remove gst-libs from gst-plugins module as it is in gst-plugins-base now

Original commit message from CVS:
remove gst-libs from gst-plugins module as it is in gst-plugins-base now

15 files changed, 0 insertions, 2543 deletions

diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am
deleted file mode 100644
index f00717b1..00000000
--- a/gst-libs/gst/audio/Makefile.am
+++ /dev/null
@@ -1,53 +0,0 @@
-# variables used for enum/marshal generation
-glib_enum_headers=multichannel.h
-glib_enum_define=GST_AUDIO
-glib_enum_prefix=gst_audio
-
-built_sources = multichannel-enumtypes.c
-built_headers = multichannel-enumtypes.h
-BUILT_SOURCES = $(built_sources) $(built_headers)
-
-librarydir = $(libdir)/gstreamer-@GST_MAJORMINOR@
-
-library_LTLIBRARIES = libgstaudio.la libgstaudiofilter.la
-noinst_LTLIBRARIES = libgstaudiofilterexample.la
-
-EXTRA_DIST = gstaudiofiltertemplate.c make_filter
-CLEANFILES = gstaudiofilterexample.c \
-	$(BUILT_SOURCES)
-
-libgstaudio_la_SOURCES = audio.c audioclock.c \
-	multichannel.c
-nodist_libgstaudio_la_SOURCES = $(built_sources)
-
-libgstaudioincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/audio
-libgstaudioinclude_HEADERS = \
-	audio.h \
-	audioclock.h \
-	gstaudiofilter.h \
-	multichannel.h
-
-nodist_libgstaudioinclude_HEADERS = \
-	multichannel-enumtypes.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)
-
-gstaudiofilterexample.c: $(srcdir)/make_filter $(srcdir)/gstaudiofiltertemplate.c
-	$(srcdir)/make_filter AudiofilterExample $(srcdir)/gstaudiofiltertemplate.c
-
-noinst_PROGRAMS = testchannels
-testchannels_SOURCES = testchannels.c
-testchannels_CFLAGS = $(GST_CFLAGS)
-testchannels_LDFLAGS = $(GST_LIBS)
-
-include $(top_srcdir)/common/glib-gen.mak
diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c
deleted file mode 100644
index b6a1edb5..00000000
--- a/gst-libs/gst/audio/audio.c
+++ /dev/null
@@ -1,280 +0,0 @@
-/* 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.
- */
-
-#ifdef HAVE_CONFIG_H
-#  include "config.h"
-#endif
-
-#include "audio.h"
-#include "multichannel-enumtypes.h"
-
-#include <gst/gststructure.h>
-
-int
-gst_audio_frame_byte_size (GstPad * pad)
-{
-/* calculate byte size of an audio frame
- * this should be moved closer to the gstreamer core
- * and be implemented for every mime type IMO
- * returns -1 if there's an error (to avoid division by zero), 
- * or the byte size if everything's ok
- */
-
-  int width = 0;
-  int channels = 0;
-  const GstCaps *caps = NULL;
-  GstStructure *structure;
-
-  /* get caps of pad */
-  caps = GST_PAD_CAPS (pad);
-
-  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;
-  }
-
-  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;
-}
-
-long
-gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
-/* calculate length of buffer in frames
- * this should be moved closer to the gstreamer core
- * and be implemented for every mime type IMO
- * returns 0 if there's an error, or the number of frames if everything's ok
- */
-{
-  int frame_byte_size = 0;
-
-  frame_byte_size = gst_audio_frame_byte_size (pad);
-  if (frame_byte_size == 0)
-    /* error */
-    return 0;
-  /* FIXME: this function assumes the buffer size to be a whole multiple
-   *        of the frame byte size
-   */
-  return GST_BUFFER_SIZE (buf) / frame_byte_size;
-}
-
-long
-gst_audio_frame_rate (GstPad * pad)
-/*
- * calculate frame rate (based on caps of pad)
- * returns 0 if failed, rate if success
- */
-{
-  const GstCaps *caps = NULL;
-  gint rate;
-  GstStructure *structure;
-
-  /* get caps of pad */
-  caps = GST_PAD_CAPS (pad);
-
-  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;
-  } else {
-    structure = gst_caps_get_structure (caps, 0);
-    gst_structure_get_int (structure, "rate", &rate);
-    return rate;
-  }
-}
-
-double
-gst_audio_length (GstPad * pad, GstBuffer * buf)
-{
-/* calculate length in seconds
- * of audio buffer buf
- * based on capabilities of pad
- */
-
-  long bytes = 0;
-  int width = 0;
-  int channels = 0;
-  int rate = 0;
-
-  double length;
-
-  const GstCaps *caps = NULL;
-  GstStructure *structure;
-
-  g_assert (GST_IS_BUFFER (buf));
-  /* get caps of pad */
-  caps = GST_PAD_CAPS (pad);
-  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));
-    length = 0.0;
-  } else {
-    structure = gst_caps_get_structure (caps, 0);
-    bytes = GST_BUFFER_SIZE (buf);
-    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);
-    g_assert (channels != 0);
-    g_assert (rate != 0);
-    length = (bytes * 8.0) / (double) (rate * channels * width);
-  }
-  /* g_print ("DEBUG: audio: returning length of %f\n", length); */
-  return length;
-}
-
-long
-gst_audio_highest_sample_value (GstPad * pad)
-/* calculate highest possible sample value
- * based on capabilities of pad
- */
-{
-  gboolean is_signed = FALSE;
-  gint width = 0;
-  const GstCaps *caps = NULL;
-  GstStructure *structure;
-
-  caps = GST_PAD_CAPS (pad);
-  if (caps == NULL) {
-    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
-        GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
-  }
-
-  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 */
-  return ((long) (1 << width));
-}
-
-gboolean
-gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
-/* check if the buffer size is a whole multiple of the frame size */
-{
-  if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
-    return TRUE;
-  else
-    return FALSE;
-}
-
-/* _getcaps helper functions
- * sets structure fields to default for audio type
- * flag determines which structure fields to set to default
- * keep these functions in sync with the templates in audio.h
- */
-
-/* private helper function
- * sets a list on the structure
- * pass in structure, fieldname for the list, type of the list values,
- * number of list values, and each of the values, terminating with NULL
- */
-static void
-_gst_audio_structure_set_list (GstStructure * structure,
-    const gchar * fieldname, GType type, int number, ...)
-{
-  va_list varargs;
-  GValue value = { 0 };
-  GArray *array;
-  int j;
-
-  g_return_if_fail (structure != NULL);
-
-  g_value_init (&value, GST_TYPE_LIST);
-  array = g_value_peek_pointer (&value);
-
-  va_start (varargs, number);
-
-  for (j = 0; j < number; ++j) {
-    int i;
-    gboolean b;
-
-    GValue list_value = { 0 };
-
-    switch (type) {
-      case G_TYPE_INT:
-        i = va_arg (varargs, int);
-
-        g_value_init (&list_value, G_TYPE_INT);
-        g_value_set_int (&list_value, i);
-        break;
-      case G_TYPE_BOOLEAN:
-        b = va_arg (varargs, gboolean);
-        g_value_init (&list_value, G_TYPE_BOOLEAN);
-        g_value_set_boolean (&list_value, b);
-        break;
-      default:
-        g_warning
-            ("_gst_audio_structure_set_list: LIST of given type not implemented.");
-    }
-    g_array_append_val (array, list_value);
-
-  }
-  gst_structure_set_value (structure, fieldname, &value);
-  va_end (varargs);
-}
-
-void
-gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
-{
-  if (flag & GST_AUDIO_FIELD_RATE)
-    gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
-        NULL);
-  if (flag & GST_AUDIO_FIELD_CHANNELS)
-    gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
-        NULL);
-  if (flag & GST_AUDIO_FIELD_ENDIANNESS)
-    _gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
-        G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
-  if (flag & GST_AUDIO_FIELD_WIDTH)
-    _gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
-        NULL);
-  if (flag & GST_AUDIO_FIELD_DEPTH)
-    gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
-  if (flag & GST_AUDIO_FIELD_SIGNED)
-    _gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
-        FALSE, NULL);
-  if (flag & GST_AUDIO_FIELD_BUFFER_FRAMES)
-    gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 1,
-        G_MAXINT, NULL);
-}
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  gst_audio_channel_position_get_type ();
-
-  return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    "gstaudio",
-    "Support services for audio plugins",
-    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN);
diff --git a/gst-libs/gst/audio/audio.def b/gst-libs/gst/audio/audio.def
deleted file mode 100644
index 96a3ac06..00000000
--- a/gst-libs/gst/audio/audio.def
+++ /dev/null
@@ -1,5 +0,0 @@
-EXPORTS
-    gst_plugin_desc
-    gst_audio_length
-    gst_audio_is_buffer_framed
-    gst_audio_highest_sample_value
diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
deleted file mode 100644
index 8556ce1f..00000000
--- a/gst-libs/gst/audio/audio.h
+++ /dev/null
@@ -1,131 +0,0 @@
-/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Library       <2001> Thomas Vander Stichele <thomas@apestaart.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.
- */
-
-#include <gst/gst.h>
-
-#include <gst/audio/audioclock.h>
-
-#ifndef __GST_AUDIO_AUDIO_H__
-#define __GST_AUDIO_AUDIO_H__
-
-G_BEGIN_DECLS
-
-/* For people that are looking at this source: the purpose of these defines is
- * to make GstCaps a bit easier, in that you don't have to know all of the
- * properties that need to be defined. you can just use these macros. currently
- * (8/01) the only plugins that use these are the passthrough, speed, volume,
- * adder, and [de]interleave plugins. These are for convenience only, and do not
- * specify the 'limits' of GStreamer. you might also use these definitions as a
- * base for making your own caps, if need be.
- *
- * For example, to make a source pad that can output streams of either mono
- * float or any channel int:
- *
- *  template = gst_pad_template_new
- *    ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
- *    gst_caps_append(gst_caps_new ("sink_int",  "audio/x-raw-int",
- *                                  GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
- *                    gst_caps_new ("sink_float", "audio/x-raw-float",
- *                                  GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
- *    NULL);
- *
- *  sinkpad = gst_pad_new_from_template(template, "sink");
- *
- * Andy Wingo, 18 August 2001
- * Thomas, 6 September 2002 */
-
-#define GST_AUDIO_DEF_RATE 44100
-
-#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, 24, 32 }, " \
-  "depth = (int) [ 1, 32 ], " \
-  "signed = (boolean) { true, false }" 
-
-
-/* "standard" int audio is native order, 16 bit stereo. */
-#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" 
-
-#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_CAPS \
-  "audio/x-raw-float, " \
-  "width = (int) 32, " \
-  "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
- * handling
- */
-
-/* get byte size of audio frame (based on caps of pad */
-int      gst_audio_frame_byte_size      (GstPad* pad);
-
-/* get length in frames of buffer */
-long     gst_audio_frame_length         (GstPad* pad, GstBuffer* buf);
-
-/* get frame rate based on caps */
-long     gst_audio_frame_rate           (GstPad *pad);
-
-/* calculate length in seconds of audio buffer buf based on caps of pad */
-double   gst_audio_length               (GstPad* pad, GstBuffer* buf);
-
-/* calculate highest possible sample value based on capabilities of pad */
-long     gst_audio_highest_sample_value (GstPad* pad);
-
-/* check if the buffer size is a whole multiple of the frame size */
-gboolean gst_audio_is_buffer_framed     (GstPad* pad, GstBuffer* buf);
-
-/* functions useful for _getcaps functions */
-typedef enum {
-  GST_AUDIO_FIELD_RATE          = (1 << 0),
-  GST_AUDIO_FIELD_CHANNELS      = (1 << 1),
-  GST_AUDIO_FIELD_ENDIANNESS    = (1 << 2),
-  GST_AUDIO_FIELD_WIDTH         = (1 << 3),
-  GST_AUDIO_FIELD_DEPTH         = (1 << 4),
-  GST_AUDIO_FIELD_SIGNED        = (1 << 5),
-  GST_AUDIO_FIELD_BUFFER_FRAMES = (1 << 6)
-} GstAudioFieldFlag;
-
-void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
-
-G_END_DECLS
-
-#endif /* __GST_AUDIO_AUDIO_H__ */
diff --git a/gst-libs/gst/audio/audio.vcproj b/gst-libs/gst/audio/audio.vcproj
deleted file mode 100644
index 21115317..00000000
--- a/gst-libs/gst/audio/audio.vcproj
+++ /dev/null
@@ -1,153 +0,0 @@
-<?xml version="1.0" encoding="Windows-1252"?>
-<VisualStudioProject
-	ProjectType="Visual C++"
-	Version="7.10"
-	Name="audio"
-	ProjectGUID="{979C216F-0ACF-4956-AE00-055A42D67893}"
-	RootNamespace="audio"
-	Keyword="Win32Proj">
-	<Platforms>
-		<Platform
-			Name="Win32"/>
-	</Platforms>
-	<Configurations>
-		<Configuration
-			Name="Debug|Win32"
-			OutputDirectory="../../../win32/Debug"
-			IntermediateDirectory="../../../win32/Debug"
-			ConfigurationType="2"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				Optimization="0"
-				AdditionalIncludeDirectories="../../../../gstreamer/win32;../../../../gstreamer;../../../../gstreamer/libs;../../../../glib;../../../../glib/glib;../../../../glib/gmodule;&quot;../../../gst-libs&quot;;../../../../popt/include;../../../../libxml2/include/libxml2"
-				PreprocessorDefinitions="WIN32;_DEBUG;_WINDOWS;_USRDLL;AUDIO_EXPORTS;HAVE_CONFIG_H;_USE_MATH_DEFINES"
-				MinimalRebuild="TRUE"
-				BasicRuntimeChecks="3"
-				RuntimeLibrary="3"
-				UsePrecompiledHeader="0"
-				WarningLevel="3"
-				Detect64BitPortabilityProblems="TRUE"
-				DebugInformationFormat="4"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				Name="VCLinkerTool"
-				AdditionalDependencies="glib-2.0.lib gmodule-2.0.lib gthread-2.0.lib gobject-2.0.lib libgstreamer.lib gstbytestream.lib iconv.lib intl.lib"
-				OutputFile="$(OutDir)/gstaudio.dll"
-				LinkIncremental="2"
-				AdditionalLibraryDirectories="../../../../gstreamer/win32/Debug;../../../../glib/glib;../../../../glib/gmodule;../../../../glib/gthread;../../../../glib/gobject;../../../../gettext/lib;../../../../libiconv/lib"
-				ModuleDefinitionFile="audio.def"
-				GenerateDebugInformation="TRUE"
-				ProgramDatabaseFile="$(OutDir)/audio.pdb"
-				SubSystem="2"
-				OptimizeReferences="2"
-				ImportLibrary="$(OutDir)/gstaudio.lib"
-				TargetMachine="1"/>
-			<Tool
-				Name="VCMIDLTool"/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy /Y $(TargetPath) c:\gstreamer\plugins"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCWebDeploymentTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-		<Configuration
-			Name="Release|Win32"
-			OutputDirectory="../../../win32/Release"
-			IntermediateDirectory="../../../win32/Release"
-			ConfigurationType="2"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../gstreamer/win32;../../../../gstreamer;../../../../gstreamer/libs;../../../../glib;../../../../glib/glib;../../../../glib/gmodule;&quot;../../../gst-libs&quot;;../../../../popt/include;../../../../libxml2/include/libxml2"
-				PreprocessorDefinitions="WIN32;NDEBUG;GST_DISABLE_GST_DEBUG;_WINDOWS;_USRDLL;AUDIO_EXPORTS;HAVE_CONFIG_H;_USE_MATH_DEFINES"
-				RuntimeLibrary="2"
-				UsePrecompiledHeader="0"
-				WarningLevel="3"
-				Detect64BitPortabilityProblems="TRUE"
-				DebugInformationFormat="3"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				Name="VCLinkerTool"
-				AdditionalDependencies="glib-2.0.lib gmodule-2.0.lib gthread-2.0.lib gobject-2.0.lib libgstreamer.lib gstbytestream.lib iconv.lib intl.lib"
-				OutputFile="$(OutDir)/gstaudio.dll"
-				LinkIncremental="1"
-				AdditionalLibraryDirectories="../../../../gstreamer/win32/Release;../../../../glib/glib;../../../../glib/gmodule;../../../../glib/gthread;../../../../glib/gobject;../../../../gettext/lib;../../../../libiconv/lib"
-				ModuleDefinitionFile="audio.def"
-				GenerateDebugInformation="TRUE"
-				SubSystem="2"
-				OptimizeReferences="2"
-				EnableCOMDATFolding="2"
-				ImportLibrary="$(OutDir)/gstaudio.lib"
-				TargetMachine="1"/>
-			<Tool
-				Name="VCMIDLTool"/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy /Y $(TargetPath) c:\gstreamer\plugins"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCWebDeploymentTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-	</Configurations>
-	<References>
-	</References>
-	<Files>
-		<Filter
-			Name="Source Files"
-			Filter="cpp;c;cxx;def;odl;idl;hpj;bat;asm;asmx"
-			UniqueIdentifier="{4FC737F1-C7A5-4376-A066-2A32D752A2FF}">
-			<File
-				RelativePath=".\audio.c">
-			</File>
-			<File
-				RelativePath=".\audio.def">
-			</File>
-			<File
-				RelativePath=".\audioclock.c">
-			</File>
-		</Filter>
-		<Filter
-			Name="Header Files"
-			Filter="h;hpp;hxx;hm;inl;inc;xsd"
-			UniqueIdentifier="{93995380-89BD-4b04-88EB-625FBE52EBFB}">
-			<File
-				RelativePath=".\audio.h">
-			</File>
-			<File
-				RelativePath=".\audioclock.h">
-			</File>
-		</Filter>
-	</Files>
-	<Globals>
-	</Globals>
-</VisualStudioProject>
diff --git a/gst-libs/gst/audio/audioclock.c b/gst-libs/gst/audio/audioclock.c
deleted file mode 100644
index 65d2694f..00000000
--- a/gst-libs/gst/audio/audioclock.c
+++ /dev/null
@@ -1,205 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2000 Wim Taymans <wtay@chello.be>
- *
- * audioclock.c: Clock for use by audio plugins
- *
- * 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 "audioclock.h"
-
-static void gst_audio_clock_class_init (GstAudioClockClass * klass);
-static void gst_audio_clock_init (GstAudioClock * clock);
-
-static GstClockTime gst_audio_clock_get_internal_time (GstClock * clock);
-static GstClockEntryStatus gst_audio_clock_id_wait_async (GstClock * clock,
-    GstClockEntry * entry);
-static void gst_audio_clock_id_unschedule (GstClock * clock,
-    GstClockEntry * entry);
-
-static GstSystemClockClass *parent_class = NULL;
-
-/* static guint gst_audio_clock_signals[LAST_SIGNAL] = { 0 }; */
-
-GType
-gst_audio_clock_get_type (void)
-{
-  static GType clock_type = 0;
-
-  if (!clock_type) {
-    static const GTypeInfo clock_info = {
-      sizeof (GstAudioClockClass),
-      NULL,
-      NULL,
-      (GClassInitFunc) gst_audio_clock_class_init,
-      NULL,
-      NULL,
-      sizeof (GstAudioClock),
-      4,
-      (GInstanceInitFunc) gst_audio_clock_init,
-      NULL
-    };
-
-    clock_type = g_type_register_static (GST_TYPE_SYSTEM_CLOCK, "GstAudioClock",
-        &clock_info, 0);
-  }
-  return clock_type;
-}
-
-
-static void
-gst_audio_clock_class_init (GstAudioClockClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstObjectClass *gstobject_class;
-  GstClockClass *gstclock_class;
-
-  gobject_class = (GObjectClass *) klass;
-  gstobject_class = (GstObjectClass *) klass;
-  gstclock_class = (GstClockClass *) klass;
-
-  parent_class = g_type_class_ref (GST_TYPE_SYSTEM_CLOCK);
-
-  gstclock_class->get_internal_time = gst_audio_clock_get_internal_time;
-  gstclock_class->wait_async = gst_audio_clock_id_wait_async;
-  gstclock_class->unschedule = gst_audio_clock_id_unschedule;
-}
-
-static void
-gst_audio_clock_init (GstAudioClock * clock)
-{
-  gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
-
-  clock->prev1 = 0;
-  clock->prev2 = 0;
-}
-
-GstClock *
-gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
-    gpointer user_data)
-{
-  GstAudioClock *aclock =
-      GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
-
-  aclock->func = func;
-  aclock->user_data = user_data;
-  aclock->adjust = 0;
-
-  return (GstClock *) aclock;
-}
-
-void
-gst_audio_clock_set_active (GstAudioClock * aclock, gboolean active)
-{
-  GstClockTime audio_time, system_time;
-  GstClock *clock;
-  GTimeVal timeval;
-
-  g_return_if_fail (GST_IS_AUDIO_CLOCK (aclock));
-  clock = GST_CLOCK (aclock);
-
-  if (active == aclock->active) {
-    /* Nothing to do. */
-    return;
-  }
-
-  audio_time = aclock->func (clock, aclock->user_data);
-
-  g_get_current_time (&timeval);
-  system_time = GST_TIMEVAL_TO_TIME (timeval);
-
-  /* Set the new adjust value in such a way that there's no abrupt
-     discontinuity, i.e. if gst_audio_clock_get_internal_time is
-     invoked right before and right after (de)activating the clock,
-     the values returned will be close to each other, and the second
-     value will be greater than or equal than the first. */
-  if (active) {
-    aclock->adjust = aclock->adjust + system_time - audio_time;
-  } else {
-    aclock->adjust = aclock->adjust + audio_time - system_time;
-  }
-
-  aclock->active = active;
-}
-
-static GstClockTime
-gst_audio_clock_get_internal_time (GstClock * clock)
-{
-  GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
-
-  if (aclock->active) {
-    return aclock->func (clock, aclock->user_data) + aclock->adjust;
-  } else {
-    GTimeVal timeval;
-
-    g_get_current_time (&timeval);
-    return GST_TIMEVAL_TO_TIME (timeval) + aclock->adjust;
-  }
-}
-
-void
-gst_audio_clock_update_time (GstAudioClock * aclock, GstClockTime time)
-{
-  /* I don't know of a purpose in updating these; perhaps they can be removed */
-  aclock->prev2 = aclock->prev1;
-  aclock->prev1 = time;
-
-  /* FIXME: the wait_async subsystem should be made threadsafe, but I don't want
-   * to lock and unlock a mutex on every iteration... */
-  while (aclock->async_entries) {
-    GstClockEntry *entry = (GstClockEntry *) aclock->async_entries->data;
-
-    if (entry->time > time)
-      break;
-
-    entry->func ((GstClock *) aclock, time, entry, entry->user_data);
-
-    aclock->async_entries = g_slist_delete_link (aclock->async_entries,
-        aclock->async_entries);
-    /* do I need to free the entry? */
-  }
-}
-
-static gint
-compare_clock_entries (GstClockEntry * entry1, GstClockEntry * entry2)
-{
-  return entry1->time - entry2->time;
-}
-
-static GstClockEntryStatus
-gst_audio_clock_id_wait_async (GstClock * clock, GstClockEntry * entry)
-{
-  GstAudioClock *aclock = (GstAudioClock *) clock;
-
-  aclock->async_entries = g_slist_insert_sorted (aclock->async_entries,
-      entry, (GCompareFunc) compare_clock_entries);
-
-  /* is this the proper return val? */
-  return GST_CLOCK_EARLY;
-}
-
-static void
-gst_audio_clock_id_unschedule (GstClock * clock, GstClockEntry * entry)
-{
-  GstAudioClock *aclock = (GstAudioClock *) clock;
-
-  aclock->async_entries = g_slist_remove (aclock->async_entries, entry);
-}
diff --git a/gst-libs/gst/audio/audioclock.h b/gst-libs/gst/audio/audioclock.h
deleted file mode 100644
index 17439242..00000000
--- a/gst-libs/gst/audio/audioclock.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *                    2000 Wim Taymans <wtay@chello.be>
- *
- * audioclock.h: Clock for use by audio plugins
- *
- * 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_CLOCK_H__
-#define __GST_AUDIO_CLOCK_H__
-
-#include <gst/gstsystemclock.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AUDIO_CLOCK \
-  (gst_audio_clock_get_type())
-#define GST_AUDIO_CLOCK(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CLOCK,GstAudioClock))
-#define GST_AUDIO_CLOCK_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CLOCK,GstAudioClockClass))
-#define GST_IS_AUDIO_CLOCK(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CLOCK))
-#define GST_IS_AUDIO_CLOCK_CLASS(obj) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CLOCK))
-
-typedef struct _GstAudioClock GstAudioClock;
-typedef struct _GstAudioClockClass GstAudioClockClass;
-
-typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock *clock, gpointer user_data);
-
-
-struct _GstAudioClock {
-  GstSystemClock clock;
-
-  GstClockTime prev1, prev2;
-
-  /* --- protected --- */
-  GstAudioClockGetTimeFunc func;
-  gpointer user_data;
-
-  GstClockTimeDiff adjust;
-
-  GSList *async_entries;
-
-  gboolean active;
-
-  gpointer _gst_reserved[GST_PADDING];
-};
-
-struct _GstAudioClockClass {
-  GstSystemClockClass parent_class;
-
-  gpointer _gst_reserved[GST_PADDING];
-};
-
-GType           gst_audio_clock_get_type 	(void);
-GstClock*	gst_audio_clock_new		(gchar *name, GstAudioClockGetTimeFunc func,
-                                                 gpointer user_data);
-void		gst_audio_clock_set_active 	(GstAudioClock *aclock, gboolean active);
-
-void		gst_audio_clock_update_time	(GstAudioClock *aclock, GstClockTime time);
-
-G_END_DECLS
-
-#endif /* __GST_AUDIO_CLOCK_H__ */
diff --git a/gst-libs/gst/audio/audiofilter.vcproj b/gst-libs/gst/audio/audiofilter.vcproj
deleted file mode 100644
index d5512606..00000000
--- a/gst-libs/gst/audio/audiofilter.vcproj
+++ /dev/null
@@ -1,144 +0,0 @@
-<?xml version="1.0" encoding="Windows-1252"?>
-<VisualStudioProject
-	ProjectType="Visual C++"
-	Version="7.10"
-	Name="audiofilter"
-	ProjectGUID="{979C216F-0ACF-4956-AE00-055A42D67894}"
-	RootNamespace="audiofilter"
-	Keyword="Win32Proj">
-	<Platforms>
-		<Platform
-			Name="Win32"/>
-	</Platforms>
-	<Configurations>
-		<Configuration
-			Name="Debug|Win32"
-			OutputDirectory="../../../win32/Debug"
-			IntermediateDirectory="../../../win32/Debug"
-			ConfigurationType="2"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				Optimization="0"
-				AdditionalIncludeDirectories="../../../../gstreamer/win32;../../../../gstreamer;../../../../gstreamer/libs;../../../../glib;../../../../glib/glib;../../../../glib/gmodule;&quot;../../../gst-libs&quot;;../../../../popt/include;../../../../libxml2/include/libxml2"
-				PreprocessorDefinitions="WIN32;_DEBUG;_WINDOWS;_USRDLL;AUDIOFILTER_EXPORTS;HAVE_CONFIG_H;_USE_MATH_DEFINES"
-				MinimalRebuild="TRUE"
-				BasicRuntimeChecks="3"
-				RuntimeLibrary="3"
-				UsePrecompiledHeader="0"
-				WarningLevel="3"
-				Detect64BitPortabilityProblems="TRUE"
-				DebugInformationFormat="4"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				Name="VCLinkerTool"
-				AdditionalDependencies="glib-2.0.lib gmodule-2.0.lib gthread-2.0.lib gobject-2.0.lib libgstreamer.lib gstbytestream.lib iconv.lib intl.lib"
-				OutputFile="$(OutDir)/gstaudiofilter.dll"
-				LinkIncremental="2"
-				AdditionalLibraryDirectories="../../../../gstreamer/win32/Debug;../../../../glib/glib;../../../../glib/gmodule;../../../../glib/gthread;../../../../glib/gobject;../../../../gettext/lib;../../../../libiconv/lib"
-				ModuleDefinitionFile="audio.def"
-				GenerateDebugInformation="TRUE"
-				ProgramDatabaseFile="$(OutDir)/audiofilter.pdb"
-				SubSystem="2"
-				OptimizeReferences="2"
-				ImportLibrary="$(OutDir)/gstaudiofilter.lib"
-				TargetMachine="1"/>
-			<Tool
-				Name="VCMIDLTool"/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy /Y $(TargetPath) c:\gstreamer\plugins"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCWebDeploymentTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-		<Configuration
-			Name="Release|Win32"
-			OutputDirectory="../../../win32/Release"
-			IntermediateDirectory="../../../win32/Release"
-			ConfigurationType="2"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../gstreamer/win32;../../../../gstreamer;../../../../gstreamer/libs;../../../../glib;../../../../glib/glib;../../../../glib/gmodule;&quot;../../../gst-libs&quot;;../../../../popt/include;../../../../libxml2/include/libxml2"
-				PreprocessorDefinitions="WIN32;NDEBUG;GST_DISABLE_GST_DEBUG;_WINDOWS;_USRDLL;AUDIOFILTER_EXPORTS;HAVE_CONFIG_H;_USE_MATH_DEFINES"
-				RuntimeLibrary="2"
-				UsePrecompiledHeader="0"
-				WarningLevel="3"
-				Detect64BitPortabilityProblems="TRUE"
-				DebugInformationFormat="3"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				Name="VCLinkerTool"
-				AdditionalDependencies="glib-2.0.lib gmodule-2.0.lib gthread-2.0.lib gobject-2.0.lib libgstreamer.lib gstbytestream.lib iconv.lib intl.lib"
-				OutputFile="$(OutDir)/gstaudiofilter.dll"
-				LinkIncremental="1"
-				AdditionalLibraryDirectories="../../../../gstreamer/win32/Release;../../../../glib/glib;../../../../glib/gmodule;../../../../glib/gthread;../../../../glib/gobject;../../../../gettext/lib;../../../../libiconv/lib"
-				ModuleDefinitionFile="audio.def"
-				GenerateDebugInformation="TRUE"
-				SubSystem="2"
-				OptimizeReferences="2"
-				EnableCOMDATFolding="2"
-				ImportLibrary="$(OutDir)/gstaudiofilter.lib"
-				TargetMachine="1"/>
-			<Tool
-				Name="VCMIDLTool"/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy /Y $(TargetPath) c:\gstreamer\plugins"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCWebDeploymentTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-	</Configurations>
-	<References>
-	</References>
-	<Files>
-		<Filter
-			Name="Source Files"
-			Filter="cpp;c;cxx;def;odl;idl;hpj;bat;asm;asmx"
-			UniqueIdentifier="{4FC737F1-C7A5-4376-A066-2A32D752A2FF}">
-			<File
-				RelativePath=".\gstaudiofilter.c">
-			</File>
-		</Filter>
-		<Filter
-			Name="Header Files"
-			Filter="h;hpp;hxx;hm;inl;inc;xsd"
-			UniqueIdentifier="{93995380-89BD-4b04-88EB-625FBE52EBFB}">
-			<File
-				RelativePath=".\gstaudiofilter.h">
-			</File>
-		</Filter>
-	</Files>
-	<Globals>
-	</Globals>
-</VisualStudioProject>
diff --git a/gst-libs/gst/audio/gstaudiofilter.c b/gst-libs/gst/audio/gstaudiofilter.c
deleted file mode 100644
index 70ae6bf9..00000000
--- a/gst-libs/gst/audio/gstaudiofilter.c
+++ /dev/null
@@ -1,313 +0,0 @@
-/* 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 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));
-
-  ret = GST_PAD_LINK_DELAYED;   /* intialise with dummy value */
-  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 (GST_PAD_LINK_FAILED (link_ret)) {
-    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);
-  } else if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float")
-      == 0) {
-    ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
-  } else {
-    g_assert_not_reached ();
-  }
-  ret &= gst_structure_get_int (structure, "rate", &audiofilter->rate);
-  ret &= gst_structure_get_int (structure, "channels", &audiofilter->channels);
-
-  if (!ret)
-    return GST_PAD_LINK_REFUSED;
-
-  audiofilter->bytes_per_sample = (audiofilter->width / 8) *
-      audiofilter->channels;
-
-  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_pad_proxy_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_pad_proxy_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;
-  }
-
-  audiofilter->size = GST_BUFFER_SIZE (inbuf);
-  audiofilter->n_samples = audiofilter->size / audiofilter->bytes_per_sample;
-
-  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
deleted file mode 100644
index 9786e16c..00000000
--- a/gst-libs/gst/audio/gstaudiofilter.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/* 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;
-  gboolean passthru;
-
-  int rate;
-  int width;
-  int channels;
-  int depth;
-
-  int n_samples;
-  int size;
-  int bytes_per_sample;
-};
-
-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/gstaudiofiltertemplate.c b/gst-libs/gst/audio/gstaudiofiltertemplate.c
deleted file mode 100644
index f1852037..00000000
--- a/gst-libs/gst/audio/gstaudiofiltertemplate.c
+++ /dev/null
@@ -1,270 +0,0 @@
-/* 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.
- */
-
-/*
- * This file was (probably) generated from
- * $Id$
- * and
- * MAKEFILTERVERSION
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-#include <gst/audio/gstaudiofilter.h>
-#include <string.h>
-
-typedef struct _GstAudiofilterTemplate GstAudiofilterTemplate;
-typedef struct _GstAudiofilterTemplateClass GstAudiofilterTemplateClass;
-
-#define GST_TYPE_AUDIOFILTER_TEMPLATE \
-  (gst_audiofilter_template_get_type())
-#define GST_AUDIOFILTER_TEMPLATE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIOFILTER_TEMPLATE,GstAudiofilterTemplate))
-#define GST_AUDIOFILTER_TEMPLATE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIOFILTER_TEMPLATE,GstAudiofilterTemplateClass))
-#define GST_IS_AUDIOFILTER_TEMPLATE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIOFILTER_TEMPLATE))
-#define GST_IS_AUDIOFILTER_TEMPLATE_CLASS(obj) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOFILTER_TEMPLATE))
-
-struct _GstAudiofilterTemplate
-{
-  GstAudiofilter audiofilter;
-
-};
-
-struct _GstAudiofilterTemplateClass
-{
-  GstAudiofilterClass parent_class;
-
-};
-
-
-enum
-{
-  /* FILL ME */
-  LAST_SIGNAL
-};
-
-enum
-{
-  ARG_0
-      /* FILL ME */
-};
-
-static void gst_audiofilter_template_base_init (gpointer g_class);
-static void gst_audiofilter_template_class_init (gpointer g_class,
-    gpointer class_data);
-static void gst_audiofilter_template_init (GTypeInstance * instance,
-    gpointer g_class);
-
-static void gst_audiofilter_template_set_property (GObject * object,
-    guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_audiofilter_template_get_property (GObject * object,
-    guint prop_id, GValue * value, GParamSpec * pspec);
-
-static void gst_audiofilter_template_setup (GstAudiofilter * audiofilter);
-static void gst_audiofilter_template_filter (GstAudiofilter * audiofilter,
-    GstBuffer * outbuf, GstBuffer * inbuf);
-static void gst_audiofilter_template_filter_inplace (GstAudiofilter *
-    audiofilter, GstBuffer * buf);
-
-GType
-gst_audiofilter_template_get_type (void)
-{
-  static GType audiofilter_template_type = 0;
-
-  if (!audiofilter_template_type) {
-    static const GTypeInfo audiofilter_template_info = {
-      sizeof (GstAudiofilterTemplateClass),
-      gst_audiofilter_template_base_init,
-      NULL,
-      gst_audiofilter_template_class_init,
-      NULL,
-      gst_audiofilter_template_init,
-      sizeof (GstAudiofilterTemplate),
-      0,
-      NULL,
-    };
-
-    audiofilter_template_type = g_type_register_static (GST_TYPE_AUDIOFILTER,
-        "GstAudiofilterTemplate", &audiofilter_template_info, 0);
-  }
-  return audiofilter_template_type;
-}
-
-static void
-gst_audiofilter_template_base_init (gpointer g_class)
-{
-  static GstElementDetails audiofilter_template_details = {
-    "Audio filter template",
-    "Filter/Effect/Audio",
-    "Filters audio",
-    "David Schleef <ds@schleef.org>"
-  };
-  GstAudiofilterTemplateClass *klass = (GstAudiofilterTemplateClass *) g_class;
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_set_details (element_class, &audiofilter_template_details);
-
-  gst_audiofilter_class_add_pad_templates (GST_AUDIOFILTER_CLASS (g_class),
-      gst_caps_from_string (GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS));
-}
-
-static void
-gst_audiofilter_template_class_init (gpointer g_class, gpointer class_data)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-  GstAudiofilterTemplateClass *klass;
-  GstAudiofilterClass *audiofilter_class;
-
-  klass = (GstAudiofilterTemplateClass *) g_class;
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-  audiofilter_class = (GstAudiofilterClass *) g_class;
-
-#if 0
-  g_object_class_install_property (gobject_class, ARG_METHOD,
-      g_param_spec_enum ("method", "method", "method",
-          GST_TYPE_AUDIOTEMPLATE_METHOD, GST_AUDIOTEMPLATE_METHOD_1,
-          G_PARAM_READWRITE));
-#endif
-
-  gobject_class->set_property = gst_audiofilter_template_set_property;
-  gobject_class->get_property = gst_audiofilter_template_get_property;
-
-  audiofilter_class->setup = gst_audiofilter_template_setup;
-  audiofilter_class->filter = gst_audiofilter_template_filter;
-  audiofilter_class->filter_inplace = gst_audiofilter_template_filter_inplace;
-  audiofilter_class->filter = NULL;
-}
-
-static void
-gst_audiofilter_template_init (GTypeInstance * instance, gpointer g_class)
-{
-  //GstAudiofilterTemplate *audiofilter_template = GST_AUDIOFILTER_TEMPLATE (instance);
-  //GstAudiofilter *audiofilter = GST_AUDIOFILTER (instance);
-
-  GST_DEBUG ("gst_audiofilter_template_init");
-
-  /* do stuff */
-
-}
-
-static void
-gst_audiofilter_template_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstAudiofilterTemplate *src;
-
-  /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (object));
-  src = GST_AUDIOFILTER_TEMPLATE (object);
-
-  GST_DEBUG ("gst_audiofilter_template_set_property");
-  switch (prop_id) {
-    default:
-      break;
-  }
-}
-
-static void
-gst_audiofilter_template_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstAudiofilterTemplate *src;
-
-  /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (object));
-  src = GST_AUDIOFILTER_TEMPLATE (object);
-
-  switch (prop_id) {
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  if (!gst_library_load ("gstaudiofilter"))
-    return FALSE;
-
-  return gst_element_register (plugin, "audiofiltertemplate", GST_RANK_NONE,
-      GST_TYPE_AUDIOFILTER_TEMPLATE);
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    "gstaudiofilter_template",
-    "Audio filter template",
-    plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
-
-     static void gst_audiofilter_template_setup (GstAudiofilter * audiofilter)
-{
-  GstAudiofilterTemplate *audiofilter_template;
-
-  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (audiofilter));
-  audiofilter_template = GST_AUDIOFILTER_TEMPLATE (audiofilter);
-
-  /* if any setup needs to be done, do it here */
-
-}
-
-/* You may choose to implement either a copying filter or an
- * in-place filter (or both).  Implementing only one will give
- * full functionality, however, implementing both will cause
- * audiofilter to use the optimal function in every situation,
- * with a minimum of memory copies. */
-
-static void
-gst_audiofilter_template_filter (GstAudiofilter * audiofilter,
-    GstBuffer * outbuf, GstBuffer * inbuf)
-{
-  GstAudiofilterTemplate *audiofilter_template;
-
-  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (audiofilter));
-  audiofilter_template = GST_AUDIOFILTER_TEMPLATE (audiofilter);
-
-  /* do something interesting here.  This simply copies the source
-   * to the destination. */
-
-  memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf), audiofilter->size);
-}
-
-static void
-gst_audiofilter_template_filter_inplace (GstAudiofilter * audiofilter,
-    GstBuffer * buf)
-{
-  GstAudiofilterTemplate *audiofilter_template;
-
-  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (audiofilter));
-  audiofilter_template = GST_AUDIOFILTER_TEMPLATE (audiofilter);
-
-  /* do something interesting here.  This simply copies the source
-   * to the destination. */
-
-}
diff --git a/gst-libs/gst/audio/make_filter b/gst-libs/gst/audio/make_filter
deleted file mode 100755
index eaeaa060..00000000
--- a/gst-libs/gst/audio/make_filter
+++ /dev/null
@@ -1,42 +0,0 @@
-#!/bin/sh
-
-LANG=C
-LC_COLLATE=C
-export LANG
-export LC_COLLATE
-
-Template=$1;
-srcfile=$2;
-
-if test x"$1" = x ; then
-	echo "$0 Objectname [srcfile]\n";
-	echo "  creates gstobjectname.{c,h} implementing GstObjectname,\n";
-	echo "  subclassing GstAudiofilter.\n";
-	exit 1;
-fi
-
-if test x"$2" = x ; then
-	srcfile="gstaudiofiltertemplate.c"
-fi
-
-id=`echo '$Id$' | sed \
-	-e 's/\$I[d]: \([^$]*\)\$/\1/g'`
-echo $id
-
-TEMPLATE=`echo $Template | tr [:lower:] [:upper:]`
-template=`echo $Template | tr [:upper:] [:lower:]`
-
-echo TEMPLATE $TEMPLATE
-echo template $template
-
-# remember to break up the Id: in the line below
-sed \
-	-e 's/gstaudiofiltertemplate\.c/SOURCEFILE/g' \
-	-e "s/AudiofilterTemplate/$Template/g" \
-	-e "s/audiofiltertemplate/$template/g" \
-	-e "s/VIDEOFILTERTEMPLATE/$TEMPLATE/g" \
-	-e 's/\$I[d]: \([^$]*\)\$/\1/g' \
-	-e 's/SOURCEFILE/gstaudiofiltertemplate\.c/g' \
-	-e "s%MAKEFILTERVERSION%$id%g" \
-	$srcfile >gst$template.c.tmp && mv gst$template.c.tmp gst$template.c
-
diff --git a/gst-libs/gst/audio/multichannel.c b/gst-libs/gst/audio/multichannel.c
deleted file mode 100644
index 67c055d9..00000000
--- a/gst-libs/gst/audio/multichannel.c
+++ /dev/null
@@ -1,634 +0,0 @@
-/* GStreamer Multichannel-Audio helper functions
- * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.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.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include "multichannel.h"
-
-#define GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME "channel-positions"
-
-/*
- * This function checks if basic assumptions apply:
- *  - does each position occur at most once?
- *  - do conflicting positions occur?
- *     + front_mono vs. front_left/right
- *     + front_center vs. front_left/right_of_center
- *     + rear_center vs. rear_left/right
- * It also adds some hacks that 0.8.x needs for compatibility:
- *  - if channels == 1, are we really mono?
- *  - if channels == 2, are we really stereo?
- */
-
-static gboolean
-gst_audio_check_channel_positions (const GstAudioChannelPosition * pos,
-    gint channels)
-{
-  gint i, n;
-  struct
-  {
-    GstAudioChannelPosition pos1[2];
-    GstAudioChannelPosition pos2[1];
-  } conf[] = {
-    /* front: mono <-> stereo */
-    { {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_MONO}},
-        /* front center: 2 <-> 1 */
-    { {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
-            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}},
-        /* rear: 2 <-> 1 */
-    { {
-    GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-            GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
-    GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}}, { {
-    GST_AUDIO_CHANNEL_POSITION_INVALID}}
-  };
-
-  /* check for invalid channel positions */
-  for (n = 0; n < channels; n++) {
-    if (pos[n] == GST_AUDIO_CHANNEL_POSITION_INVALID) {
-      g_warning ("Position %d is invalid, not allowed", n);
-      return FALSE;
-    }
-  }
-
-  /* check for multiple position occurrences */
-  for (i = GST_AUDIO_CHANNEL_POSITION_INVALID + 1;
-      i < GST_AUDIO_CHANNEL_POSITION_NUM; i++) {
-    gint count = 0;
-
-    for (n = 0; n < channels; n++) {
-      if (pos[n] == i)
-        count++;
-    }
-
-    if (count > 1) {
-      g_warning ("Channel position %d occurred %d times, not allowed",
-          i, count);
-      return FALSE;
-    }
-  }
-
-  /* check for position conflicts */
-  for (i = 0; conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; i++) {
-    gboolean found1 = FALSE, found2 = FALSE;
-
-    for (n = 0; n < channels; n++) {
-      if (pos[n] == conf[i].pos1[0] || pos[n] == conf[i].pos1[1])
-        found1 = TRUE;
-      else if (pos[n] == conf[i].pos2[0])
-        found2 = TRUE;
-    }
-
-    if (found1 && found2) {
-      g_warning ("Found conflicting channel positions %d/%d and %d",
-          conf[i].pos1[0], conf[i].pos1[1], conf[i].pos2[0]);
-      return FALSE;
-    }
-  }
-
-  /* 0.8.x evilry */
-  if ((channels == 1 && pos[0] != GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) ||
-      (channels == 2 && (pos[0] != GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT ||
-              pos[1] != GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT))) {
-    g_warning ("0.8.x: channels=1 implies mono; channels=2 implies stereo");
-    return FALSE;
-  }
-
-  return TRUE;
-}
-
-/**
- * gst_audio_get_channel_positions:
- * @str: A #GstStructure to retrieve channel positions from.
- *
- * Retrieves a number of (fixed!) audio channel positions from
- * the provided #GstStructure and returns it as a newly allocated
- * array. The caller should g_free () this array. The caller
- * should also check that the members in this #GstStructure are
- * indeed "fixed" before calling this function.
- *
- * Returns: a newly allocated array containing the channel
- * positions as provided in the given #GstStructure. Returns
- * NULL on error.
- */
-
-GstAudioChannelPosition *
-gst_audio_get_channel_positions (GstStructure * str)
-{
-  GstAudioChannelPosition *pos;
-  gint channels, n;
-  const GValue *pos_val_arr, *pos_val_entry;
-  gboolean res;
-  GType t;
-
-  /* get number of channels, general type checkups */
-  g_return_val_if_fail (str != NULL, NULL);
-  res = gst_structure_get_int (str, "channels", &channels);
-  g_return_val_if_fail (res, NULL);
-  g_return_val_if_fail (channels > 0, NULL);
-  pos_val_arr = gst_structure_get_value (str,
-      GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME);
-
-  /* The following checks are here to retain compatibility for plugins not
-   * implementing this property. They expect that channels=1 implies mono
-   * and channels=2 implies stereo, so we follow that.
-   * This might be removed during 0.9.x. */
-  if (!pos_val_arr && (channels == 1 || channels == 2)) {
-    pos = g_new (GstAudioChannelPosition, channels);
-    if (channels == 1) {
-      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
-    } else {
-      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
-      pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
-    }
-    return pos;
-  }
-  g_return_val_if_fail (pos_val_arr != NULL, NULL);
-  g_return_val_if_fail (gst_value_list_get_size (pos_val_arr) == channels,
-      NULL);
-  for (n = 0; n < channels; n++) {
-    t = G_VALUE_TYPE (gst_value_list_get_value (pos_val_arr, n));
-    g_return_val_if_fail (t == GST_TYPE_AUDIO_CHANNEL_POSITION, NULL);
-  }
-
-  /* ... and fill array */
-  pos = g_new (GstAudioChannelPosition, channels);
-  for (n = 0; n < channels; n++) {
-    pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
-    pos[n] = g_value_get_enum (pos_val_entry);
-  }
-
-  if (!gst_audio_check_channel_positions (pos, channels)) {
-    g_free (pos);
-    return NULL;
-  }
-
-  return pos;
-}
-
-/**
- * gst_audio_set_channel_positions:
- * @str: A #GstStructure to retrieve channel positions from.
- * @pos: an array of channel positions. The number of members
- *       in this array should be equal to the (fixed!) number
- *       of the "channels" property in the given #GstStructure.
- *
- * Adds a "channel-positions" property to the given #GstStructure,
- * which will represent the channel positions as given in the
- * provided #GstAudioChannelPosition array.
- */
-
-void
-gst_audio_set_channel_positions (GstStructure * str,
-    const GstAudioChannelPosition * pos)
-{
-  GValue pos_val_arr = { 0 }, pos_val_entry = {
-  0};
-  gint channels, n;
-  gboolean res;
-
-  /* get number of channels, checkups */
-  g_return_if_fail (str != NULL);
-  g_return_if_fail (pos != NULL);
-  res = gst_structure_get_int (str, "channels", &channels);
-  g_return_if_fail (res);
-  g_return_if_fail (channels > 0);
-  if (!gst_audio_check_channel_positions (pos, channels))
-    return;
-
-  /* build gvaluearray from positions */
-  g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
-  g_value_init (&pos_val_arr, GST_TYPE_FIXED_LIST);
-  for (n = 0; n < channels; n++) {
-    g_value_set_enum (&pos_val_entry, pos[n]);
-    gst_value_list_append_value (&pos_val_arr, &pos_val_entry);
-  }
-  g_value_unset (&pos_val_entry);
-
-  /* add to structure */
-  gst_structure_set_value (str,
-      GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME, &pos_val_arr);
-  g_value_unset (&pos_val_arr);
-}
-
-/**
- * gst_audio_set_structure_channel_positions_list:
- * @str: #GstStructure to set the list of channel positions
- *       on.
- * @pos: the array containing one or more possible audio
- *       channel positions that we should add in each value
- *       of the array in the given structure.
- * @num_positions: the number of values in pos.
- *
- * Sets a (possibly non-fixed) list of possible audio channel
- * positions (given in pos) on the given structure. The
- * structure, after this function has been called, will contain
- * a "channel-positions" property with an array of the size of
- * the "channels" property value in the given structure (note
- * that this means that the channels property in the provided
- * structure should be fixed!). Each value in the array will
- * contain each of the values given in the pos array.
- */
-
-void
-gst_audio_set_structure_channel_positions_list (GstStructure * str,
-    const GstAudioChannelPosition * pos, gint num_positions)
-{
-  gint channels, n, c;
-  GValue pos_val_arr = { 0 }, pos_val_list = {
-  0}, pos_val_entry = {
-  0};
-  gboolean res;
-
-  /* get number of channels, general type checkups */
-  g_return_if_fail (str != NULL);
-  g_return_if_fail (num_positions > 0);
-  g_return_if_fail (pos != NULL);
-  res = gst_structure_get_int (str, "channels", &channels);
-  g_return_if_fail (res);
-  g_return_if_fail (channels > 0);
-
-  /* 0.8.x: channels=1 or channels=2 is mono/stereo, no positions needed
-   * there (we discard them anyway) */
-  if (channels == 1 || channels == 2)
-    return;
-
-  /* create the array of lists */
-  g_value_init (&pos_val_arr, GST_TYPE_FIXED_LIST);
-  g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
-  for (n = 0; n < channels; n++) {
-    g_value_init (&pos_val_list, GST_TYPE_LIST);
-    for (c = 0; c < num_positions; c++) {
-      g_value_set_enum (&pos_val_entry, pos[c]);
-      gst_value_list_append_value (&pos_val_list, &pos_val_entry);
-    }
-    gst_value_list_append_value (&pos_val_arr, &pos_val_list);
-    g_value_unset (&pos_val_list);
-  }
-  g_value_unset (&pos_val_entry);
-  gst_structure_set_value (str, GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME,
-      &pos_val_arr);
-  g_value_unset (&pos_val_arr);
-}
-
-/*
- * Helper function for below. The structure will be conserved,
- * but might be cut down. Any additional structures that were
- * created will be stored in the returned caps.
- */
-
-static GstCaps *
-add_list_to_struct (GstStructure * str,
-    const GstAudioChannelPosition * pos, gint num_positions)
-{
-  GstCaps *caps = gst_caps_new_empty ();
-  const GValue *chan_val;
-
-  chan_val = gst_structure_get_value (str, "channels");
-  if (G_VALUE_TYPE (chan_val) == G_TYPE_INT) {
-    gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
-  } else if (G_VALUE_TYPE (chan_val) == GST_TYPE_LIST) {
-    gint size;
-    const GValue *sub_val;
-
-    size = gst_value_list_get_size (chan_val);
-    sub_val = gst_value_list_get_value (chan_val, 0);
-    gst_structure_set_value (str, "channels", sub_val);
-    gst_caps_append (caps, add_list_to_struct (str, pos, num_positions));
-    while (--size > 0) {
-      str = gst_structure_copy (str);
-      sub_val = gst_value_list_get_value (chan_val, size);
-      gst_structure_set_value (str, "channels", sub_val);
-      gst_caps_append (caps, add_list_to_struct (str, pos, num_positions));
-      gst_caps_append_structure (caps, str);
-    }
-  } else if (G_VALUE_TYPE (chan_val) == GST_TYPE_INT_RANGE) {
-    gint min, max;
-
-    min = gst_value_get_int_range_min (chan_val);
-    max = gst_value_get_int_range_max (chan_val);
-
-    gst_structure_set (str, "channels", G_TYPE_INT, min, NULL);
-    gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
-    for (++min; min < max; min++) {
-      str = gst_structure_copy (str);
-      gst_structure_set (str, "channels", G_TYPE_INT, min, NULL);
-      gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
-      gst_caps_append_structure (caps, str);
-    }
-  } else {
-    g_warning ("Unknown value type for channels property");
-  }
-
-  return caps;
-}
-
-/**
- * gst_audio_set_caps_channel_positions_list:
- * @caps: #GstCaps to set the list of channel positions on.
- * @pos: the array containing one or more possible audio
- *       channel positions that we should add in each value
- *       of the array in the given structure.
- * @num_positions: the number of values in pos.
- *
- * Sets a (possibly non-fixed) list of possible audio channel
- * positions (given in pos) on the given caps. Each of the
- * structures of the caps, after this function has been called,
- * will contain a "channel-positions" property with an array.
- * Each value in the array will contain each of the values given
- * in the pos array. Note that the size of the caps might be
- * increased by this, since each structure with a "channel-
- * positions" property needs to have a fixed "channels" property.
- * The input caps is not required to have this.
- */
-
-void
-gst_audio_set_caps_channel_positions_list (GstCaps * caps,
-    const GstAudioChannelPosition * pos, gint num_positions)
-{
-  gint size, n;
-
-  /* get number of channels, general type checkups */
-  g_return_if_fail (caps != NULL);
-  g_return_if_fail (num_positions > 0);
-  g_return_if_fail (pos != NULL);
-
-  size = gst_caps_get_size (caps);
-  for (n = 0; n < size; n++) {
-    gst_caps_append (caps, add_list_to_struct (gst_caps_get_structure (caps,
-                n), pos, num_positions));
-  }
-}
-
-/**
- * gst_audio_fixate_channel_positions:
- * @str: a #GstStructure containing a (possibly unfixed)
- *       "channel-positions" property.
- *
- * Custom fixate function. Elements that implement some sort of
- * channel conversion algorhithm should use this function for
- * fixating on GstAudioChannelPosition properties. It will take
- * care of equal channel positioning (left/right). Caller g_free()s
- * the return value. The input properties may be (and are supposed
- * to be) unfixed.
- * Note that this function is mostly a hack because we currently
- * have no way to add default fixation functions for new GTypes.
- *
- * Returns: fixed values that the caller could use as a fixed
- * set of #GstAudioChannelPosition values.
- */
-
-GstAudioChannelPosition *
-gst_audio_fixate_channel_positions (GstStructure * str)
-{
-  GstAudioChannelPosition *pos;
-  gint channels, n, num_unfixed = 0, i, c;
-  const GValue *pos_val_arr, *pos_val_entry, *pos_val;
-  gboolean res, is_stereo = TRUE;
-  GType t;
-
-  /*
-   * We're going to do this cluelessly. We'll make an array of values that
-   * conflict with each other and, for each iteration in this array, pick
-   * either one until all unknown values are filled. This might not work in
-   * corner cases but should work OK for the general case.
-   */
-  struct
-  {
-    GstAudioChannelPosition pos1[2];
-    GstAudioChannelPosition pos2[1];
-  } conf[] = {
-    /* front: mono <-> stereo */
-    { {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_MONO}},
-        /* front center: 2 <-> 1 */
-    { {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
-            GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
-    GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}},
-        /* rear: 2 <-> 1 */
-    { {
-    GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-            GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
-    GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}}, { {
-    GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
-    GST_AUDIO_CHANNEL_POSITION_LFE}}, { {
-    GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
-            GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
-    GST_AUDIO_CHANNEL_POSITION_INVALID}}, { {
-    GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
-    GST_AUDIO_CHANNEL_POSITION_INVALID}}
-  };
-  struct
-  {
-    gint num_opt[3];
-    guint num_opts[3];
-    gboolean is_fixed[3];
-    gint choice;                /* -1 is none, 0 is the two, 1 is the one */
-  } opt;
-
-  /* get number of channels, general type checkups */
-  g_return_val_if_fail (str != NULL, NULL);
-  res = gst_structure_get_int (str, "channels", &channels);
-  g_return_val_if_fail (res, NULL);
-  g_return_val_if_fail (channels > 0, NULL);
-
-  /* 0.8.x mono/stereo checks */
-  pos_val_arr = gst_structure_get_value (str,
-      GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME);
-  if (!pos_val_arr && (channels == 1 || channels == 2)) {
-    pos = g_new (GstAudioChannelPosition, channels);
-    if (channels == 1) {
-      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
-    } else {
-      pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
-      pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
-    }
-    return pos;
-  }
-  g_return_val_if_fail (pos_val_arr != NULL, NULL);
-  g_return_val_if_fail (gst_value_list_get_size (pos_val_arr) == channels,
-      NULL);
-  for (n = 0; n < channels; n++) {
-    t = G_VALUE_TYPE (gst_value_list_get_value (pos_val_arr, n));
-    g_return_val_if_fail (t == GST_TYPE_LIST ||
-        t == GST_TYPE_AUDIO_CHANNEL_POSITION, NULL);
-  }
-
-  /* all unknown, to start with */
-  pos = g_new (GstAudioChannelPosition, channels);
-  for (n = 0; n < channels; n++)
-    pos[n] = GST_AUDIO_CHANNEL_POSITION_INVALID;
-  num_unfixed = channels;
-
-  /* Iterate the array of conflicting values */
-  for (i = 0; conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID ||
-      conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; i++) {
-    /* front/center only important if not mono (obviously) */
-    if (conf[i].pos1[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER &&
-        !is_stereo)
-      continue;
-
-    /* init values */
-    for (n = 0; n < 3; n++) {
-      opt.num_opt[n] = -1;
-      opt.num_opts[n] = -1;
-      opt.is_fixed[n] = FALSE;
-    }
-
-    /* Now, we'll see for each channel if it allows for any of the values in
-     * the set of conflicting audio channel positions and keep scores. */
-    for (n = 0; n < channels; n++) {
-      /* if the channel is already taken, don't bother */
-      if (pos[n] != GST_AUDIO_CHANNEL_POSITION_INVALID)
-        continue;
-
-      pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
-      t = G_VALUE_TYPE (pos_val_entry);
-      if (t == GST_TYPE_LIST) {
-        /* This algorhythm is suboptimal. */
-        for (c = 0; c < gst_value_list_get_size (pos_val_entry); c++) {
-          pos_val = gst_value_list_get_value (pos_val_entry, c);
-          if (g_value_get_enum (pos_val) == conf[i].pos1[0] &&
-              opt.num_opts[0] > gst_value_list_get_size (pos_val_entry) &&
-              !opt.is_fixed[0]) {
-            /* Now test if the old position of num_opt[0] also allows for
-             * the other channel (which was skipped previously). If so,
-             * keep score. */
-            if (opt.num_opt[0] != -1) {
-              gint c1;
-
-              pos_val_entry = gst_value_list_get_value (pos_val_arr,
-                  opt.num_opt[0]);
-              if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) {
-                for (c1 = 0; c1 < gst_value_list_get_size (pos_val_entry); c1++) {
-                  pos_val = gst_value_list_get_value (pos_val_entry, c1);
-                  if (g_value_get_enum (pos_val) == conf[i].pos1[1] &&
-                      opt.num_opts[1] > opt.num_opts[0] && !opt.is_fixed[1]) {
-                    opt.num_opts[1] = opt.num_opts[0];
-                    opt.num_opt[1] = opt.num_opt[0];
-                  }
-                }
-                pos_val = gst_value_list_get_value (pos_val_entry, c);
-              }
-              pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
-            }
-
-            /* and save values */
-            opt.num_opts[0] = gst_value_list_get_size (pos_val_entry);
-            opt.num_opt[0] = n;
-          } else if (g_value_get_enum (pos_val) == conf[i].pos1[1] &&
-              opt.num_opts[1] > gst_value_list_get_size (pos_val_entry) &&
-              !opt.is_fixed[1] && n != opt.num_opt[0]) {
-            opt.num_opts[1] = gst_value_list_get_size (pos_val_entry);
-            opt.num_opt[1] = n;
-          }
-
-          /* 2 goes separately, because 0/1 vs. 2 are separate */
-          if (g_value_get_enum (pos_val) == conf[i].pos2[0] &&
-              opt.num_opts[2] > gst_value_list_get_size (pos_val_entry) &&
-              !opt.is_fixed[2]) {
-            opt.num_opts[2] = gst_value_list_get_size (pos_val_entry);
-            opt.num_opt[2] = n;
-          }
-        }
-      } else {
-        if (g_value_get_enum (pos_val_entry) == conf[i].pos1[0]) {
-          opt.num_opt[0] = n;
-          opt.is_fixed[0] = TRUE;
-        } else if (g_value_get_enum (pos_val_entry) == conf[i].pos1[1]) {
-          opt.num_opt[1] = n;
-          opt.is_fixed[1] = TRUE;
-        } else if (g_value_get_enum (pos_val_entry) == conf[i].pos2[0]) {
-          opt.num_opt[2] = n;
-          opt.is_fixed[2] = TRUE;
-        }
-      }
-    }
-
-    /* check our results and choose either one */
-    if ((opt.is_fixed[0] || opt.is_fixed[1]) && opt.is_fixed[2]) {
-      g_warning ("Pre-fixated on both %d/%d and %d - conflict!",
-          conf[i].pos1[0], conf[i].pos1[1], conf[i].pos2[0]);
-      g_free (pos);
-      return NULL;
-    } else if ((opt.is_fixed[0] && opt.num_opt[1] == -1) ||
-        (opt.is_fixed[1] && opt.num_opt[0] == -1)) {
-      g_warning ("Pre-fixated one side, but other side n/a of %d/%d",
-          conf[i].pos1[0], conf[i].pos1[1]);
-      g_free (pos);
-      return NULL;
-    } else if (opt.is_fixed[0] || opt.is_fixed[1]) {
-      opt.choice = 0;
-    } else if (opt.is_fixed[2]) {
-      opt.choice = 1;
-    } else if (opt.num_opt[0] != -1 && opt.num_opt[1] != -1) {
-      opt.choice = 0;
-    } else if (opt.num_opt[2] != -1) {
-      opt.choice = 1;
-    } else {
-      opt.choice = -1;
-    }
-
-    /* stereo? Note that we keep is_stereo to TRUE if we didn't decide on
-     * any arrangement. The mono/stereo channels might be handled elsewhere
-     * which is clearly outside the scope of this element, so we cannot
-     * know and expect the application to handle that then. */
-    if (conf[i].pos2[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO &&
-        opt.choice == 1) {
-      is_stereo = FALSE;
-    }
-
-    /* now actually decide what we'll do and fixate on that */
-    if (opt.choice == 0) {
-      g_assert (conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID &&
-          conf[i].pos1[1] != GST_AUDIO_CHANNEL_POSITION_INVALID);
-      pos[opt.num_opt[0]] = conf[i].pos1[0];
-      pos[opt.num_opt[1]] = conf[i].pos1[1];
-      num_unfixed -= 2;
-    } else if (opt.choice == 1) {
-      g_assert (conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID);
-      pos[opt.num_opt[2]] = conf[i].pos2[0];
-      num_unfixed--;
-    }
-  }
-
-  /* safety check */
-  if (num_unfixed > 0) {
-    g_warning ("%d unfixed channel positions left after fixation!",
-        num_unfixed);
-    g_free (pos);
-    return NULL;
-  }
-
-  if (!gst_audio_check_channel_positions (pos, channels)) {
-    g_free (pos);
-    return NULL;
-  }
-
-  return pos;
-}
diff --git a/gst-libs/gst/audio/multichannel.h b/gst-libs/gst/audio/multichannel.h
deleted file mode 100644
index 3b322224..00000000
--- a/gst-libs/gst/audio/multichannel.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/* GStreamer Multichannel-Audio helper functions
- * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.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_MULTICHANNEL_H__
-#define __GST_AUDIO_MULTICHANNEL_H__
-
-#include <gst/audio/audio.h>
-#include <gst/audio/multichannel-enumtypes.h>
-
-G_BEGIN_DECLS
-
-typedef enum {
-  GST_AUDIO_CHANNEL_POSITION_INVALID = -1,
-
-  /* Main front speakers. Mono and left/right are mututally exclusive! */
-  GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
-  GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-  GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-
-  /* rear. Left/right and center are mututally exclusive! */
-  GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
-  GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
-
-  /* subwoofer/low-frequency */
-  GST_AUDIO_CHANNEL_POSITION_LFE,
-
-  /* Center front speakers. Center and left/right_of_center cannot be
-   * used together! */
-  GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
-
-  /* sides */
-  GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
-  GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
-
-  /* don't use - counter */
-  GST_AUDIO_CHANNEL_POSITION_NUM
-} GstAudioChannelPosition;
-
-/* Retrieves or sets the positions from/to a GstStructure. Only
- * works with fixed caps, caller should check for that! Caller
- * g_free()s result of the getter. */
-GstAudioChannelPosition *
-	gst_audio_get_channel_positions	(GstStructure *str);
-void	gst_audio_set_channel_positions	(GstStructure *str,
-					 const GstAudioChannelPosition *pos);
-
-/* Sets a (non-fixed) list of possible audio channel positions
- * on a structure (this requires the "channels" property to
- * be fixed!) or on a caps (here, the "channels" property may be
- * unfixed and the caps may even contain multiple structures). */
-void	gst_audio_set_structure_channel_positions_list
-					(GstStructure *str,
-					 const GstAudioChannelPosition *pos,
-					 gint          num_positions);
-void	gst_audio_set_caps_channel_positions_list
-					(GstCaps      *caps,
-					 const GstAudioChannelPosition *pos,
-					 gint          num_positions);
-
-/* Custom fixate function. Elements that implement some sort of
- * channel conversion algorhithm should use this function for
- * fixating on GstAudioChannelPosition properties. It will take
- * care of equal channel positioning (left/right). Caller g_free()s
- * the return value. The input properties may be (and are supposed
- * to be) unfixed. */
-GstAudioChannelPosition *
-	gst_audio_fixate_channel_positions (GstStructure *str);
-
-G_END_DECLS
-
-#endif /* __GST_AUDIO_MULTICHANNEL_H__ */
diff --git a/gst-libs/gst/audio/testchannels.c b/gst-libs/gst/audio/testchannels.c
deleted file mode 100644
index b886c820..00000000
--- a/gst-libs/gst/audio/testchannels.c
+++ /dev/null
@@ -1,55 +0,0 @@
-/* GStreamer Multichannel Test
- * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.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.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <gst/gst.h>
-
-#include <multichannel.c>
-#include <multichannel-enumtypes.c>
-
-gint
-main (gint argc, gchar * argv[])
-{
-  gchar *str;
-  GstCaps *caps;
-  GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-    GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
-  };
-
-  /* register multichannel type */
-  gst_init (&argc, &argv);
-  gst_audio_channel_position_get_type ();
-
-  /* test some caps-string conversions */
-  caps = gst_caps_new_simple ("audio/x-raw-int",
-      "channels", G_TYPE_INT, 2, NULL);
-  str = gst_caps_to_string (caps);
-  g_print ("Test caps #1: %s\n", str);
-  g_free (str);
-  gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
-  str = gst_caps_to_string (caps);
-  g_print ("Test caps #2: %s\n", str);
-  g_free (str);
-  gst_caps_free (caps);
-
-  return 0;
-}