (G_BYTE_ORDER),\
-    "width",         GST_PROPS_INT (32),\
-    "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
-    NULL)
+#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS \
+  "audio/x-raw-float, " \
+  "rate = (int) [ 1, MAX ], " \
+  "channels = (int) 1, " \
+  "endianness = (int) BYTE_ORDER, " \
+  "buffer-frames = (int) [ 1, MAX]"
 
 /*
  * this library defines and implements some helper functions for audio
diff --git a/gst-libs/gst/audio/gstaudiofilter.c b/gst-libs/gst/audio/gstaudiofilter.c
new file mode 100644
index 00000000..46e7c7de
--- /dev/null
+++ b/gst-libs/gst/audio/gstaudiofilter.c
@@ -0,0 +1,322 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2003> David Schleef <ds@schleef.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+/*#define DEBUG_ENABLED */
+#include <gstaudiofilter.h>
+
+#include <string.h>
+
+
+/* GstAudiofilter signals and args */
+enum {
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum {
+  ARG_0,
+  ARG_METHOD,
+  /* FILL ME */
+};
+
+static void     gst_audiofilter_base_init       (gpointer g_class);
+static void	gst_audiofilter_class_init	(gpointer g_class, gpointer class_data);
+static void	gst_audiofilter_init		(GTypeInstance *instance, gpointer g_class);
+
+static void	gst_audiofilter_set_property		(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
+static void	gst_audiofilter_get_property		(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
+
+static void	gst_audiofilter_chain		(GstPad *pad, GstData *_data);
+GstCaps * gst_audiofilter_class_get_capslist(GstAudiofilterClass *klass);
+
+static GstElementClass *parent_class = NULL;
+
+GType
+gst_audiofilter_get_type (void)
+{
+  static GType audiofilter_type = 0;
+
+  if (!audiofilter_type) {
+    static const GTypeInfo audiofilter_info = {
+      sizeof(GstAudiofilterClass),
+      gst_audiofilter_base_init,
+      NULL,
+      gst_audiofilter_class_init,
+      NULL,
+      NULL,
+      sizeof(GstAudiofilter),
+      0,
+      gst_audiofilter_init,
+    };
+    audiofilter_type = g_type_register_static(GST_TYPE_ELEMENT,
+	"GstAudiofilter", &audiofilter_info, G_TYPE_FLAG_ABSTRACT);
+  }
+  return audiofilter_type;
+}
+
+static void gst_audiofilter_base_init (gpointer g_class)
+{
+  static GstElementDetails audiofilter_details = {
+    "Audio filter base class",
+    "Filter/Effect/Audio",
+    "Filters audio",
+    "David Schleef <ds@schleef.org>"
+  };
+  GstAudiofilterClass *klass = (GstAudiofilterClass *) g_class;
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_set_details (element_class, &audiofilter_details);
+}
+
+static void gst_audiofilter_class_init (gpointer g_class, gpointer class_data)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstAudiofilterClass *klass;
+
+  klass = (GstAudiofilterClass *)g_class;
+  gobject_class = (GObjectClass*)klass;
+  gstelement_class = (GstElementClass*)klass;
+
+  parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
+
+  gobject_class->set_property = gst_audiofilter_set_property;
+  gobject_class->get_property = gst_audiofilter_get_property;
+}
+
+static GstCaps *
+gst_audiofilter_getcaps (GstPad *pad)
+{
+  GstAudiofilter *audiofilter;
+  GstCaps *othercaps;
+  GstAudiofilterClass *audiofilter_class;
+
+  GST_DEBUG("gst_audiofilter_sink_getcaps");
+  audiofilter = GST_AUDIOFILTER (gst_pad_get_parent (pad));
+
+  audiofilter_class = GST_AUDIOFILTER_CLASS (
+      G_OBJECT_GET_CLASS (audiofilter));
+  
+  if (pad == audiofilter->srcpad) {
+    othercaps = gst_pad_get_allowed_caps (audiofilter->sinkpad);
+  } else {
+    othercaps = gst_pad_get_allowed_caps (audiofilter->srcpad);
+  }
+
+  return gst_caps_intersect (othercaps, audiofilter_class->caps);
+}
+
+static GstPadLinkReturn
+gst_audiofilter_link (GstPad *pad, const GstCaps *caps)
+{
+  GstAudiofilter *audiofilter;
+  GstPadLinkReturn ret;
+  GstPadLinkReturn link_ret;
+  GstStructure *structure;
+  GstAudiofilterClass *audiofilter_class;
+
+  GST_DEBUG("gst_audiofilter_link");
+  audiofilter = GST_AUDIOFILTER (gst_pad_get_parent (pad));
+  audiofilter_class = GST_AUDIOFILTER_CLASS (
+      G_OBJECT_GET_CLASS (audiofilter));
+  
+
+  if (pad == audiofilter->srcpad) {
+    link_ret = gst_pad_try_set_caps (audiofilter->sinkpad, caps);
+  } else {
+    link_ret = gst_pad_try_set_caps (audiofilter->srcpad, caps);
+  }
+
+  if (link_ret != GST_PAD_LINK_OK) return link_ret;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
+    ret = gst_structure_get_int (structure, "depth", &audiofilter->depth);
+    ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
+    ret &= gst_structure_get_int (structure, "channels", &audiofilter->channels);
+  } else if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float")
+      == 0) {
+
+  } else {
+    g_assert_not_reached();
+  }
+  ret &= gst_structure_get_int (structure, "rate", &audiofilter->rate);
+
+  if (audiofilter_class->setup) (audiofilter_class->setup) (audiofilter);
+
+  return GST_PAD_LINK_OK;
+}
+
+static void
+gst_audiofilter_init (GTypeInstance *instance, gpointer g_class)
+{
+  GstAudiofilter *audiofilter = GST_AUDIOFILTER (instance);
+  GstPadTemplate *pad_template;
+
+  GST_DEBUG("gst_audiofilter_init");
+
+  pad_template = gst_element_class_get_pad_template(GST_ELEMENT_CLASS(g_class),
+      "sink");
+  g_return_if_fail(pad_template != NULL);
+  audiofilter->sinkpad = gst_pad_new_from_template(pad_template, "sink");
+  gst_element_add_pad(GST_ELEMENT(audiofilter),audiofilter->sinkpad);
+  gst_pad_set_chain_function(audiofilter->sinkpad,gst_audiofilter_chain);
+  gst_pad_set_link_function(audiofilter->sinkpad,gst_audiofilter_link);
+  gst_pad_set_getcaps_function(audiofilter->sinkpad,gst_audiofilter_getcaps);
+
+  pad_template = gst_element_class_get_pad_template(GST_ELEMENT_CLASS(g_class),
+      "src");
+  g_return_if_fail(pad_template != NULL);
+  audiofilter->srcpad = gst_pad_new_from_template(pad_template, "src");
+  gst_element_add_pad(GST_ELEMENT(audiofilter),audiofilter->srcpad);
+  gst_pad_set_link_function(audiofilter->srcpad,gst_audiofilter_link);
+  gst_pad_set_getcaps_function(audiofilter->srcpad,gst_audiofilter_getcaps);
+
+  audiofilter->inited = FALSE;
+}
+
+static void
+gst_audiofilter_chain (GstPad *pad, GstData *data)
+{
+  GstBuffer *inbuf = GST_BUFFER (data);
+  GstAudiofilter *audiofilter;
+  GstBuffer *outbuf;
+  GstAudiofilterClass *audiofilter_class;
+
+  GST_DEBUG ("gst_audiofilter_chain");
+
+  g_return_if_fail (pad != NULL);
+  g_return_if_fail (GST_IS_PAD (pad));
+  g_return_if_fail (inbuf != NULL);
+
+  audiofilter = GST_AUDIOFILTER (gst_pad_get_parent (pad));
+  //g_return_if_fail (audiofilter->inited);
+  audiofilter_class = GST_AUDIOFILTER_CLASS (
+      G_OBJECT_GET_CLASS (audiofilter));
+
+  GST_DEBUG ("gst_audiofilter_chain: got buffer of %d bytes in '%s'",
+      GST_BUFFER_SIZE(inbuf), GST_OBJECT_NAME (audiofilter));
+ 
+  if(audiofilter->passthru){
+    gst_pad_push(audiofilter->srcpad, data);
+    return;
+  }
+
+  if (gst_data_is_writable(data)) {
+    if (audiofilter_class->filter_inplace) {
+      (audiofilter_class->filter_inplace) (audiofilter, inbuf);
+      outbuf = inbuf;
+    } else {
+      outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE(inbuf));
+      GST_BUFFER_DURATION(outbuf) = GST_BUFFER_DURATION(inbuf);
+      GST_BUFFER_TIMESTAMP(outbuf) = GST_BUFFER_TIMESTAMP(inbuf);
+
+      (audiofilter_class->filter) (audiofilter, outbuf, inbuf);
+      gst_buffer_unref(inbuf);
+    }
+  } else {
+    outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE(inbuf));
+    GST_BUFFER_DURATION(outbuf) = GST_BUFFER_DURATION(inbuf);
+    GST_BUFFER_TIMESTAMP(outbuf) = GST_BUFFER_TIMESTAMP(inbuf);
+    if (audiofilter_class->filter) {
+      (audiofilter_class->filter) (audiofilter, outbuf, inbuf);
+    } else {
+      memcpy(GST_BUFFER_DATA(outbuf), GST_BUFFER_DATA(inbuf),
+          GST_BUFFER_SIZE(inbuf));
+
+      (audiofilter_class->filter_inplace) (audiofilter, outbuf);
+    }
+    gst_buffer_unref(inbuf);
+  }
+
+  gst_pad_push(audiofilter->srcpad, GST_DATA (outbuf));
+}
+
+static void
+gst_audiofilter_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+{
+  GstAudiofilter *src;
+
+  /* it's not null if we got it, but it might not be ours */
+  g_return_if_fail(GST_IS_AUDIOFILTER(object));
+  src = GST_AUDIOFILTER(object);
+
+  GST_DEBUG("gst_audiofilter_set_property");
+  switch (prop_id) {
+    default:
+      break;
+  }
+}
+
+static void
+gst_audiofilter_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+{
+  GstAudiofilter *src;
+
+  /* it's not null if we got it, but it might not be ours */
+  g_return_if_fail(GST_IS_AUDIOFILTER(object));
+  src = GST_AUDIOFILTER(object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+void gst_audiofilter_class_add_pad_templates (
+    GstAudiofilterClass *audiofilter_class, const GstCaps *caps)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (audiofilter_class);
+
+  audiofilter_class->caps = gst_caps_copy(caps);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_pad_template_new("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+        gst_caps_copy(caps)));
+
+  gst_element_class_add_pad_template (element_class,
+      gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+        gst_caps_copy(caps)));
+}
+
+static gboolean
+plugin_init (GstPlugin *plugin)
+{
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (
+  GST_VERSION_MAJOR,
+  GST_VERSION_MINOR,
+  "gstaudiofilter",
+  "Audio filter parent class",
+  plugin_init,
+  VERSION,
+  "LGPL",
+  GST_PACKAGE,
+  GST_ORIGIN
+)
+
diff --git a/gst-libs/gst/audio/gstaudiofilter.h b/gst-libs/gst/audio/gstaudiofilter.h
new file mode 100644
index 00000000..4723c329
--- /dev/null
+++ b/gst-libs/gst/audio/gstaudiofilter.h
@@ -0,0 +1,84 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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_AUDIOFILTER_H__
+#define __GST_AUDIOFILTER_H__
+
+
+#include <gst/gst.h>
+
+
+G_BEGIN_DECLS
+
+typedef struct _GstAudiofilter GstAudiofilter;
+typedef struct _GstAudiofilterClass GstAudiofilterClass;
+
+typedef void (*GstAudiofilterFilterFunc)(GstAudiofilter *filter,
+    GstBuffer *outbuf, GstBuffer *inbuf);
+typedef void (*GstAudiofilterInplaceFilterFunc)(GstAudiofilter *filter,
+    GstBuffer *buffer);
+
+typedef void (*GstAudiofilterSetupFunc) (GstAudiofilter *filter);
+
+
+#define GST_TYPE_AUDIOFILTER \
+  (gst_audiofilter_get_type())
+#define GST_AUDIOFILTER(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIOFILTER,GstAudiofilter))
+#define GST_AUDIOFILTER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIOFILTER,GstAudiofilterClass))
+#define GST_IS_AUDIOFILTER(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIOFILTER))
+#define GST_IS_AUDIOFILTER_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOFILTER))
+
+struct _GstAudiofilter {
+  GstElement element;
+
+  GstPad *sinkpad,*srcpad;
+
+  /* audio state */
+  gboolean inited;
+
+  int rate;
+  int width;
+  int channels;
+  int depth;
+  gboolean passthru;
+
+};
+
+struct _GstAudiofilterClass {
+  GstElementClass parent_class;
+
+  GstCaps *caps;
+  GstAudiofilterSetupFunc setup;
+  GstAudiofilterInplaceFilterFunc filter_inplace;
+  GstAudiofilterFilterFunc filter;
+};
+
+GType gst_audiofilter_get_type(void);
+
+void gst_audiofilter_class_add_pad_templates (GstAudiofilterClass *audiofilterclass, const GstCaps *caps);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIOFILTER_H__ */
+
diff --git a/gst-libs/gst/audio/gstaudiofilterexample.c b/gst-libs/gst/audio/gstaudiofilterexample.c
new file mode 100644
index 00000000..15e111af
--- /dev/null
+++ b/gst-libs/gst/audio/gstaudiofilterexample.c
@@ -0,0 +1,170 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2003> David Schleef <ds@schleef.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiofilter.h>
+
+typedef struct _GstAudiofilterExample GstAudiofilterExample;
+typedef struct _GstAudiofilterExampleClass GstAudiofilterExampleClass;
+
+#define GST_TYPE_AUDIOFILTER_EXAMPLE \
+  (gst_audiofilter_example_get_type())
+#define GST_AUDIOFILTER_EXAMPLE(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIOFILTER_EXAMPLE,GstAudiofilterExample))
+#define GST_AUDIOFILTER_EXAMPLE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIOFILTER_EXAMPLE,GstAudiofilterExampleClass))
+#define GST_IS_AUDIOFILTER_EXAMPLE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIOFILTER_EXAMPLE))
+#define GST_IS_AUDIOFILTER_EXAMPLE_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOFILTER_EXAMPLE))
+
+struct _GstAudiofilterExample {
+  GstAudiofilter audiofilter;
+
+};
+
+struct _GstAudiofilterExampleClass {
+  GstAudiofilterClass parent_class;
+
+};
+
+
+/* GstAudiofilterExample signals and args */
+enum {
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum {
+  ARG_0,
+  ARG_METHOD,
+  /* FILL ME */
+};
+
+static void     gst_audiofilter_example_base_init       (gpointer g_class);
+static void	gst_audiofilter_example_class_init	(gpointer g_class, gpointer class_data);
+
+static void	gst_audiofilter_example_set_property		(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
+static void	gst_audiofilter_example_get_property		(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
+
+GType
+gst_audiofilter_example_get_type (void)
+{
+  static GType audiofilter_example_type = 0;
+
+  if (!audiofilter_example_type) {
+    static const GTypeInfo audiofilter_example_info = {
+      sizeof(GstAudiofilterExampleClass),
+      gst_audiofilter_example_base_init,
+      NULL,
+      gst_audiofilter_example_class_init,
+      NULL,
+      NULL,
+      sizeof(GstAudiofilterExample),
+      0,
+      NULL,
+    };
+    audiofilter_example_type = g_type_register_static(GST_TYPE_AUDIOFILTER,
+	"GstAudiofilterExample", &audiofilter_example_info, 0);
+  }
+  return audiofilter_example_type;
+}
+
+static void gst_audiofilter_example_base_init (gpointer g_class)
+{
+  static GstElementDetails audiofilter_example_details = {
+    "Audio filter example",
+    "Filter/Effect/Audio",
+    "Filters audio",
+    "David Schleef <ds@schleef.org>"
+  };
+  GstAudiofilterExampleClass *klass = (GstAudiofilterExampleClass *) g_class;
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_set_details (element_class, &audiofilter_example_details);
+}
+
+static void gst_audiofilter_example_class_init (gpointer g_class, gpointer class_data)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstAudiofilterExampleClass *klass;
+
+  klass = (GstAudiofilterExampleClass *)g_class;
+  gobject_class = (GObjectClass*)klass;
+  gstelement_class = (GstElementClass*)klass;
+
+  gobject_class->set_property = gst_audiofilter_example_set_property;
+  gobject_class->get_property = gst_audiofilter_example_get_property;
+}
+
+static void
+gst_audiofilter_example_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+{
+  GstAudiofilterExample *src;
+
+  /* it's not null if we got it, but it might not be ours */
+  g_return_if_fail(GST_IS_AUDIOFILTER_EXAMPLE(object));
+  src = GST_AUDIOFILTER_EXAMPLE(object);
+
+  GST_DEBUG("gst_audiofilter_example_set_property");
+  switch (prop_id) {
+    default:
+      break;
+  }
+}
+
+static void
+gst_audiofilter_example_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+{
+  GstAudiofilterExample *src;
+
+  /* it's not null if we got it, but it might not be ours */
+  g_return_if_fail(GST_IS_AUDIOFILTER_EXAMPLE(object));
+  src = GST_AUDIOFILTER_EXAMPLE(object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+plugin_init (GstPlugin *plugin)
+{
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (
+  GST_VERSION_MAJOR,
+  GST_VERSION_MINOR,
+  "gstaudiofilter_example",
+  "Audio filter example",
+  plugin_init,
+  VERSION,
+  "LGPL",
+  GST_PACKAGE,
+  GST_ORIGIN
+)