Remove audioparsers plugin, it has been moved to -good

29 files changed, 3 insertions, 6841 deletions

diff --git a/Makefile.am b/Makefile.am
index 4bcaa888c..27200c59b 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -49,6 +49,7 @@ CRUFT_FILES = \
 	$(top_builddir)/ext/jack/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
+	$(top_builddir)/gst/audioparsers/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/imagefreeze/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/selector/.libs/*.{so,dll,DLL,dylib} \
@@ -56,6 +57,7 @@ CRUFT_FILES = \
 	$(top_builddir)/gst/valve/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/videoparsers/.libs/libgsth263parse* \
 	$(top_builddir)/sys/oss4/.libs/*.{so,dll,DLL,dylib} \
+	$(top_builddir)/tests/check/elements/{aac,ac3,amr,flac,mpegaudio,dca}parse \
 	$(top_builddir)/tests/check/elements/autocolorspace \
 	$(top_builddir)/tests/check/elements/capssetter \
 	$(top_builddir)/tests/check/elements/imagefreeze \
diff --git a/android/aacparse.mk b/android/aacparse.mk
deleted file mode 100644
index 67d823382..000000000
--- a/android/aacparse.mk
+++ /dev/null
@@ -1,48 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-
-aacparse_LOCAL_SRC_FILES:= \
-	gst/aacparse/gstaacparse.c \
-	gst/aacparse/gstbaseparse.c 
-
-LOCAL_SRC_FILES:= $(addprefix ../,$(aacparse_LOCAL_SRC_FILES))
-
-LOCAL_SHARED_LIBRARIES := \
-    libgstreamer-0.10       \
-    libgstbase-0.10         \
-    libglib-2.0             \
-    libgthread-2.0          \
-    libgmodule-2.0          \
-    libgobject-2.0   	    \
-    libgstinterfaces-0.10   
-
-LOCAL_MODULE:= libgstaacparse
-
-LOCAL_C_INCLUDES := \
-    $(LOCAL_PATH)/..          			\
-    $(LOCAL_PATH)/../gst-libs  			\
-    $(LOCAL_PATH)       				\
-    $(TARGET_OUT_HEADERS)/gstreamer-0.10 \
-	$(TARGET_OUT_HEADERS)/glib-2.0 		\
-    $(TARGET_OUT_HEADERS)/glib-2.0/glib \
-    external/libxml2/include
-
-ifeq ($(STECONF_ANDROID_VERSION),"FROYO")
-LOCAL_SHARED_LIBRARIES += libicuuc 
-LOCAL_C_INCLUDES += external/icu4c/common
-endif
-
-
-LOCAL_CFLAGS := -DHAVE_CONFIG_H	
-#
-# define LOCAL_PRELINK_MODULE to false to not use pre-link map
-#
-LOCAL_PRELINK_MODULE := false
-
-#It's a gstreamer plugins, and it must be installed on ..../lib/gstreamer-0.10
-LOCAL_MODULE_PATH := $(TARGET_OUT)/lib/gstreamer-0.10
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/android/amrparse.mk b/android/amrparse.mk
deleted file mode 100644
index 183c52da4..000000000
--- a/android/amrparse.mk
+++ /dev/null
@@ -1,48 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-
-amrparse_LOCAL_SRC_FILES:= \
-	gst/amrparse/gstamrparse.c \
-	gst/amrparse/gstbaseparse.c 
-
-LOCAL_SRC_FILES:= $(addprefix ../,$(amrparse_LOCAL_SRC_FILES))
-
-LOCAL_SHARED_LIBRARIES := \
-    libgstreamer-0.10       \
-    libgstbase-0.10         \
-    libglib-2.0             \
-    libgthread-2.0          \
-    libgmodule-2.0          \
-    libgobject-2.0
-
-LOCAL_MODULE:= libgstamrparse
-
-LOCAL_C_INCLUDES := \
-    $(LOCAL_PATH)/../ext/amrwbenc  		\
-    $(LOCAL_PATH)/..           			\
-    $(LOCAL_PATH)/../gst-libs   		\
-	$(LOCAL_PATH)       				\
-    $(TARGET_OUT_HEADERS)/gstreamer-0.10 \
-	$(TARGET_OUT_HEADERS)/glib-2.0 		\
-    $(TARGET_OUT_HEADERS)/glib-2.0/glib \
-    external/libxml2/include
-
-ifeq ($(STECONF_ANDROID_VERSION),"FROYO")
-LOCAL_SHARED_LIBRARIES += libicuuc 
-LOCAL_C_INCLUDES += external/icu4c/common
-endif
-
-LOCAL_CFLAGS := -DHAVE_CONFIG_H	
-#
-# define LOCAL_PRELINK_MODULE to false to not use pre-link map
-#
-LOCAL_PRELINK_MODULE := false
-
-#It's a gstreamer plugins, and it must be installed on ..../lib/gstreamer-0.10
-LOCAL_MODULE_PATH := $(TARGET_OUT)/lib/gstreamer-0.10
-
-include $(BUILD_SHARED_LIBRARY)
-
diff --git a/configure.ac b/configure.ac
index 5fb1c3aba..d454f0867 100644
--- a/configure.ac
+++ b/configure.ac
@@ -291,7 +291,6 @@ AG_GST_CHECK_PLUGIN(adpcmdec)
 AG_GST_CHECK_PLUGIN(adpcmenc)
 AG_GST_CHECK_PLUGIN(aiff)
 AG_GST_CHECK_PLUGIN(asfmux)
-AG_GST_CHECK_PLUGIN(audioparsers)
 AG_GST_CHECK_PLUGIN(autoconvert)
 AG_GST_CHECK_PLUGIN(bayer)
 AG_GST_CHECK_PLUGIN(camerabin)
@@ -1742,7 +1741,6 @@ gst/adpcmdec/Makefile
 gst/adpcmenc/Makefile
 gst/aiff/Makefile
 gst/asfmux/Makefile
-gst/audioparsers/Makefile
 gst/autoconvert/Makefile
 gst/bayer/Makefile
 gst/camerabin/Makefile
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am
index 1efa6ad00..a939357ac 100644
--- a/docs/plugins/Makefile.am
+++ b/docs/plugins/Makefile.am
@@ -136,12 +136,6 @@ EXTRA_HFILES = \
 	$(top_srcdir)/ext/zbar/gstzbar.h \
 	$(top_srcdir)/gst/aiff/aiffparse.h \
 	$(top_srcdir)/gst/aiff/aiffmux.h \
-	$(top_srcdir)/gst/audioparsers/gstaacparse.h \
-	$(top_srcdir)/gst/audioparsers/gstac3parse.h \
-	$(top_srcdir)/gst/audioparsers/gstamrparse.h \
-	$(top_srcdir)/gst/audioparsers/gstflacparse.h \
-	$(top_srcdir)/gst/audioparsers/gstdcaparse.h \
-	$(top_srcdir)/gst/audioparsers/gstmpegaudioparse.h \
 	$(top_srcdir)/gst/autoconvert/gstautoconvert.h \
 	$(top_srcdir)/gst/camerabin/gstcamerabin.h \
 	$(top_srcdir)/gst/coloreffects/gstcoloreffects.h \
diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
index c3e14550c..fd393a723 100644
--- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
@@ -17,11 +17,8 @@
 
   <chapter>
     <title>gst-plugins-bad Elements</title>
-    <xi:include href="xml/element-aacparse.xml" />
-    <xi:include href="xml/element-ac3parse.xml" />
     <xi:include href="xml/element-aiffparse.xml" />
     <xi:include href="xml/element-aiffmux.xml" />
-    <xi:include href="xml/element-amrparse.xml" />
     <xi:include href="xml/element-amrwbenc.xml" />
     <xi:include href="xml/element-assrender.xml" />
     <xi:include href="xml/element-autoconvert.xml" />
@@ -42,7 +39,6 @@
     <xi:include href="xml/element-cvsobel.xml" />
     <xi:include href="xml/element-dataurisrc.xml" />
     <!--xi:include href="xml/element-dc1394.xml" /-->
-    <xi:include href="xml/element-dcaparse.xml" />
     <xi:include href="xml/element-dccpclientsink.xml" />
     <xi:include href="xml/element-dccpclientsrc.xml" />
     <xi:include href="xml/element-dccpserversink.xml" />
@@ -65,7 +61,6 @@
     <xi:include href="xml/element-facedetect.xml" />
     <xi:include href="xml/element-festival.xml" />
     <xi:include href="xml/element-fisheye.xml" />
-    <xi:include href="xml/element-flacparse.xml" />
     <xi:include href="xml/element-fpsdisplaysink.xml" />
     <xi:include href="xml/element-freeze.xml" />
     <xi:include href="xml/element-gaussianblur.xml" />
@@ -84,7 +79,6 @@
     <xi:include href="xml/element-mimdec.xml" />
     <xi:include href="xml/element-mirror.xml" />
     <xi:include href="xml/element-modplug.xml" />
-    <xi:include href="xml/element-mpegaudioparse.xml" />
     <xi:include href="xml/element-mpeg2enc.xml" />
     <xi:include href="xml/element-mplex.xml" />
     <xi:include href="xml/element-mythtvsrc.xml" />
@@ -135,7 +129,6 @@
   <chapter>
     <title>gst-plugins-bad Plugins</title>
     <xi:include href="xml/plugin-aiff.xml" />
-    <xi:include href="xml/plugin-audioparsersbad.xml" />
     <xi:include href="xml/plugin-autoconvert.xml" />
     <xi:include href="xml/plugin-legacyresample.xml" />
     <xi:include href="xml/plugin-amrwbenc.xml" />
diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt
index 4a2af52ea..aa7a81dc6 100644
--- a/docs/plugins/gst-plugins-bad-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt
@@ -1,32 +1,4 @@
 <SECTION>
-<FILE>element-aacparse</FILE>
-<TITLE>aacparse</TITLE>
-GstAacParse
-<SUBSECTION Standard>
-GstAacParseClass
-GST_AACPARSE
-GST_AACPARSE_CLASS
-GST_IS_AACPARSE
-GST_IS_AACPARSE_CLASS
-GST_TYPE_AACPARSE
-gst_aacparse_get_type
-</SECTION>
-
-<SECTION>
-<FILE>element-ac3parse</FILE>
-<TITLE>ac3parse</TITLE>
-GstAc3Parse
-<SUBSECTION Standard>
-GstAc3ParseClass
-GST_AC3_PARSE
-GST_AC3_PARSE_CLASS
-GST_IS_AC3_PARSE
-GST_IS_AC3_PARSE_CLASS
-GST_TYPE_AC3_PARSE
-gst_ac3_parse_get_type
-</SECTION>
-
-<SECTION>
 <FILE>element-aiffmux</FILE>
 <TITLE>aiffmux</TITLE>
 GstAiffMux
@@ -56,20 +28,6 @@ gst_aiff_parse_get_type
 </SECTION>
 
 <SECTION>
-<FILE>element-amrparse</FILE>
-<TITLE>amrparse</TITLE>
-GstAmrParse
-<SUBSECTION Standard>
-GstAmrParseClass
-GST_AMRPARSE
-GST_AMRPARSE_CLASS
-GST_IS_AMRPARSE
-GST_IS_AMRPARSE_CLASS
-GST_TYPE_AMRPARSE
-gst_amrparse_get_type
-</SECTION>
-
-<SECTION>
 <FILE>element-amrwbenc</FILE>
 <TITLE>amrwbenc</TITLE>
 GstAmrwbEnc
@@ -378,20 +336,6 @@ gst_dc1394_get_type
 </SECTION>
 
 <SECTION>
-<FILE>element-dcaparse</FILE>
-<TITLE>dcaparse</TITLE>
-GstDCAParse
-<SUBSECTION Standard>
-GstDCAParseClass
-GST_DCA_PARSE
-GST_DCA_PARSE_CLASS
-GST_IS_DCA_PARSE
-GST_IS_DCA_PARSE_CLASS
-GST_TYPE_DCA_PARSE
-gst_dca_parse_get_type
-</SECTION>
-
-<SECTION>
 <FILE>element-dccpclientsink</FILE>
 <TITLE>dccpclientsink</TITLE>
 GstDCCPClientSink
@@ -752,20 +696,6 @@ gst_fisheye_plugin_init
 </SECTION>
 
 <SECTION>
-<FILE>element-flacparse</FILE>
-<TITLE>flacparse</TITLE>
-GstFlacParse
-<SUBSECTION Standard>
-GstFlacParseClass
-GST_FLAC_PARSE
-GST_FLAC_PARSE_CLASS
-GST_IS_FLAC_PARSE
-GST_IS_FLAC_PARSE_CLASS
-GST_TYPE_FLAC_PARSE
-gst_flac_parse_get_type
-</SECTION>
-
-<SECTION>
 <FILE>element-fpsdisplaysink</FILE>
 <TITLE>fpsdisplaysink</TITLE>
 GstFPSDisplaySink
@@ -1035,20 +965,6 @@ gst_modplug_get_type
 </SECTION>
 
 <SECTION>
-<FILE>element-mpegaudioparse</FILE>
-<TITLE>mpegaudioparse</TITLE>
-GstMpegAudioParse
-<SUBSECTION Standard>
-GstMpegAudioParseClass
-GST_MPEG_AUDIO_PARSE
-GST_MPEG_AUDIO_PARSE_CLASS
-GST_IS_MPEG_AUDIO_PARSE
-GST_IS_MPEG_AUDIO_PARSE_CLASS
-GST_TYPE_MPEG_AUDIO_PARSE
-gst_mpeg_audio_parse_get_type
-</SECTION>
-
-<SECTION>
 <FILE>element-mpeg2enc</FILE>
 <TITLE>mpeg2enc</TITLE>
 GstMpeg2enc
diff --git a/docs/plugins/inspect/plugin-audioparsersbad.xml b/docs/plugins/inspect/plugin-audioparsersbad.xml
deleted file mode 100644
index eb908af26..000000000
--- a/docs/plugins/inspect/plugin-audioparsersbad.xml
+++ /dev/null
@@ -1,139 +0,0 @@
-<plugin>
-  <name>audioparsersbad</name>
-  <description>audioparsers</description>
-  <filename>../../gst/audioparsers/.libs/libgstaudioparsersbad.so</filename>
-  <basename>libgstaudioparsersbad.so</basename>
-  <version>0.10.21.1</version>
-  <license>LGPL</license>
-  <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins git</package>
-  <origin>Unknown package origin</origin>
-  <elements>
-    <element>
-      <name>aacparse</name>
-      <longname>AAC audio stream parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>Advanced Audio Coding parser</description>
-      <author>Stefan Kost &lt;stefan.kost@nokia.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/mpeg, framed=(boolean)false, mpegversion=(int){ 2, 4 }</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/mpeg, framed=(boolean)true, mpegversion=(int){ 2, 4 }, stream-format=(string){ raw, adts, adif }</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>ac3parse</name>
-      <longname>AC3 audio stream parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>AC3 parser</description>
-      <author>Tim-Philipp Müller &lt;tim centricular net&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-ac3, framed=(boolean)false; audio/x-eac3, framed=(boolean)false; audio/ac3, framed=(boolean)false</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-ac3, framed=(boolean)true, channels=(int)[ 1, 6 ], rate=(int)[ 32000, 48000 ]; audio/x-eac3, framed=(boolean)true, channels=(int)[ 1, 6 ], rate=(int)[ 32000, 48000 ]</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>amrparse</name>
-      <longname>AMR audio stream parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>Adaptive Multi-Rate audio parser</description>
-      <author>Ronald Bultje &lt;rbultje@ronald.bitfreak.net&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-amr-nb-sh; audio/x-amr-wb-sh</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/AMR, rate=(int)8000, channels=(int)1; audio/AMR-WB, rate=(int)16000, channels=(int)1</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>dcaparse</name>
-      <longname>DTS Coherent Acoustics audio stream parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>DCA parser</description>
-      <author>Tim-Philipp Müller &lt;tim centricular net&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-dts, framed=(boolean)false</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-dts, framed=(boolean)true, channels=(int)[ 1, 8 ], rate=(int)[ 8000, 192000 ]</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>flacparse</name>
-      <longname>FLAC audio parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>Parses audio with the FLAC lossless audio codec</description>
-      <author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-flac, framed=(boolean)false</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/x-flac, framed=(boolean)true, channels=(int)[ 1, 8 ], rate=(int)[ 1, 655350 ]</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>mpegaudioparse</name>
-      <longname>MPEG1 Audio Parser</longname>
-      <class>Codec/Parser/Audio</class>
-      <description>Parses and frames mpeg1 audio streams (levels 1-3), provides seek</description>
-      <author>Jan Schmidt &lt;thaytan@mad.scientist.com&gt;,Mark Nauwelaerts &lt;mark.nauwelaerts@collabora.co.uk&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/mpeg, mpegversion=(int)1, parsed=(boolean)false</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ], parsed=(boolean)true</details>
-        </caps>
-      </pads>
-    </element>
-  </elements>
-</plugin>
\ No newline at end of file
diff --git a/gst/audioparsers/Makefile.am b/gst/audioparsers/Makefile.am
deleted file mode 100644
index 77039c715..000000000
--- a/gst/audioparsers/Makefile.am
+++ /dev/null
@@ -1,20 +0,0 @@
-plugin_LTLIBRARIES = libgstaudioparsersbad.la
-
-libgstaudioparsersbad_la_SOURCES = \
-	gstaacparse.c gstamrparse.c gstac3parse.c \
-	gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \
-	plugin.c
-
-libgstaudioparsersbad_la_CFLAGS = \
-	-I$(top_srcdir)/gst-libs \
-	$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
-libgstaudioparsersbad_la_LIBADD = \
-	$(top_builddir)/gst-libs/gst/baseparse/libgstbaseparse-$(GST_MAJORMINOR).la \
-	$(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
-	-lgstaudio-$(GST_MAJORMINOR) \
-	$(GST_BASE_LIBS) $(GST_LIBS)
-libgstaudioparsersbad_la_LDFLAGS = $(PACKAGE_LIBS) $(GST_PLUGIN_LDFLAGS)
-libgstaudioparsersbad_la_LIBTOOLFLAGS = --tag=disable-static
-
-noinst_HEADERS = gstaacparse.h gstamrparse.h gstac3parse.h \
-	gstdcaparse.h gstflacparse.h gstmpegaudioparse.h
diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c
deleted file mode 100644
index 09e3e71f2..000000000
--- a/gst/audioparsers/gstaacparse.c
+++ /dev/null
@@ -1,715 +0,0 @@
-/* GStreamer AAC parser plugin
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-aacparse
- * @short_description: AAC parser
- * @see_also: #GstAmrParse
- *
- * This is an AAC parser which handles both ADIF and ADTS stream formats.
- *
- * As ADIF format is not framed, it is not seekable and stream duration cannot
- * be determined either. However, ADTS format AAC clips can be seeked, and parser
- * can also estimate playback position and clip duration.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
- * ]|
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-
-#include "gstaacparse.h"
-
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/mpeg, "
-        "framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, "
-        "stream-format = (string) { raw, adts, adif };"));
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/mpeg, "
-        "framed = (boolean) false, " "mpegversion = (int) { 2, 4 };"));
-
-GST_DEBUG_CATEGORY_STATIC (gst_aacparse_debug);
-#define GST_CAT_DEFAULT gst_aacparse_debug
-
-
-#define ADIF_MAX_SIZE 40        /* Should be enough */
-#define ADTS_MAX_SIZE 10        /* Should be enough */
-
-
-#define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
-
-gboolean gst_aacparse_start (GstBaseParse * parse);
-gboolean gst_aacparse_stop (GstBaseParse * parse);
-
-static gboolean gst_aacparse_sink_setcaps (GstBaseParse * parse,
-    GstCaps * caps);
-
-gboolean gst_aacparse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * size, gint * skipsize);
-
-GstFlowReturn gst_aacparse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-
-gboolean gst_aacparse_convert (GstBaseParse * parse,
-    GstFormat src_format,
-    gint64 src_value, GstFormat dest_format, gint64 * dest_value);
-
-gint gst_aacparse_get_frame_overhead (GstBaseParse * parse, GstBuffer * buffer);
-
-gboolean gst_aacparse_event (GstBaseParse * parse, GstEvent * event);
-
-#define _do_init(bla) \
-    GST_DEBUG_CATEGORY_INIT (gst_aacparse_debug, "aacparse", 0, \
-    "AAC audio stream parser");
-
-GST_BOILERPLATE_FULL (GstAacParse, gst_aacparse, GstBaseParse,
-    GST_TYPE_BASE_PARSE, _do_init);
-
-static inline gint
-gst_aacparse_get_sample_rate_from_index (guint sr_idx)
-{
-  static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
-    32000, 24000, 22050, 16000, 12000, 11025, 8000
-  };
-
-  if (sr_idx < G_N_ELEMENTS (aac_sample_rates))
-    return aac_sample_rates[sr_idx];
-  GST_WARNING ("Invalid sample rate index %u", sr_idx);
-  return 0;
-}
-
-/**
- * gst_aacparse_base_init:
- * @klass: #GstElementClass.
- *
- */
-static void
-gst_aacparse_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_template));
-
-  gst_element_class_set_details_simple (element_class,
-      "AAC audio stream parser", "Codec/Parser/Audio",
-      "Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
-}
-
-
-/**
- * gst_aacparse_class_init:
- * @klass: #GstAacParseClass.
- *
- */
-static void
-gst_aacparse_class_init (GstAacParseClass * klass)
-{
-  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
-
-  parse_class->start = GST_DEBUG_FUNCPTR (gst_aacparse_start);
-  parse_class->stop = GST_DEBUG_FUNCPTR (gst_aacparse_stop);
-  parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aacparse_sink_setcaps);
-  parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_aacparse_parse_frame);
-  parse_class->check_valid_frame =
-      GST_DEBUG_FUNCPTR (gst_aacparse_check_valid_frame);
-}
-
-
-/**
- * gst_aacparse_init:
- * @aacparse: #GstAacParse.
- * @klass: #GstAacParseClass.
- *
- */
-static void
-gst_aacparse_init (GstAacParse * aacparse, GstAacParseClass * klass)
-{
-  GST_DEBUG ("initialized");
-}
-
-
-/**
- * gst_aacparse_set_src_caps:
- * @aacparse: #GstAacParse.
- * @sink_caps: (proposed) caps of sink pad
- *
- * Set source pad caps according to current knowledge about the
- * audio stream.
- *
- * Returns: TRUE if caps were successfully set.
- */
-static gboolean
-gst_aacparse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
-{
-  GstStructure *s;
-  GstCaps *src_caps = NULL;
-  gboolean res = FALSE;
-  const gchar *stream_format;
-
-  GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
-  if (sink_caps)
-    src_caps = gst_caps_copy (sink_caps);
-  else
-    src_caps = gst_caps_new_simple ("audio/mpeg", NULL);
-
-  gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
-      "mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);
-
-  switch (aacparse->header_type) {
-    case DSPAAC_HEADER_NONE:
-      stream_format = "raw";
-      break;
-    case DSPAAC_HEADER_ADTS:
-      stream_format = "adts";
-      break;
-    case DSPAAC_HEADER_ADIF:
-      stream_format = "adif";
-      break;
-    default:
-      stream_format = NULL;
-  }
-
-  s = gst_caps_get_structure (src_caps, 0);
-  if (aacparse->sample_rate > 0)
-    gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
-  if (aacparse->channels > 0)
-    gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
-  if (stream_format)
-    gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
-
-  GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
-
-  res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
-  gst_caps_unref (src_caps);
-  return res;
-}
-
-
-/**
- * gst_aacparse_sink_setcaps:
- * @sinkpad: GstPad
- * @caps: GstCaps
- *
- * Implementation of "set_sink_caps" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE on success.
- */
-static gboolean
-gst_aacparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
-{
-  GstAacParse *aacparse;
-  GstStructure *structure;
-  gchar *caps_str;
-  const GValue *value;
-
-  aacparse = GST_AACPARSE (parse);
-  structure = gst_caps_get_structure (caps, 0);
-  caps_str = gst_caps_to_string (caps);
-
-  GST_DEBUG_OBJECT (aacparse, "setcaps: %s", caps_str);
-  g_free (caps_str);
-
-  /* This is needed at least in case of RTP
-   * Parses the codec_data information to get ObjectType,
-   * number of channels and samplerate */
-  value = gst_structure_get_value (structure, "codec_data");
-  if (value) {
-    GstBuffer *buf = gst_value_get_buffer (value);
-
-    if (buf) {
-      const guint8 *buffer = GST_BUFFER_DATA (buf);
-      guint sr_idx;
-
-      sr_idx = ((buffer[0] & 0x07) << 1) | ((buffer[1] & 0x80) >> 7);
-      aacparse->object_type = (buffer[0] & 0xf8) >> 3;
-      aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
-      aacparse->channels = (buffer[1] & 0x78) >> 3;
-      aacparse->header_type = DSPAAC_HEADER_NONE;
-      aacparse->mpegversion = 4;
-
-      GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d",
-          aacparse->object_type, aacparse->sample_rate, aacparse->channels);
-
-      /* arrange for metadata and get out of the way */
-      gst_aacparse_set_src_caps (aacparse, caps);
-      gst_base_parse_set_format (parse,
-          GST_BASE_PARSE_FORMAT_PASSTHROUGH, TRUE);
-    } else
-      return FALSE;
-
-    /* caps info overrides */
-    gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
-    gst_structure_get_int (structure, "channels", &aacparse->channels);
-  }
-
-  return TRUE;
-}
-
-
-/**
- * gst_aacparse_adts_get_frame_len:
- * @data: block of data containing an ADTS header.
- *
- * This function calculates ADTS frame length from the given header.
- *
- * Returns: size of the ADTS frame.
- */
-static inline guint
-gst_aacparse_adts_get_frame_len (const guint8 * data)
-{
-  return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
-}
-
-
-/**
- * gst_aacparse_check_adts_frame:
- * @aacparse: #GstAacParse.
- * @data: Data to be checked.
- * @avail: Amount of data passed.
- * @framesize: If valid ADTS frame was found, this will be set to tell the
- *             found frame size in bytes.
- * @needed_data: If frame was not found, this may be set to tell how much
- *               more data is needed in the next round to detect the frame
- *               reliably. This may happen when a frame header candidate
- *               is found but it cannot be guaranteed to be the header without
- *               peeking the following data.
- *
- * Check if the given data contains contains ADTS frame. The algorithm
- * will examine ADTS frame header and calculate the frame size. Also, another
- * consecutive ADTS frame header need to be present after the found frame.
- * Otherwise the data is not considered as a valid ADTS frame. However, this
- * "extra check" is omitted when EOS has been received. In this case it is
- * enough when data[0] contains a valid ADTS header.
- *
- * This function may set the #needed_data to indicate that a possible frame
- * candidate has been found, but more data (#needed_data bytes) is needed to
- * be absolutely sure. When this situation occurs, FALSE will be returned.
- *
- * When a valid frame is detected, this function will use
- * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
- * to set the needed bytes for next frame.This way next data chunk is already
- * of correct size.
- *
- * Returns: TRUE if the given data contains a valid ADTS header.
- */
-static gboolean
-gst_aacparse_check_adts_frame (GstAacParse * aacparse,
-    const guint8 * data, const guint avail, gboolean drain,
-    guint * framesize, guint * needed_data)
-{
-  if (G_UNLIKELY (avail < 2))
-    return FALSE;
-
-  if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
-    *framesize = gst_aacparse_adts_get_frame_len (data);
-
-    /* In EOS mode this is enough. No need to examine the data further */
-    if (drain) {
-      return TRUE;
-    }
-
-    if (*framesize + ADTS_MAX_SIZE > avail) {
-      /* We have found a possible frame header candidate, but can't be
-         sure since we don't have enough data to check the next frame */
-      GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
-          *framesize + ADTS_MAX_SIZE, avail);
-      *needed_data = *framesize + ADTS_MAX_SIZE;
-      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
-          *framesize + ADTS_MAX_SIZE);
-      return FALSE;
-    }
-
-    if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
-      guint nextlen = gst_aacparse_adts_get_frame_len (data + (*framesize));
-
-      GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
-      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
-          nextlen + ADTS_MAX_SIZE);
-      return TRUE;
-    }
-  }
-  return FALSE;
-}
-
-/* caller ensure sufficient data */
-static inline void
-gst_aacparse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
-    gint * rate, gint * channels, gint * object, gint * version)
-{
-
-  if (rate) {
-    gint sr_idx = (data[2] & 0x3c) >> 2;
-
-    *rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
-  }
-  if (channels)
-    *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
-
-  if (version)
-    *version = (data[1] & 0x08) ? 2 : 4;
-  if (object)
-    *object = (data[2] & 0xc0) >> 6;
-}
-
-/**
- * gst_aacparse_detect_stream:
- * @aacparse: #GstAacParse.
- * @data: A block of data that needs to be examined for stream characteristics.
- * @avail: Size of the given datablock.
- * @framesize: If valid stream was found, this will be set to tell the
- *             first frame size in bytes.
- * @skipsize: If valid stream was found, this will be set to tell the first
- *            audio frame position within the given data.
- *
- * Examines the given piece of data and try to detect the format of it. It
- * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
- * header. If the stream is detected, TRUE will be returned and #framesize
- * is set to indicate the found frame size. Additionally, #skipsize might
- * be set to indicate the number of bytes that need to be skipped, a.k.a. the
- * position of the frame inside given data chunk.
- *
- * Returns: TRUE on success.
- */
-static gboolean
-gst_aacparse_detect_stream (GstAacParse * aacparse,
-    const guint8 * data, const guint avail, gboolean drain,
-    guint * framesize, gint * skipsize)
-{
-  gboolean found = FALSE;
-  guint need_data = 0;
-  guint i = 0;
-
-  GST_DEBUG_OBJECT (aacparse, "Parsing header data");
-
-  /* FIXME: No need to check for ADIF if we are not in the beginning of the
-     stream */
-
-  /* Can we even parse the header? */
-  if (avail < ADTS_MAX_SIZE)
-    return FALSE;
-
-  for (i = 0; i < avail - 4; i++) {
-    if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
-        strncmp ((char *) data + i, "ADIF", 4) == 0) {
-      found = TRUE;
-
-      if (i) {
-        /* Trick: tell the parent class that we didn't find the frame yet,
-           but make it skip 'i' amount of bytes. Next time we arrive
-           here we have full frame in the beginning of the data. */
-        *skipsize = i;
-        return FALSE;
-      }
-      break;
-    }
-  }
-  if (!found) {
-    if (i)
-      *skipsize = i;
-    return FALSE;
-  }
-
-  if (gst_aacparse_check_adts_frame (aacparse, data, avail, drain,
-          framesize, &need_data)) {
-    gint rate, channels;
-
-    GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
-
-    aacparse->header_type = DSPAAC_HEADER_ADTS;
-    gst_aacparse_parse_adts_header (aacparse, data, &rate, &channels,
-        &aacparse->object_type, &aacparse->mpegversion);
-
-    gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse),
-        rate, 1024, 2, 2);
-
-    GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
-        rate, channels, aacparse->object_type, aacparse->mpegversion);
-
-    return TRUE;
-  } else if (need_data) {
-    /* This tells the parent class not to skip any data */
-    *skipsize = 0;
-    return FALSE;
-  }
-
-  if (avail < ADIF_MAX_SIZE)
-    return FALSE;
-
-  if (memcmp (data + i, "ADIF", 4) == 0) {
-    const guint8 *adif;
-    int skip_size = 0;
-    int bitstream_type;
-    int sr_idx;
-
-    aacparse->header_type = DSPAAC_HEADER_ADIF;
-    aacparse->mpegversion = 4;
-
-    /* no way to seek this */
-    gst_base_parse_set_seek (GST_BASE_PARSE (aacparse),
-        GST_BASE_PARSE_SEEK_NONE, 0);
-
-    /* Skip the "ADIF" bytes */
-    adif = data + i + 4;
-
-    /* copyright string */
-    if (adif[0] & 0x80)
-      skip_size += 9;           /* skip 9 bytes */
-
-    bitstream_type = adif[0 + skip_size] & 0x10;
-    aacparse->bitrate =
-        ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
-        ((unsigned int) adif[1 + skip_size] << 11) |
-        ((unsigned int) adif[2 + skip_size] << 3) |
-        ((unsigned int) adif[3 + skip_size] & 0xe0);
-
-    /* CBR */
-    if (bitstream_type == 0) {
-#if 0
-      /* Buffer fullness parsing. Currently not needed... */
-      guint num_elems = 0;
-      guint fullness = 0;
-
-      num_elems = (adif[3 + skip_size] & 0x1e);
-      GST_INFO ("ADIF num_config_elems: %d", num_elems);
-
-      fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
-          ((unsigned int) adif[4 + skip_size] << 11) |
-          ((unsigned int) adif[5 + skip_size] << 3) |
-          ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
-
-      GST_INFO ("ADIF buffer fullness: %d", fullness);
-#endif
-      aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
-          ((adif[7 + skip_size] & 0x80) >> 7);
-      sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
-    }
-    /* VBR */
-    else {
-      aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
-      sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
-          ((adif[5 + skip_size] & 0x80) >> 7);
-    }
-
-    /* FIXME: This gives totally wrong results. Duration calculation cannot
-       be based on this */
-    aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
-
-    /* baseparse is not given any fps,
-     * so it will give up on timestamps, seeking, etc */
-
-    /* FIXME: Can we assume this? */
-    aacparse->channels = 2;
-
-    GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
-        aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
-
-    gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
-
-    /* arrange for metadata and get out of the way */
-    gst_aacparse_set_src_caps (aacparse,
-        GST_PAD_CAPS (GST_BASE_PARSE_SINK_PAD (aacparse)));
-    gst_base_parse_set_format (GST_BASE_PARSE (aacparse),
-        GST_BASE_PARSE_FORMAT_PASSTHROUGH, TRUE);
-
-    *framesize = avail;
-    return TRUE;
-  }
-
-  /* This should never happen */
-  return FALSE;
-}
-
-
-/**
- * gst_aacparse_check_valid_frame:
- * @parse: #GstBaseParse.
- * @buffer: #GstBuffer.
- * @framesize: If the buffer contains a valid frame, its size will be put here
- * @skipsize: How much data parent class should skip in order to find the
- *            frame header.
- *
- * Implementation of "check_valid_frame" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE if buffer contains a valid frame.
- */
-gboolean
-gst_aacparse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
-{
-  const guint8 *data;
-  GstAacParse *aacparse;
-  gboolean ret = FALSE;
-  gboolean sync;
-  GstBuffer *buffer;
-
-  aacparse = GST_AACPARSE (parse);
-  buffer = frame->buffer;
-  data = GST_BUFFER_DATA (buffer);
-
-  sync = GST_BASE_PARSE_FRAME_SYNC (frame);
-
-  if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
-      aacparse->header_type == DSPAAC_HEADER_NONE) {
-    /* There is nothing to parse */
-    *framesize = GST_BUFFER_SIZE (buffer);
-    ret = TRUE;
-
-  } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || sync == FALSE) {
-
-    ret = gst_aacparse_detect_stream (aacparse, data, GST_BUFFER_SIZE (buffer),
-        GST_BASE_PARSE_FRAME_DRAIN (frame), framesize, skipsize);
-
-  } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
-    guint needed_data = 1024;
-
-    ret = gst_aacparse_check_adts_frame (aacparse, data,
-        GST_BUFFER_SIZE (buffer), GST_BASE_PARSE_FRAME_DRAIN (frame),
-        framesize, &needed_data);
-
-    if (!ret) {
-      GST_DEBUG ("buffer didn't contain valid frame");
-      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
-          needed_data);
-    }
-
-  } else {
-    GST_DEBUG ("buffer didn't contain valid frame");
-    gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
-  }
-
-  return ret;
-}
-
-
-/**
- * gst_aacparse_parse_frame:
- * @parse: #GstBaseParse.
- * @buffer: #GstBuffer.
- *
- * Implementation of "parse_frame" vmethod in #GstBaseParse class.
- *
- * Also determines frame overhead.
- * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
- * a per-frame header.
- *
- * We're making a couple of simplifying assumptions:
- *
- * 1. We count Program Configuration Elements rather than searching for them
- *    in the streams to discount them - the overhead is negligible.
- *
- * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
- *    bits, which should still not be significant enough to warrant the
- *    additional parsing through the headers
- *
- * Returns: GST_FLOW_OK if frame was successfully parsed and can be pushed
- *          forward. Otherwise appropriate error is returned.
- */
-GstFlowReturn
-gst_aacparse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
-  GstAacParse *aacparse;
-  GstBuffer *buffer;
-  GstFlowReturn ret = GST_FLOW_OK;
-  gint rate, channels;
-
-  aacparse = GST_AACPARSE (parse);
-  buffer = frame->buffer;
-
-  if (G_UNLIKELY (aacparse->header_type != DSPAAC_HEADER_ADTS))
-    return ret;
-
-  /* see above */
-  frame->overhead = 7;
-
-  gst_aacparse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer),
-      &rate, &channels, NULL, NULL);
-  GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
-
-  if (G_UNLIKELY (rate != aacparse->sample_rate
-          || channels != aacparse->channels)) {
-    aacparse->sample_rate = rate;
-    aacparse->channels = channels;
-
-    if (!gst_aacparse_set_src_caps (aacparse,
-            GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) {
-      /* If linking fails, we need to return appropriate error */
-      ret = GST_FLOW_NOT_LINKED;
-    }
-
-    gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse),
-        aacparse->sample_rate, 1024, 2, 2);
-  }
-
-  return ret;
-}
-
-
-/**
- * gst_aacparse_start:
- * @parse: #GstBaseParse.
- *
- * Implementation of "start" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE if startup succeeded.
- */
-gboolean
-gst_aacparse_start (GstBaseParse * parse)
-{
-  GstAacParse *aacparse;
-
-  aacparse = GST_AACPARSE (parse);
-  GST_DEBUG ("start");
-  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
-  return TRUE;
-}
-
-
-/**
- * gst_aacparse_stop:
- * @parse: #GstBaseParse.
- *
- * Implementation of "stop" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE is stopping succeeded.
- */
-gboolean
-gst_aacparse_stop (GstBaseParse * parse)
-{
-  GST_DEBUG ("stop");
-  return TRUE;
-}
diff --git a/gst/audioparsers/gstaacparse.h b/gst/audioparsers/gstaacparse.h
deleted file mode 100644
index e62bf651f..000000000
--- a/gst/audioparsers/gstaacparse.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/* GStreamer AAC parser
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * 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_AACPARSE_H__
-#define __GST_AACPARSE_H__
-
-#include <gst/gst.h>
-#include <gst/baseparse/gstbaseparse.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AACPARSE \
-  (gst_aacparse_get_type())
-#define GST_AACPARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AACPARSE, GstAacParse))
-#define GST_AACPARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AACPARSE, GstAacParseClass))
-#define GST_IS_AACPARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AACPARSE))
-#define GST_IS_AACPARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AACPARSE))
-
-
-/**
- * GstAacHeaderType:
- * @DSPAAC_HEADER_NOT_PARSED: Header not parsed yet.
- * @DSPAAC_HEADER_UNKNOWN: Unknown (not recognized) header.
- * @DSPAAC_HEADER_ADIF: ADIF header found.
- * @DSPAAC_HEADER_ADTS: ADTS header found.
- * @DSPAAC_HEADER_NONE: Raw stream, no header.
- *
- * Type header enumeration set in #header_type.
- */
-typedef enum {
-  DSPAAC_HEADER_NOT_PARSED,
-  DSPAAC_HEADER_UNKNOWN,
-  DSPAAC_HEADER_ADIF,
-  DSPAAC_HEADER_ADTS,
-  DSPAAC_HEADER_NONE
-} GstAacHeaderType;
-
-
-typedef struct _GstAacParse GstAacParse;
-typedef struct _GstAacParseClass GstAacParseClass;
-
-/**
- * GstAacParse:
- * @element: the parent element.
- * @object_type: AAC object type of the stream.
- * @bitrate:  Current media bitrate.
- * @sample_rate: Current media samplerate.
- * @channels: Current media channel count.
- * @frames_per_sec: FPS value of the current stream.
- * @header_type: #GstAacHeaderType indicating the current stream type.
- * @framecount: The amount of frames that has been processed this far.
- * @bytecount: The amount of bytes that has been processed this far.
- * @sync: Tells whether the parser is in sync (a.k.a. not searching for header)
- * @eos: End-of-Stream indicator. Set when EOS event arrives.
- * @duration: Duration of the current stream.
- * @ts: Current stream timestamp.
- *
- * The opaque GstAacParse data structure.
- */
-struct _GstAacParse {
-  GstBaseParse element;
-
-  /* Stream type -related info */
-  gint           object_type;
-  gint           bitrate;
-  gint           sample_rate;
-  gint           channels;
-  gint           mpegversion;
-
-  GstAacHeaderType header_type;
-};
-
-/**
- * GstAacParseClass:
- * @parent_class: Element parent class.
- *
- * The opaque GstAacParseClass data structure.
- */
-struct _GstAacParseClass {
-  GstBaseParseClass parent_class;
-};
-
-GType gst_aacparse_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_AACPARSE_H__ */
diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c
deleted file mode 100644
index e001bc37e..000000000
--- a/gst/audioparsers/gstac3parse.c
+++ /dev/null
@@ -1,507 +0,0 @@
-/* GStreamer AC3 parser
- * Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
- * Copyright (C) 2009 Mark Nauwelaerts <mnauw users sf net>
- * Copyright (C) 2009 Nokia Corporation. All rights reserved.
- *   Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-/**
- * SECTION:element-ac3parse
- * @short_description: AC3 parser
- * @see_also: #GstAmrParse, #GstAACParse
- *
- * This is an AC3 parser.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioresample ! audioconvert ! autoaudiosink
- * ]|
- * </refsect2>
- */
-
-/* TODO:
- *  - add support for audio/x-private1-ac3 as well
- *  - should accept framed and unframed input (needs decodebin fixes first)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-
-#include "gstac3parse.h"
-#include <gst/base/gstbytereader.h>
-#include <gst/base/gstbitreader.h>
-
-GST_DEBUG_CATEGORY_STATIC (ac3_parse_debug);
-#define GST_CAT_DEFAULT ac3_parse_debug
-
-static const struct
-{
-  const guint bit_rate;         /* nominal bit rate */
-  const guint frame_size[3];    /* frame size for 32kHz, 44kHz, and 48kHz */
-} frmsizcod_table[38] = {
-  {
-    32, {
-  64, 69, 96}}, {
-    32, {
-  64, 70, 96}}, {
-    40, {
-  80, 87, 120}}, {
-    40, {
-  80, 88, 120}}, {
-    48, {
-  96, 104, 144}}, {
-    48, {
-  96, 105, 144}}, {
-    56, {
-  112, 121, 168}}, {
-    56, {
-  112, 122, 168}}, {
-    64, {
-  128, 139, 192}}, {
-    64, {
-  128, 140, 192}}, {
-    80, {
-  160, 174, 240}}, {
-    80, {
-  160, 175, 240}}, {
-    96, {
-  192, 208, 288}}, {
-    96, {
-  192, 209, 288}}, {
-    112, {
-  224, 243, 336}}, {
-    112, {
-  224, 244, 336}}, {
-    128, {
-  256, 278, 384}}, {
-    128, {
-  256, 279, 384}}, {
-    160, {
-  320, 348, 480}}, {
-    160, {
-  320, 349, 480}}, {
-    192, {
-  384, 417, 576}}, {
-    192, {
-  384, 418, 576}}, {
-    224, {
-  448, 487, 672}}, {
-    224, {
-  448, 488, 672}}, {
-    256, {
-  512, 557, 768}}, {
-    256, {
-  512, 558, 768}}, {
-    320, {
-  640, 696, 960}}, {
-    320, {
-  640, 697, 960}}, {
-    384, {
-  768, 835, 1152}}, {
-    384, {
-  768, 836, 1152}}, {
-    448, {
-  896, 975, 1344}}, {
-    448, {
-  896, 976, 1344}}, {
-    512, {
-  1024, 1114, 1536}}, {
-    512, {
-  1024, 1115, 1536}}, {
-    576, {
-  1152, 1253, 1728}}, {
-    576, {
-  1152, 1254, 1728}}, {
-    640, {
-  1280, 1393, 1920}}, {
-    640, {
-  1280, 1394, 1920}}
-};
-
-static const guint fscod_rates[4] = { 48000, 44100, 32000, 0 };
-static const guint acmod_chans[8] = { 2, 1, 2, 3, 3, 4, 4, 5 };
-static const guint numblks[4] = { 1, 2, 3, 6 };
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-ac3, framed = (boolean) true, "
-        " channels = (int) [ 1, 6 ], rate = (int) [ 32000, 48000 ]; "
-        "audio/x-eac3, framed = (boolean) true, "
-        " channels = (int) [ 1, 6 ], rate = (int) [ 32000, 48000 ] "));
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-ac3, framed = (boolean) false; "
-        "audio/x-eac3, framed = (boolean) false; "
-        "audio/ac3, framed = (boolean) false "));
-
-static void gst_ac3_parse_finalize (GObject * object);
-
-static gboolean gst_ac3_parse_start (GstBaseParse * parse);
-static gboolean gst_ac3_parse_stop (GstBaseParse * parse);
-static gboolean gst_ac3_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * size, gint * skipsize);
-static GstFlowReturn gst_ac3_parse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-
-GST_BOILERPLATE (GstAc3Parse, gst_ac3_parse, GstBaseParse, GST_TYPE_BASE_PARSE);
-
-static void
-gst_ac3_parse_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_template));
-
-  gst_element_class_set_details_simple (element_class,
-      "AC3 audio stream parser", "Codec/Parser/Audio",
-      "AC3 parser", "Tim-Philipp Müller <tim centricular net>");
-}
-
-static void
-gst_ac3_parse_class_init (GstAc3ParseClass * klass)
-{
-  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
-  GObjectClass *object_class = G_OBJECT_CLASS (klass);
-
-  GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0,
-      "AC3 audio stream parser");
-
-  object_class->finalize = gst_ac3_parse_finalize;
-
-  parse_class->start = GST_DEBUG_FUNCPTR (gst_ac3_parse_start);
-  parse_class->stop = GST_DEBUG_FUNCPTR (gst_ac3_parse_stop);
-  parse_class->check_valid_frame =
-      GST_DEBUG_FUNCPTR (gst_ac3_parse_check_valid_frame);
-  parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_ac3_parse_parse_frame);
-}
-
-static void
-gst_ac3_parse_reset (GstAc3Parse * ac3parse)
-{
-  ac3parse->channels = -1;
-  ac3parse->sample_rate = -1;
-  ac3parse->eac = FALSE;
-}
-
-static void
-gst_ac3_parse_init (GstAc3Parse * ac3parse, GstAc3ParseClass * klass)
-{
-  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (ac3parse), 64 * 2);
-  gst_ac3_parse_reset (ac3parse);
-}
-
-static void
-gst_ac3_parse_finalize (GObject * object)
-{
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_ac3_parse_start (GstBaseParse * parse)
-{
-  GstAc3Parse *ac3parse = GST_AC3_PARSE (parse);
-
-  GST_DEBUG_OBJECT (parse, "starting");
-
-  gst_ac3_parse_reset (ac3parse);
-
-  return TRUE;
-}
-
-static gboolean
-gst_ac3_parse_stop (GstBaseParse * parse)
-{
-  GST_DEBUG_OBJECT (parse, "stopping");
-
-  return TRUE;
-}
-
-static gboolean
-gst_ac3_parse_frame_header_ac3 (GstAc3Parse * ac3parse, GstBuffer * buf,
-    guint * frame_size, guint * rate, guint * chans, guint * blks, guint * sid)
-{
-  GstBitReader bits = GST_BIT_READER_INIT_FROM_BUFFER (buf);
-  guint8 fscod, frmsizcod, bsid, bsmod, acmod, lfe_on;
-
-  GST_LOG_OBJECT (ac3parse, "parsing ac3");
-
-  gst_bit_reader_skip_unchecked (&bits, 16 + 16);
-  fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2);
-  frmsizcod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 6);
-
-  if (G_UNLIKELY (fscod == 3 || frmsizcod >= G_N_ELEMENTS (frmsizcod_table))) {
-    GST_DEBUG_OBJECT (ac3parse, "bad fscod=%d frmsizcod=%d", fscod, frmsizcod);
-    return FALSE;
-  }
-
-  bsid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 5);
-  bsmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3);
-  acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3);
-
-  /* spec not quite clear here: decoder should decode if less than 8,
-   * but seemingly only defines 6 and 8 cases */
-  if (bsid > 8) {
-    GST_DEBUG_OBJECT (ac3parse, "unexpected bsid=%d", bsid);
-    return FALSE;
-  } else if (bsid != 8 && bsid != 6) {
-    GST_DEBUG_OBJECT (ac3parse, "undefined bsid=%d", bsid);
-  }
-
-  if ((acmod & 0x1) && (acmod != 0x1))  /* 3 front channels */
-    gst_bit_reader_skip_unchecked (&bits, 2);
-  if ((acmod & 0x4))            /* if a surround channel exists */
-    gst_bit_reader_skip_unchecked (&bits, 2);
-  if (acmod == 0x2)             /* if in 2/0 mode */
-    gst_bit_reader_skip_unchecked (&bits, 2);
-
-  lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1);
-
-  if (frame_size)
-    *frame_size = frmsizcod_table[frmsizcod].frame_size[fscod] * 2;
-  if (rate)
-    *rate = fscod_rates[fscod];
-  if (chans)
-    *chans = acmod_chans[acmod] + lfe_on;
-  if (blks)
-    *blks = 6;
-  if (sid)
-    *sid = 0;
-
-  return TRUE;
-}
-
-static gboolean
-gst_ac3_parse_frame_header_eac3 (GstAc3Parse * ac3parse, GstBuffer * buf,
-    guint * frame_size, guint * rate, guint * chans, guint * blks, guint * sid)
-{
-  GstBitReader bits = GST_BIT_READER_INIT_FROM_BUFFER (buf);
-  guint16 frmsiz, sample_rate, blocks;
-  guint8 strmtyp, fscod, fscod2, acmod, lfe_on, strmid, numblkscod;
-
-  GST_LOG_OBJECT (ac3parse, "parsing e-ac3");
-
-  gst_bit_reader_skip_unchecked (&bits, 16);
-  strmtyp = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* strmtyp     */
-  if (G_UNLIKELY (strmtyp == 3)) {
-    GST_DEBUG_OBJECT (ac3parse, "bad strmtyp %d", strmtyp);
-    return FALSE;
-  }
-
-  strmid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3);  /* substreamid */
-  frmsiz = gst_bit_reader_get_bits_uint16_unchecked (&bits, 11);        /* frmsiz      */
-  fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2);   /* fscod       */
-  if (fscod == 3) {
-    fscod2 = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2);        /* fscod2      */
-    if (G_UNLIKELY (fscod2 == 3)) {
-      GST_DEBUG_OBJECT (ac3parse, "invalid fscod2");
-      return FALSE;
-    }
-    sample_rate = fscod_rates[fscod2] / 2;
-    blocks = 6;
-  } else {
-    numblkscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2);    /* numblkscod  */
-    sample_rate = fscod_rates[fscod];
-    blocks = numblks[numblkscod];
-  }
-
-  acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3);   /* acmod       */
-  lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1);  /* lfeon       */
-
-  gst_bit_reader_skip_unchecked (&bits, 5);     /* bsid        */
-
-  if (frame_size)
-    *frame_size = (frmsiz + 1) * 2;
-  if (rate)
-    *rate = sample_rate;
-  if (chans)
-    *chans = acmod_chans[acmod] + lfe_on;
-  if (blks)
-    *blks = blocks;
-  if (sid)
-    *sid = (strmtyp & 0x1) << 3 | strmid;
-
-  return TRUE;
-}
-
-static gboolean
-gst_ac3_parse_frame_header (GstAc3Parse * parse, GstBuffer * buf,
-    guint * framesize, guint * rate, guint * chans, guint * blocks,
-    guint * sid, gboolean * eac)
-{
-  GstBitReader bits = GST_BIT_READER_INIT_FROM_BUFFER (buf);
-  guint16 sync;
-  guint8 bsid;
-
-  GST_MEMDUMP_OBJECT (parse, "AC3 frame sync", GST_BUFFER_DATA (buf), 16);
-
-  sync = gst_bit_reader_get_bits_uint16_unchecked (&bits, 16);
-  gst_bit_reader_skip_unchecked (&bits, 16 + 8);
-  bsid = gst_bit_reader_peek_bits_uint8_unchecked (&bits, 5);
-
-  if (G_UNLIKELY (sync != 0x0b77))
-    return FALSE;
-
-  GST_LOG_OBJECT (parse, "bsid = %d", bsid);
-
-  if (bsid <= 10) {
-    if (eac)
-      *eac = FALSE;
-    return gst_ac3_parse_frame_header_ac3 (parse, buf, framesize, rate, chans,
-        blocks, sid);
-  } else if (bsid <= 16) {
-    if (eac)
-      *eac = TRUE;
-    return gst_ac3_parse_frame_header_eac3 (parse, buf, framesize, rate, chans,
-        blocks, sid);
-  } else {
-    GST_DEBUG_OBJECT (parse, "unexpected bsid %d", bsid);
-    return FALSE;
-  }
-}
-
-static gboolean
-gst_ac3_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
-{
-  GstAc3Parse *ac3parse = GST_AC3_PARSE (parse);
-  GstBuffer *buf = frame->buffer;
-  GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
-  gint off;
-  gboolean sync, drain;
-
-  if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6))
-    return FALSE;
-
-  off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000, 0x0b770000,
-      0, GST_BUFFER_SIZE (buf));
-
-  GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
-
-  /* didn't find anything that looks like a sync word, skip */
-  if (off < 0) {
-    *skipsize = GST_BUFFER_SIZE (buf) - 3;
-    return FALSE;
-  }
-
-  /* possible frame header, but not at offset 0? skip bytes before sync */
-  if (off > 0) {
-    *skipsize = off;
-    return FALSE;
-  }
-
-  /* make sure the values in the frame header look sane */
-  if (!gst_ac3_parse_frame_header (ac3parse, buf, framesize, NULL, NULL,
-          NULL, NULL, NULL)) {
-    *skipsize = off + 2;
-    return FALSE;
-  }
-
-  GST_LOG_OBJECT (parse, "got frame");
-
-  sync = GST_BASE_PARSE_FRAME_SYNC (frame);
-  drain = GST_BASE_PARSE_FRAME_DRAIN (frame);
-
-  if (!sync && !drain) {
-    guint16 word = 0;
-
-    GST_DEBUG_OBJECT (ac3parse, "resyncing; checking next frame syncword");
-
-    if (!gst_byte_reader_skip (&reader, *framesize) ||
-        !gst_byte_reader_get_uint16_be (&reader, &word)) {
-      GST_DEBUG_OBJECT (ac3parse, "... but not sufficient data");
-      gst_base_parse_set_min_frame_size (parse, *framesize + 6);
-      *skipsize = 0;
-      return FALSE;
-    } else {
-      if (word != 0x0b77) {
-        GST_DEBUG_OBJECT (ac3parse, "0x%x not OK", word);
-        *skipsize = off + 2;
-        return FALSE;
-      } else {
-        /* ok, got sync now, let's assume constant frame size */
-        gst_base_parse_set_min_frame_size (parse, *framesize);
-      }
-    }
-  }
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_ac3_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
-  GstAc3Parse *ac3parse = GST_AC3_PARSE (parse);
-  GstBuffer *buf = frame->buffer;
-  guint fsize, rate, chans, blocks, sid;
-  gboolean eac;
-
-  if (!gst_ac3_parse_frame_header (ac3parse, buf, &fsize, &rate, &chans,
-          &blocks, &sid, &eac))
-    goto broken_header;
-
-  GST_LOG_OBJECT (parse, "size: %u, rate: %u, chans: %u", fsize, rate, chans);
-
-  if (G_UNLIKELY (sid)) {
-    /* dependent frame, no need to (ac)count for or consider further */
-    GST_LOG_OBJECT (parse, "sid: %d", sid);
-    frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME;
-    /* TODO maybe also mark as DELTA_UNIT,
-     * if that does not surprise baseparse elsewhere */
-    /* occupies same time space as previous base frame */
-    if (G_LIKELY (GST_BUFFER_TIMESTAMP (buf) >= GST_BUFFER_DURATION (buf)))
-      GST_BUFFER_TIMESTAMP (buf) -= GST_BUFFER_DURATION (buf);
-    /* only return if we already arranged for caps */
-    if (G_LIKELY (ac3parse->sample_rate > 0))
-      return GST_FLOW_OK;
-  }
-
-  if (G_UNLIKELY (ac3parse->sample_rate != rate || ac3parse->channels != chans
-          || ac3parse->eac != ac3parse->eac)) {
-    GstCaps *caps = gst_caps_new_simple (eac ? "audio/x-eac3" : "audio/x-ac3",
-        "framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate,
-        "channels", G_TYPE_INT, chans, NULL);
-    gst_buffer_set_caps (buf, caps);
-    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
-    gst_caps_unref (caps);
-
-    ac3parse->sample_rate = rate;
-    ac3parse->channels = chans;
-    ac3parse->eac = eac;
-
-    gst_base_parse_set_frame_props (parse, rate, 256 * blocks, 2, 2);
-  }
-
-  return GST_FLOW_OK;
-
-/* ERRORS */
-broken_header:
-  {
-    /* this really shouldn't ever happen */
-    GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL));
-    return GST_FLOW_ERROR;
-  }
-}
diff --git a/gst/audioparsers/gstac3parse.h b/gst/audioparsers/gstac3parse.h
deleted file mode 100644
index 781554be5..000000000
--- a/gst/audioparsers/gstac3parse.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/* GStreamer AC3 parser
- * Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
- * Copyright (C) 2009 Mark Nauwelaerts <mnauw users sf net>
- * Copyright (C) 2009 Nokia Corporation. All rights reserved.
- *   Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * 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_AC3_PARSE_H__
-#define __GST_AC3_PARSE_H__
-
-#include <gst/gst.h>
-#include <gst/baseparse/gstbaseparse.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AC3_PARSE \
-  (gst_ac3_parse_get_type())
-#define GST_AC3_PARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AC3_PARSE, GstAc3Parse))
-#define GST_AC3_PARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AC3_PARSE, GstAc3ParseClass))
-#define GST_IS_AC3_PARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AC3_PARSE))
-#define GST_IS_AC3_PARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AC3_PARSE))
-
-typedef struct _GstAc3Parse GstAc3Parse;
-typedef struct _GstAc3ParseClass GstAc3ParseClass;
-
-/**
- * GstAc3Parse:
- *
- * The opaque GstAc3Parse object
- */
-struct _GstAc3Parse {
-  GstBaseParse baseparse;
-
-  /*< private >*/
-  gint         sample_rate;
-  gint         channels;
-  gboolean     eac;
-};
-
-/**
- * GstAc3ParseClass:
- * @parent_class: Element parent class.
- *
- * The opaque GstAc3ParseClass data structure.
- */
-struct _GstAc3ParseClass {
-  GstBaseParseClass baseparse_class;
-};
-
-GType gst_ac3_parse_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_AC3_PARSE_H__ */
diff --git a/gst/audioparsers/gstamrparse.c b/gst/audioparsers/gstamrparse.c
deleted file mode 100644
index 42481a2b2..000000000
--- a/gst/audioparsers/gstamrparse.c
+++ /dev/null
@@ -1,378 +0,0 @@
-/* GStreamer Adaptive Multi-Rate parser plugin
- * Copyright (C) 2006 Edgard Lima <edgard.lima@indt.org.br>
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-amrparse
- * @short_description: AMR parser
- * @see_also: #GstAmrnbDec, #GstAmrnbEnc
- *
- * This is an AMR parser capable of handling both narrow-band and wideband
- * formats.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink
- * ]|
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-
-#include "gstamrparse.h"
-
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1;"
-        "audio/AMR-WB, " "rate = (int) 16000, " "channels = (int) 1;")
-    );
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-amr-nb-sh; audio/x-amr-wb-sh"));
-
-GST_DEBUG_CATEGORY_STATIC (gst_amrparse_debug);
-#define GST_CAT_DEFAULT gst_amrparse_debug
-
-static const gint block_size_nb[16] =
-    { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
-
-static const gint block_size_wb[16] =
-    { 17, 23, 32, 36, 40, 46, 50, 58, 60, 5, -1, -1, -1, -1, 0, 0 };
-
-/* AMR has a "hardcoded" framerate of 50fps */
-#define AMR_FRAMES_PER_SECOND 50
-#define AMR_FRAME_DURATION (GST_SECOND/AMR_FRAMES_PER_SECOND)
-#define AMR_MIME_HEADER_SIZE 9
-
-gboolean gst_amrparse_start (GstBaseParse * parse);
-gboolean gst_amrparse_stop (GstBaseParse * parse);
-
-static gboolean gst_amrparse_sink_setcaps (GstBaseParse * parse,
-    GstCaps * caps);
-
-gboolean gst_amrparse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize);
-
-GstFlowReturn gst_amrparse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-
-#define _do_init(bla) \
-    GST_DEBUG_CATEGORY_INIT (gst_amrparse_debug, "amrparse", 0, \
-                             "AMR-NB audio stream parser");
-
-GST_BOILERPLATE_FULL (GstAmrParse, gst_amrparse, GstBaseParse,
-    GST_TYPE_BASE_PARSE, _do_init);
-
-
-/**
- * gst_amrparse_base_init:
- * @klass: #GstElementClass.
- *
- */
-static void
-gst_amrparse_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_template));
-
-  gst_element_class_set_details_simple (element_class,
-      "AMR audio stream parser", "Codec/Parser/Audio",
-      "Adaptive Multi-Rate audio parser",
-      "Ronald Bultje <rbultje@ronald.bitfreak.net>");
-}
-
-
-/**
- * gst_amrparse_class_init:
- * @klass: GstAmrParseClass.
- *
- */
-static void
-gst_amrparse_class_init (GstAmrParseClass * klass)
-{
-  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
-
-  parse_class->start = GST_DEBUG_FUNCPTR (gst_amrparse_start);
-  parse_class->stop = GST_DEBUG_FUNCPTR (gst_amrparse_stop);
-  parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amrparse_sink_setcaps);
-  parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_amrparse_parse_frame);
-  parse_class->check_valid_frame =
-      GST_DEBUG_FUNCPTR (gst_amrparse_check_valid_frame);
-}
-
-
-/**
- * gst_amrparse_init:
- * @amrparse: #GstAmrParse
- * @klass: #GstAmrParseClass.
- *
- */
-static void
-gst_amrparse_init (GstAmrParse * amrparse, GstAmrParseClass * klass)
-{
-  /* init rest */
-  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);
-  GST_DEBUG ("initialized");
-
-}
-
-
-/**
- * gst_amrparse_set_src_caps:
- * @amrparse: #GstAmrParse.
- *
- * Set source pad caps according to current knowledge about the
- * audio stream.
- *
- * Returns: TRUE if caps were successfully set.
- */
-static gboolean
-gst_amrparse_set_src_caps (GstAmrParse * amrparse)
-{
-  GstCaps *src_caps = NULL;
-  gboolean res = FALSE;
-
-  if (amrparse->wide) {
-    GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-WB");
-    src_caps = gst_caps_new_simple ("audio/AMR-WB",
-        "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 16000, NULL);
-  } else {
-    GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-NB");
-    /* Max. size of NB frame is 31 bytes, so we can set the min. frame
-       size to 32 (+1 for next frame header) */
-    gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 32);
-    src_caps = gst_caps_new_simple ("audio/AMR",
-        "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
-  }
-  gst_pad_use_fixed_caps (GST_BASE_PARSE (amrparse)->srcpad);
-  res = gst_pad_set_caps (GST_BASE_PARSE (amrparse)->srcpad, src_caps);
-  gst_caps_unref (src_caps);
-  return res;
-}
-
-
-/**
- * gst_amrparse_sink_setcaps:
- * @sinkpad: GstPad
- * @caps: GstCaps
- *
- * Returns: TRUE on success.
- */
-static gboolean
-gst_amrparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
-{
-  GstAmrParse *amrparse;
-  GstStructure *structure;
-  const gchar *name;
-
-  amrparse = GST_AMRPARSE (parse);
-  structure = gst_caps_get_structure (caps, 0);
-  name = gst_structure_get_name (structure);
-
-  GST_DEBUG_OBJECT (amrparse, "setcaps: %s", name);
-
-  if (!strncmp (name, "audio/x-amr-wb-sh", 17)) {
-    amrparse->block_size = block_size_wb;
-    amrparse->wide = 1;
-  } else if (!strncmp (name, "audio/x-amr-nb-sh", 17)) {
-    amrparse->block_size = block_size_nb;
-    amrparse->wide = 0;
-  } else {
-    GST_WARNING ("Unknown caps");
-    return FALSE;
-  }
-
-  amrparse->need_header = FALSE;
-  gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);
-  gst_amrparse_set_src_caps (amrparse);
-  return TRUE;
-}
-
-/**
- * gst_amrparse_parse_header:
- * @amrparse: #GstAmrParse
- * @data: Header data to be parsed.
- * @skipsize: Output argument where the frame size will be stored.
- *
- * Check if the given data contains an AMR mime header.
- *
- * Returns: TRUE on success.
- */
-static gboolean
-gst_amrparse_parse_header (GstAmrParse * amrparse,
-    const guint8 * data, gint * skipsize)
-{
-  GST_DEBUG_OBJECT (amrparse, "Parsing header data");
-
-  if (!memcmp (data, "#!AMR-WB\n", 9)) {
-    GST_DEBUG_OBJECT (amrparse, "AMR-WB detected");
-    amrparse->block_size = block_size_wb;
-    amrparse->wide = TRUE;
-    *skipsize = amrparse->header = 9;
-  } else if (!memcmp (data, "#!AMR\n", 6)) {
-    GST_DEBUG_OBJECT (amrparse, "AMR-NB detected");
-    amrparse->block_size = block_size_nb;
-    amrparse->wide = FALSE;
-    *skipsize = amrparse->header = 6;
-  } else
-    return FALSE;
-
-  gst_amrparse_set_src_caps (amrparse);
-  return TRUE;
-}
-
-
-/**
- * gst_amrparse_check_valid_frame:
- * @parse: #GstBaseParse.
- * @buffer: #GstBuffer.
- * @framesize: Output variable where the found frame size is put.
- * @skipsize: Output variable which tells how much data needs to be skipped
- *            until a frame header is found.
- *
- * Implementation of "check_valid_frame" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE if the given data contains valid frame.
- */
-gboolean
-gst_amrparse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
-{
-  GstBuffer *buffer;
-  const guint8 *data;
-  gint fsize, mode, dsize;
-  GstAmrParse *amrparse;
-
-  amrparse = GST_AMRPARSE (parse);
-  buffer = frame->buffer;
-  data = GST_BUFFER_DATA (buffer);
-  dsize = GST_BUFFER_SIZE (buffer);
-
-  GST_LOG ("buffer: %d bytes", dsize);
-
-  if (amrparse->need_header) {
-    if (dsize >= AMR_MIME_HEADER_SIZE &&
-        gst_amrparse_parse_header (amrparse, data, skipsize)) {
-      amrparse->need_header = FALSE;
-      gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);
-    } else {
-      GST_WARNING ("media doesn't look like a AMR format");
-    }
-    /* We return FALSE, so this frame won't get pushed forward. Instead,
-       the "skip" value is set, so next time we will receive a valid frame. */
-    return FALSE;
-  }
-
-  /* Does this look like a possible frame header candidate? */
-  if ((data[0] & 0x83) == 0) {
-    /* Yep. Retrieve the frame size */
-    mode = (data[0] >> 3) & 0x0F;
-    fsize = amrparse->block_size[mode] + 1;     /* +1 for the header byte */
-
-    /* We recognize this data as a valid frame when:
-     *     - We are in sync. There is no need for extra checks then
-     *     - We are in EOS. There might not be enough data to check next frame
-     *     - Sync is lost, but the following data after this frame seem
-     *       to contain a valid header as well (and there is enough data to
-     *       perform this check)
-     */
-    if (fsize &&
-        (GST_BASE_PARSE_FRAME_SYNC (frame) || GST_BASE_PARSE_FRAME_DRAIN (frame)
-            || (dsize > fsize && (data[fsize] & 0x83) == 0))) {
-      *framesize = fsize;
-      return TRUE;
-    }
-  }
-
-  GST_LOG ("sync lost");
-  return FALSE;
-}
-
-
-/**
- * gst_amrparse_parse_frame:
- * @parse: #GstBaseParse.
- * @buffer: #GstBuffer.
- *
- * Implementation of "parse" vmethod in #GstBaseParse class.
- *
- * Returns: #GstFlowReturn defining the parsing status.
- */
-GstFlowReturn
-gst_amrparse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
-  return GST_FLOW_OK;
-}
-
-
-/**
- * gst_amrparse_start:
- * @parse: #GstBaseParse.
- *
- * Implementation of "start" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE on success.
- */
-gboolean
-gst_amrparse_start (GstBaseParse * parse)
-{
-  GstAmrParse *amrparse;
-
-  amrparse = GST_AMRPARSE (parse);
-  GST_DEBUG ("start");
-  amrparse->need_header = TRUE;
-  amrparse->header = 0;
-  return TRUE;
-}
-
-
-/**
- * gst_amrparse_stop:
- * @parse: #GstBaseParse.
- *
- * Implementation of "stop" vmethod in #GstBaseParse class.
- *
- * Returns: TRUE on success.
- */
-gboolean
-gst_amrparse_stop (GstBaseParse * parse)
-{
-  GstAmrParse *amrparse;
-
-  amrparse = GST_AMRPARSE (parse);
-  GST_DEBUG ("stop");
-  amrparse->need_header = TRUE;
-  amrparse->header = 0;
-  return TRUE;
-}
diff --git a/gst/audioparsers/gstamrparse.h b/gst/audioparsers/gstamrparse.h
deleted file mode 100644
index 04cd6e763..000000000
--- a/gst/audioparsers/gstamrparse.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/* GStreamer Adaptive Multi-Rate parser
- * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * 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_AMRPARSE_H__
-#define __GST_AMRPARSE_H__
-
-#include <gst/gst.h>
-#include <gst/baseparse/gstbaseparse.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AMRPARSE \
-  (gst_amrparse_get_type())
-#define GST_AMRPARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AMRPARSE, GstAmrParse))
-#define GST_AMRPARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AMRPARSE, GstAmrParseClass))
-#define GST_IS_AMRPARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AMRPARSE))
-#define GST_IS_AMRPARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AMRPARSE))
-
-
-typedef struct _GstAmrParse GstAmrParse;
-typedef struct _GstAmrParseClass GstAmrParseClass;
-
-/**
- * GstAmrParse:
- * @element: the parent element.
- * @block_size: Pointer to frame size lookup table.
- * @need_header: Tells whether the MIME header should be read in the beginning.
- * @wide: Wideband mode.
- * @eos: Indicates the EOS situation. Set when EOS event is received.
- * @sync: Tells whether the parser is in sync.
- * @framecount: Total amount of frames handled.
- * @bytecount: Total amount of bytes handled.
- * @ts: Timestamp of the current media.
- *
- * The opaque GstAacParse data structure.
- */
-struct _GstAmrParse {
-  GstBaseParse element;
-  const gint *block_size;
-  gboolean need_header;
-  gint header;
-  gboolean wide;
-};
-
-/**
- * GstAmrParseClass:
- * @parent_class: Element parent class.
- *
- * The opaque GstAmrParseClass data structure.
- */
-struct _GstAmrParseClass {
-  GstBaseParseClass parent_class;
-};
-
-GType gst_amrparse_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_AMRPARSE_H__ */
diff --git a/gst/audioparsers/gstdcaparse.c b/gst/audioparsers/gstdcaparse.c
deleted file mode 100644
index 7e478e47d..000000000
--- a/gst/audioparsers/gstdcaparse.c
+++ /dev/null
@@ -1,451 +0,0 @@
-/* GStreamer DCA parser
- * Copyright (C) 2010 Tim-Philipp Müller <tim centricular net>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-dcaparse
- * @short_description: DCA (DTS Coherent Acoustics) parser
- * @see_also: #GstAmrParse, #GstAACParse, #GstAc3Parse
- *
- * This is a DCA (DTS Coherent Acoustics) parser.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
- * ]|
- * </refsect2>
- */
-
-/* TODO:
- *  - should accept framed and unframed input (needs decodebin fixes first)
- *  - seeking in raw .dts files doesn't seem to work, but duration estimate ok
- *
- *  - if frames have 'odd' durations, the frame durations (plus timestamps)
- *    aren't adjusted up occasionally to make up for rounding error gaps.
- *    (e.g. if 512 samples per frame @ 48kHz = 10.666666667 ms/frame)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-
-#include "gstdcaparse.h"
-#include <gst/base/gstbytereader.h>
-#include <gst/base/gstbitreader.h>
-
-GST_DEBUG_CATEGORY_STATIC (dca_parse_debug);
-#define GST_CAT_DEFAULT dca_parse_debug
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-dts,"
-        " framed = (boolean) true,"
-        " channels = (int) [ 1, 8 ],"
-        " rate = (int) [ 8000, 192000 ],"
-        " depth = (int) { 14, 16 },"
-        " endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }"));
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-dts, framed = (boolean) false"));
-
-static void gst_dca_parse_finalize (GObject * object);
-
-static gboolean gst_dca_parse_start (GstBaseParse * parse);
-static gboolean gst_dca_parse_stop (GstBaseParse * parse);
-static gboolean gst_dca_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * size, gint * skipsize);
-static GstFlowReturn gst_dca_parse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-
-GST_BOILERPLATE (GstDcaParse, gst_dca_parse, GstBaseParse, GST_TYPE_BASE_PARSE);
-
-static void
-gst_dca_parse_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_template));
-
-  gst_element_class_set_details_simple (element_class,
-      "DTS Coherent Acoustics audio stream parser", "Codec/Parser/Audio",
-      "DCA parser", "Tim-Philipp Müller <tim centricular net>");
-}
-
-static void
-gst_dca_parse_class_init (GstDcaParseClass * klass)
-{
-  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
-  GObjectClass *object_class = G_OBJECT_CLASS (klass);
-
-  GST_DEBUG_CATEGORY_INIT (dca_parse_debug, "dcaparse", 0,
-      "DCA audio stream parser");
-
-  object_class->finalize = gst_dca_parse_finalize;
-
-  parse_class->start = GST_DEBUG_FUNCPTR (gst_dca_parse_start);
-  parse_class->stop = GST_DEBUG_FUNCPTR (gst_dca_parse_stop);
-  parse_class->check_valid_frame =
-      GST_DEBUG_FUNCPTR (gst_dca_parse_check_valid_frame);
-  parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_parse_frame);
-}
-
-static void
-gst_dca_parse_reset (GstDcaParse * dcaparse)
-{
-  dcaparse->channels = -1;
-  dcaparse->rate = -1;
-  dcaparse->depth = -1;
-  dcaparse->endianness = -1;
-  dcaparse->block_size = -1;
-  dcaparse->frame_size = -1;
-  dcaparse->last_sync = 0;
-}
-
-static void
-gst_dca_parse_init (GstDcaParse * dcaparse, GstDcaParseClass * klass)
-{
-  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (dcaparse),
-      DCA_MIN_FRAMESIZE);
-  gst_dca_parse_reset (dcaparse);
-}
-
-static void
-gst_dca_parse_finalize (GObject * object)
-{
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_dca_parse_start (GstBaseParse * parse)
-{
-  GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
-
-  GST_DEBUG_OBJECT (parse, "starting");
-
-  gst_dca_parse_reset (dcaparse);
-
-  return TRUE;
-}
-
-static gboolean
-gst_dca_parse_stop (GstBaseParse * parse)
-{
-  GST_DEBUG_OBJECT (parse, "stopping");
-
-  return TRUE;
-}
-
-static gboolean
-gst_dca_parse_parse_header (GstDcaParse * dcaparse,
-    const GstByteReader * reader, guint * frame_size,
-    guint * sample_rate, guint * channels, guint * depth,
-    gint * endianness, guint * num_blocks, guint * samples_per_block,
-    gboolean * terminator)
-{
-  static const int sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025,
-    22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000
-  };
-  static const guint8 channels_table[16] = { 1, 2, 2, 2, 2, 3, 3, 4, 4, 5,
-    6, 6, 6, 7, 8, 8
-  };
-  GstByteReader r = *reader;
-  guint16 hdr[8];
-  guint32 marker;
-  guint chans, lfe, i;
-
-  if (gst_byte_reader_get_remaining (&r) < (4 + sizeof (hdr)))
-    return FALSE;
-
-  marker = gst_byte_reader_peek_uint32_be_unchecked (&r);
-
-  /* raw big endian or 14-bit big endian */
-  if (marker == 0x7FFE8001 || marker == 0x1FFFE800) {
-    for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
-      hdr[i] = gst_byte_reader_get_uint16_be_unchecked (&r);
-  } else
-    /* raw little endian or 14-bit little endian */
-  if (marker == 0xFE7F0180 || marker == 0xFF1F00E8) {
-    for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
-      hdr[i] = gst_byte_reader_get_uint16_le_unchecked (&r);
-  } else {
-    return FALSE;
-  }
-
-  GST_LOG_OBJECT (dcaparse, "dts sync marker 0x%08x at offset %u", marker,
-      gst_byte_reader_get_pos (reader));
-
-  /* 14-bit mode */
-  if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) {
-    if ((hdr[2] & 0xFFF0) != 0x07F0)
-      return FALSE;
-    /* discard top 2 bits (2 void), shift in 2 */
-    hdr[0] = (hdr[0] << 2) | ((hdr[1] >> 12) & 0x0003);
-    /* discard top 4 bits (2 void, 2 shifted into hdr[0]), shift in 4 etc. */
-    hdr[1] = (hdr[1] << 4) | ((hdr[2] >> 10) & 0x000F);
-    hdr[2] = (hdr[2] << 6) | ((hdr[3] >> 8) & 0x003F);
-    hdr[3] = (hdr[3] << 8) | ((hdr[4] >> 6) & 0x00FF);
-    hdr[4] = (hdr[4] << 10) | ((hdr[5] >> 4) & 0x03FF);
-    hdr[5] = (hdr[5] << 12) | ((hdr[6] >> 2) & 0x0FFF);
-    hdr[6] = (hdr[6] << 14) | ((hdr[7] >> 0) & 0x3FFF);
-    g_assert (hdr[0] == 0x7FFE && hdr[1] == 0x8001);
-  }
-
-  GST_LOG_OBJECT (dcaparse, "frame header: %04x%04x%04x%04x",
-      hdr[2], hdr[3], hdr[4], hdr[5]);
-
-  *terminator = (hdr[2] & 0x80) ? FALSE : TRUE;
-  *samples_per_block = ((hdr[2] >> 10) & 0x1f) + 1;
-  *num_blocks = ((hdr[2] >> 2) & 0x7F) + 1;
-  *frame_size = (((hdr[2] & 0x03) << 12) | (hdr[3] >> 4)) + 1;
-  chans = ((hdr[3] & 0x0F) << 2) | (hdr[4] >> 14);
-  *sample_rate = sample_rates[(hdr[4] >> 10) & 0x0F];
-  lfe = (hdr[5] >> 9) & 0x03;
-
-  GST_TRACE_OBJECT (dcaparse, "frame size %u, num_blocks %u, rate %u, "
-      "samples per block %u", *frame_size, *num_blocks, *sample_rate,
-      *samples_per_block);
-
-  if (*num_blocks < 6 || *frame_size < 96 || *sample_rate == 0)
-    return FALSE;
-
-  if (marker == 0x1FFFE800 || marker == 0xFF1F00E8)
-    *frame_size = (*frame_size * 16) / 14;      /* FIXME: round up? */
-
-  if (chans < G_N_ELEMENTS (channels_table))
-    *channels = channels_table[chans] + ((lfe) ? 1 : 0);
-  else
-    *channels = 0;
-
-  if (depth)
-    *depth = (marker == 0x1FFFE800 || marker == 0xFF1F00E8) ? 14 : 16;
-  if (endianness)
-    *endianness = (marker == 0xFE7F0180 || marker == 0xFF1F00E8) ?
-        G_LITTLE_ENDIAN : G_BIG_ENDIAN;
-
-  GST_TRACE_OBJECT (dcaparse, "frame size %u, channels %u, rate %u, "
-      "num_blocks %u, samples_per_block %u", *frame_size, *channels,
-      *sample_rate, *num_blocks, *samples_per_block);
-
-  return TRUE;
-}
-
-static gint
-gst_dca_parse_find_sync (GstDcaParse * dcaparse, GstByteReader * reader,
-    const GstBuffer * buf, guint32 * sync)
-{
-  guint32 best_sync = 0;
-  guint best_offset = G_MAXUINT;
-  gint off;
-
-  /* FIXME: verify syncs via _parse_header() here already */
-
-  /* Raw little endian */
-  off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xfe7f0180,
-      0, GST_BUFFER_SIZE (buf));
-  if (off >= 0 && off < best_offset) {
-    best_offset = off;
-    best_sync = 0xfe7f0180;
-  }
-
-  /* Raw big endian */
-  off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x7ffe8001,
-      0, GST_BUFFER_SIZE (buf));
-  if (off >= 0 && off < best_offset) {
-    best_offset = off;
-    best_sync = 0x7ffe8001;
-  }
-
-  /* FIXME: check next 2 bytes as well for 14-bit formats (but then don't
-   * forget to adjust the *skipsize= in _check_valid_frame() */
-
-  /* 14-bit little endian  */
-  off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xff1f00e8,
-      0, GST_BUFFER_SIZE (buf));
-  if (off >= 0 && off < best_offset) {
-    best_offset = off;
-    best_sync = 0xff1f00e8;
-  }
-
-  /* 14-bit big endian  */
-  off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x1fffe800,
-      0, GST_BUFFER_SIZE (buf));
-  if (off >= 0 && off < best_offset) {
-    best_offset = off;
-    best_sync = 0x1fffe800;
-  }
-
-  if (best_offset == G_MAXUINT)
-    return -1;
-
-  *sync = best_sync;
-  return best_offset;
-}
-
-static gboolean
-gst_dca_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
-{
-  GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
-  GstBuffer *buf = frame->buffer;
-  GstByteReader r = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
-  gboolean parser_draining;
-  gboolean parser_in_sync;
-  gboolean terminator;
-  guint32 sync = 0;
-  guint size, rate, chans, num_blocks, samples_per_block;
-  gint off = -1;
-
-  if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 16))
-    return FALSE;
-
-  parser_in_sync = GST_BASE_PARSE_FRAME_SYNC (frame);
-
-  if (G_LIKELY (parser_in_sync && dcaparse->last_sync != 0)) {
-    off = gst_byte_reader_masked_scan_uint32 (&r, 0xffffffff,
-        dcaparse->last_sync, 0, GST_BUFFER_SIZE (buf));
-  }
-
-  if (G_UNLIKELY (off < 0)) {
-    off = gst_dca_parse_find_sync (dcaparse, &r, buf, &sync);
-  }
-
-  /* didn't find anything that looks like a sync word, skip */
-  if (off < 0) {
-    *skipsize = GST_BUFFER_SIZE (buf) - 3;
-    GST_DEBUG_OBJECT (dcaparse, "no sync, skipping %d bytes", *skipsize);
-    return FALSE;
-  }
-
-  GST_LOG_OBJECT (parse, "possible sync %08x at buffer offset %d", sync, off);
-
-  /* possible frame header, but not at offset 0? skip bytes before sync */
-  if (off > 0) {
-    *skipsize = off;
-    return FALSE;
-  }
-
-  /* make sure the values in the frame header look sane */
-  if (!gst_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, NULL,
-          NULL, &num_blocks, &samples_per_block, &terminator)) {
-    *skipsize = 4;
-    return FALSE;
-  }
-
-  GST_LOG_OBJECT (parse, "got frame, sync %08x, size %u, rate %d, channels %d",
-      sync, size, rate, chans);
-
-  *framesize = size;
-
-  dcaparse->last_sync = sync;
-
-  parser_draining = GST_BASE_PARSE_FRAME_DRAIN (frame);
-
-  if (!parser_in_sync && !parser_draining) {
-    /* check for second frame to be sure */
-    GST_DEBUG_OBJECT (dcaparse, "resyncing; checking next frame syncword");
-    if (GST_BUFFER_SIZE (buf) >= (size + 16)) {
-      guint s2, r2, c2, n2, s3;
-      gboolean t;
-
-      GST_MEMDUMP ("buf", GST_BUFFER_DATA (buf), size + 16);
-      gst_byte_reader_init_from_buffer (&r, buf);
-      gst_byte_reader_skip_unchecked (&r, size);
-
-      if (!gst_dca_parse_parse_header (dcaparse, &r, &s2, &r2, &c2, NULL, NULL,
-              &n2, &s3, &t)) {
-        GST_DEBUG_OBJECT (dcaparse, "didn't find second syncword");
-        *skipsize = 4;
-        return FALSE;
-      }
-
-      /* ok, got sync now, let's assume constant frame size */
-      gst_base_parse_set_min_frame_size (parse, size);
-    } else {
-      /* FIXME: baseparse always seems to hand us buffers of min_frame_size
-       * bytes, which is unhelpful here */
-      GST_LOG_OBJECT (dcaparse, "next sync out of reach (%u < %u)",
-          GST_BUFFER_SIZE (buf), size + 16);
-      /* *skipsize = 0; */
-      /* return FALSE; */
-    }
-  }
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_dca_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
-  GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
-  GstBuffer *buf = frame->buffer;
-  GstByteReader r = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
-  guint size, rate, chans, depth, block_size, num_blocks, samples_per_block;
-  gint endianness;
-  gboolean terminator;
-
-  if (!gst_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, &depth,
-          &endianness, &num_blocks, &samples_per_block, &terminator))
-    goto broken_header;
-
-  block_size = num_blocks * samples_per_block;
-
-  if (G_UNLIKELY (dcaparse->rate != rate || dcaparse->channels != chans
-          || dcaparse->depth != depth || dcaparse->endianness != endianness
-          || (!terminator && dcaparse->block_size != block_size)
-          || (size != dcaparse->frame_size))) {
-    GstCaps *caps;
-
-    caps = gst_caps_new_simple ("audio/x-dts",
-        "framed", G_TYPE_BOOLEAN, TRUE,
-        "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans,
-        "endianness", G_TYPE_INT, endianness, "depth", G_TYPE_INT, depth,
-        "block-size", G_TYPE_INT, block_size, "frame-size", G_TYPE_INT, size,
-        NULL);
-    gst_buffer_set_caps (buf, caps);
-    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
-    gst_caps_unref (caps);
-
-    dcaparse->rate = rate;
-    dcaparse->channels = chans;
-    dcaparse->depth = depth;
-    dcaparse->endianness = endianness;
-    dcaparse->block_size = block_size;
-    dcaparse->frame_size = size;
-
-    gst_base_parse_set_frame_props (parse, rate, block_size, 0, 0);
-  }
-
-  return GST_FLOW_OK;
-
-/* ERRORS */
-broken_header:
-  {
-    /* this really shouldn't ever happen */
-    GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL));
-    return GST_FLOW_ERROR;
-  }
-}
diff --git a/gst/audioparsers/gstdcaparse.h b/gst/audioparsers/gstdcaparse.h
deleted file mode 100644
index eefc2526c..000000000
--- a/gst/audioparsers/gstdcaparse.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/* GStreamer DCA parser
- * Copyright (C) 2010 Tim-Philipp Müller <tim centricular 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_DCA_PARSE_H__
-#define __GST_DCA_PARSE_H__
-
-#include <gst/gst.h>
-#include <gst/baseparse/gstbaseparse.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_DCA_PARSE \
-  (gst_dca_parse_get_type())
-#define GST_DCA_PARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_DCA_PARSE, GstDcaParse))
-#define GST_DCA_PARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_DCA_PARSE, GstDcaParseClass))
-#define GST_IS_DCA_PARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_DCA_PARSE))
-#define GST_IS_DCA_PARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_DCA_PARSE))
-
-#define DCA_MIN_FRAMESIZE 96
-#define DCA_MAX_FRAMESIZE 18725 /* 16384*16/14 */
-
-typedef struct _GstDcaParse GstDcaParse;
-typedef struct _GstDcaParseClass GstDcaParseClass;
-
-/**
- * GstDcaParse:
- *
- * The opaque GstDcaParse object
- */
-struct _GstDcaParse {
-  GstBaseParse baseparse;
-
-  /*< private >*/
-  gint         rate;
-  gint         channels;
-  gint         depth;
-  gint         endianness;
-  gint         block_size;
-  gint         frame_size;
-
-  guint32      last_sync;
-};
-
-/**
- * GstDcaParseClass:
- * @parent_class: Element parent class.
- *
- * The opaque GstDcaParseClass data structure.
- */
-struct _GstDcaParseClass {
-  GstBaseParseClass baseparse_class;
-};
-
-GType gst_dca_parse_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_DCA_PARSE_H__ */
diff --git a/gst/audioparsers/gstflacparse.c b/gst/audioparsers/gstflacparse.c
deleted file mode 100644
index 8306e8e8b..000000000
--- a/gst/audioparsers/gstflacparse.c
+++ /dev/null
@@ -1,1354 +0,0 @@
-/* GStreamer
- *
- * Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>.
- * Copyright (C) 2009 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * Copyright (C) 2009 Nokia Corporation. All rights reserved.
- *   Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-flacparse
- * @see_also: flacdec, oggdemux, vorbisparse
- *
- * The flacparse element will parse the header packets of the FLAC
- * stream and put them as the streamheader in the caps. This is used in the
- * multifdsink case where you want to stream live FLAC streams to multiple
- * clients, each client has to receive the streamheaders first before they can
- * consume the FLAC packets.
- *
- * This element also makes sure that the buffers that it pushes out are properly
- * timestamped and that their offset and offset_end are set. The buffers that
- * flacparse outputs have all of the metadata that oggmux expects to receive,
- * which allows you to (for example) remux an ogg/flac or convert a native FLAC
- * format file to an ogg bitstream.
- *
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * gst-launch -v filesrc location=sine.flac ! flacparse ! identity \
- *            ! oggmux ! filesink location=sine-remuxed.ogg
- * ]| This pipeline converts a native FLAC format file to an ogg bitstream.
- * It also illustrates that the streamheader is set in the caps, and that each
- * buffer has the timestamp, duration, offset, and offset_end set.
- * </refsect2>
- *
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "gstflacparse.h"
-
-#include <string.h>
-#include <gst/tag/tag.h>
-#include <gst/audio/audio.h>
-
-#include <gst/base/gstbitreader.h>
-#include <gst/base/gstbytereader.h>
-
-GST_DEBUG_CATEGORY_STATIC (flacparse_debug);
-#define GST_CAT_DEFAULT flacparse_debug
-
-/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
-static const guint8 crc8_table[256] = {
-  0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
-  0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
-  0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
-  0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
-  0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
-  0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
-  0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
-  0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
-  0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
-  0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
-  0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
-  0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
-  0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
-  0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
-  0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
-  0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
-  0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
-  0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
-  0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
-  0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
-  0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
-  0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
-  0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
-  0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
-  0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
-  0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
-  0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
-  0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
-  0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
-  0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
-  0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
-  0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
-};
-
-static guint8
-gst_flac_calculate_crc8 (const guint8 * data, guint length)
-{
-  guint8 crc = 0;
-
-  while (length--) {
-    crc = crc8_table[crc ^ *data];
-    ++data;
-  }
-
-  return crc;
-}
-
-/* CRC-16, poly = x^16 + x^15 + x^2 + x^0, init = 0 */
-static const guint16 crc16_table[256] = {
-  0x0000, 0x8005, 0x800f, 0x000a, 0x801b, 0x001e, 0x0014, 0x8011,
-  0x8033, 0x0036, 0x003c, 0x8039, 0x0028, 0x802d, 0x8027, 0x0022,
-  0x8063, 0x0066, 0x006c, 0x8069, 0x0078, 0x807d, 0x8077, 0x0072,
-  0x0050, 0x8055, 0x805f, 0x005a, 0x804b, 0x004e, 0x0044, 0x8041,
-  0x80c3, 0x00c6, 0x00cc, 0x80c9, 0x00d8, 0x80dd, 0x80d7, 0x00d2,
-  0x00f0, 0x80f5, 0x80ff, 0x00fa, 0x80eb, 0x00ee, 0x00e4, 0x80e1,
-  0x00a0, 0x80a5, 0x80af, 0x00aa, 0x80bb, 0x00be, 0x00b4, 0x80b1,
-  0x8093, 0x0096, 0x009c, 0x8099, 0x0088, 0x808d, 0x8087, 0x0082,
-  0x8183, 0x0186, 0x018c, 0x8189, 0x0198, 0x819d, 0x8197, 0x0192,
-  0x01b0, 0x81b5, 0x81bf, 0x01ba, 0x81ab, 0x01ae, 0x01a4, 0x81a1,
-  0x01e0, 0x81e5, 0x81ef, 0x01ea, 0x81fb, 0x01fe, 0x01f4, 0x81f1,
-  0x81d3, 0x01d6, 0x01dc, 0x81d9, 0x01c8, 0x81cd, 0x81c7, 0x01c2,
-  0x0140, 0x8145, 0x814f, 0x014a, 0x815b, 0x015e, 0x0154, 0x8151,
-  0x8173, 0x0176, 0x017c, 0x8179, 0x0168, 0x816d, 0x8167, 0x0162,
-  0x8123, 0x0126, 0x012c, 0x8129, 0x0138, 0x813d, 0x8137, 0x0132,
-  0x0110, 0x8115, 0x811f, 0x011a, 0x810b, 0x010e, 0x0104, 0x8101,
-  0x8303, 0x0306, 0x030c, 0x8309, 0x0318, 0x831d, 0x8317, 0x0312,
-  0x0330, 0x8335, 0x833f, 0x033a, 0x832b, 0x032e, 0x0324, 0x8321,
-  0x0360, 0x8365, 0x836f, 0x036a, 0x837b, 0x037e, 0x0374, 0x8371,
-  0x8353, 0x0356, 0x035c, 0x8359, 0x0348, 0x834d, 0x8347, 0x0342,
-  0x03c0, 0x83c5, 0x83cf, 0x03ca, 0x83db, 0x03de, 0x03d4, 0x83d1,
-  0x83f3, 0x03f6, 0x03fc, 0x83f9, 0x03e8, 0x83ed, 0x83e7, 0x03e2,
-  0x83a3, 0x03a6, 0x03ac, 0x83a9, 0x03b8, 0x83bd, 0x83b7, 0x03b2,
-  0x0390, 0x8395, 0x839f, 0x039a, 0x838b, 0x038e, 0x0384, 0x8381,
-  0x0280, 0x8285, 0x828f, 0x028a, 0x829b, 0x029e, 0x0294, 0x8291,
-  0x82b3, 0x02b6, 0x02bc, 0x82b9, 0x02a8, 0x82ad, 0x82a7, 0x02a2,
-  0x82e3, 0x02e6, 0x02ec, 0x82e9, 0x02f8, 0x82fd, 0x82f7, 0x02f2,
-  0x02d0, 0x82d5, 0x82df, 0x02da, 0x82cb, 0x02ce, 0x02c4, 0x82c1,
-  0x8243, 0x0246, 0x024c, 0x8249, 0x0258, 0x825d, 0x8257, 0x0252,
-  0x0270, 0x8275, 0x827f, 0x027a, 0x826b, 0x026e, 0x0264, 0x8261,
-  0x0220, 0x8225, 0x822f, 0x022a, 0x823b, 0x023e, 0x0234, 0x8231,
-  0x8213, 0x0216, 0x021c, 0x8219, 0x0208, 0x820d, 0x8207, 0x0202
-};
-
-static guint16
-gst_flac_calculate_crc16 (const guint8 * data, guint length)
-{
-  guint16 crc = 0;
-
-  while (length--) {
-    crc = ((crc << 8) ^ crc16_table[(crc >> 8) ^ *data]) & 0xffff;
-    data++;
-  }
-
-  return crc;
-}
-
-enum
-{
-  PROP_0,
-  PROP_CHECK_FRAME_CHECKSUMS
-};
-
-#define DEFAULT_CHECK_FRAME_CHECKSUMS FALSE
-
-static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-flac, framed = (boolean) true, "
-        "channels = (int) [ 1, 8 ], " "rate = (int) [ 1, 655350 ]")
-    );
-
-static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-flac, framed = (boolean) false")
-    );
-
-static void gst_flac_parse_finalize (GObject * object);
-static void gst_flac_parse_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void gst_flac_parse_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-static gboolean gst_flac_parse_start (GstBaseParse * parse);
-static gboolean gst_flac_parse_stop (GstBaseParse * parse);
-static gboolean gst_flac_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize);
-static GstFlowReturn gst_flac_parse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-static GstFlowReturn gst_flac_parse_pre_push_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-
-GST_BOILERPLATE (GstFlacParse, gst_flac_parse, GstBaseParse,
-    GST_TYPE_BASE_PARSE);
-
-static void
-gst_flac_parse_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_factory));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_factory));
-
-  gst_element_class_set_details_simple (element_class, "FLAC audio parser",
-      "Codec/Parser/Audio",
-      "Parses audio with the FLAC lossless audio codec",
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-
-  GST_DEBUG_CATEGORY_INIT (flacparse_debug, "flacparse", 0,
-      "Flac parser element");
-}
-
-static void
-gst_flac_parse_class_init (GstFlacParseClass * klass)
-{
-  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
-  GstBaseParseClass *baseparse_class = GST_BASE_PARSE_CLASS (klass);
-
-  gobject_class->finalize = gst_flac_parse_finalize;
-  gobject_class->set_property = gst_flac_parse_set_property;
-  gobject_class->get_property = gst_flac_parse_get_property;
-
-  g_object_class_install_property (gobject_class, PROP_CHECK_FRAME_CHECKSUMS,
-      g_param_spec_boolean ("check-frame-checksums", "Check Frame Checksums",
-          "Check the overall checksums of every frame",
-          DEFAULT_CHECK_FRAME_CHECKSUMS,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  baseparse_class->start = GST_DEBUG_FUNCPTR (gst_flac_parse_start);
-  baseparse_class->stop = GST_DEBUG_FUNCPTR (gst_flac_parse_stop);
-  baseparse_class->check_valid_frame =
-      GST_DEBUG_FUNCPTR (gst_flac_parse_check_valid_frame);
-  baseparse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_flac_parse_parse_frame);
-  baseparse_class->pre_push_frame =
-      GST_DEBUG_FUNCPTR (gst_flac_parse_pre_push_frame);
-}
-
-static void
-gst_flac_parse_init (GstFlacParse * flacparse, GstFlacParseClass * klass)
-{
-  flacparse->check_frame_checksums = DEFAULT_CHECK_FRAME_CHECKSUMS;
-}
-
-static void
-gst_flac_parse_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (object);
-
-  switch (prop_id) {
-    case PROP_CHECK_FRAME_CHECKSUMS:
-      flacparse->check_frame_checksums = g_value_get_boolean (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_flac_parse_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (object);
-
-  switch (prop_id) {
-    case PROP_CHECK_FRAME_CHECKSUMS:
-      g_value_set_boolean (value, flacparse->check_frame_checksums);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_flac_parse_finalize (GObject * object)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (object);
-
-  if (flacparse->tags) {
-    gst_tag_list_free (flacparse->tags);
-    flacparse->tags = NULL;
-  }
-
-  g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL);
-  g_list_free (flacparse->headers);
-  flacparse->headers = NULL;
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_flac_parse_start (GstBaseParse * parse)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (parse);
-
-  flacparse->state = GST_FLAC_PARSE_STATE_INIT;
-  flacparse->min_blocksize = 0;
-  flacparse->max_blocksize = 0;
-  flacparse->min_framesize = 0;
-  flacparse->max_framesize = 0;
-
-  flacparse->upstream_length = -1;
-
-  flacparse->samplerate = 0;
-  flacparse->channels = 0;
-  flacparse->bps = 0;
-  flacparse->total_samples = 0;
-
-  flacparse->offset = GST_CLOCK_TIME_NONE;
-  flacparse->blocking_strategy = 0;
-  flacparse->block_size = 0;
-  flacparse->sample_number = 0;
-
-  /* "fLaC" marker */
-  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), 4);
-  /* inform baseclass we can come up with ts, based on counters in packets */
-  gst_base_parse_set_format (GST_BASE_PARSE (flacparse),
-      GST_BASE_PARSE_FORMAT_HAS_TIME, TRUE);
-
-  return TRUE;
-}
-
-static gboolean
-gst_flac_parse_stop (GstBaseParse * parse)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (parse);
-
-  if (flacparse->tags) {
-    gst_tag_list_free (flacparse->tags);
-    flacparse->tags = NULL;
-  }
-
-  g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL);
-  g_list_free (flacparse->headers);
-  flacparse->headers = NULL;
-
-  return TRUE;
-}
-
-static const guint8 sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };
-
-static const guint16 blocksize_table[16] = {
-  0, 192, 576 << 0, 576 << 1, 576 << 2, 576 << 3, 0, 0,
-  256 << 0, 256 << 1, 256 << 2, 256 << 3, 256 << 4, 256 << 5, 256 << 6,
-  256 << 7,
-};
-
-static const guint32 sample_rate_table[16] = {
-  0,
-  88200, 176400, 192000,
-  8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
-  0, 0, 0, 0,
-};
-
-typedef enum
-{
-  FRAME_HEADER_VALID,
-  FRAME_HEADER_INVALID,
-  FRAME_HEADER_MORE_DATA
-} FrameHeaderCheckReturn;
-
-static FrameHeaderCheckReturn
-gst_flac_parse_frame_header_is_valid (GstFlacParse * flacparse,
-    const guint8 * data, guint size, gboolean set, guint16 * block_size_ret)
-{
-  GstBitReader reader = GST_BIT_READER_INIT (data, size);
-  guint8 blocking_strategy;
-  guint16 block_size;
-  guint32 samplerate = 0;
-  guint64 sample_number;
-  guint8 channels, bps;
-  guint8 tmp = 0;
-  guint8 actual_crc, expected_crc = 0;
-
-  /* Skip 14 bit sync code */
-  gst_bit_reader_skip_unchecked (&reader, 14);
-
-  /* Must be 0 */
-  if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) != 0)
-    goto error;
-
-  /* 0 == fixed block size, 1 == variable block size */
-  blocking_strategy = gst_bit_reader_get_bits_uint8_unchecked (&reader, 1);
-
-  /* block size index, calculation of the real blocksize below */
-  block_size = gst_bit_reader_get_bits_uint16_unchecked (&reader, 4);
-  if (block_size == 0)
-    goto error;
-
-  /* sample rate index, calculation of the real samplerate below */
-  samplerate = gst_bit_reader_get_bits_uint16_unchecked (&reader, 4);
-  if (samplerate == 0x0f)
-    goto error;
-
-  /* channel assignment */
-  channels = gst_bit_reader_get_bits_uint8_unchecked (&reader, 4);
-  if (channels < 8) {
-    channels++;
-  } else if (channels <= 10) {
-    channels = 2;
-  } else if (channels > 10) {
-    goto error;
-  }
-  if (flacparse->channels && flacparse->channels != channels)
-    goto error;
-
-  /* bits per sample */
-  bps = gst_bit_reader_get_bits_uint8_unchecked (&reader, 3);
-  if (bps == 0x03 || bps == 0x07) {
-    goto error;
-  } else if (bps == 0 && flacparse->bps == 0) {
-    goto need_streaminfo;
-  }
-  bps = sample_size_table[bps];
-  if (flacparse->bps && bps != flacparse->bps)
-    goto error;
-
-  /* reserved, must be 0 */
-  if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) != 0)
-    goto error;
-
-  /* read "utf8" encoded sample/frame number */
-  {
-    gint len = 0;
-
-    len = gst_bit_reader_get_bits_uint8_unchecked (&reader, 8);
-
-    /* This is slightly faster than a loop */
-    if (!(len & 0x80)) {
-      sample_number = len;
-      len = 0;
-    } else if ((len & 0xc0) && !(len & 0x20)) {
-      sample_number = len & 0x1f;
-      len = 1;
-    } else if ((len & 0xe0) && !(len & 0x10)) {
-      sample_number = len & 0x0f;
-      len = 2;
-    } else if ((len & 0xf0) && !(len & 0x08)) {
-      sample_number = len & 0x07;
-      len = 3;
-    } else if ((len & 0xf8) && !(len & 0x04)) {
-      sample_number = len & 0x03;
-      len = 4;
-    } else if ((len & 0xfc) && !(len & 0x02)) {
-      sample_number = len & 0x01;
-      len = 5;
-    } else if ((len & 0xfe) && !(len & 0x01)) {
-      sample_number = len & 0x0;
-      len = 6;
-    } else {
-      goto error;
-    }
-
-    if ((blocking_strategy == 0 && len > 5) ||
-        (blocking_strategy == 1 && len > 6))
-      goto error;
-
-    while (len > 0) {
-      if (!gst_bit_reader_get_bits_uint8 (&reader, &tmp, 8))
-        goto need_more_data;
-
-      if ((tmp & 0xc0) != 0x80)
-        goto error;
-
-      sample_number <<= 6;
-      sample_number |= (tmp & 0x3f);
-      len--;
-    }
-  }
-
-  /* calculate real blocksize from the blocksize index */
-  if (block_size == 0) {
-    goto error;
-  } else if (block_size == 6) {
-    if (!gst_bit_reader_get_bits_uint16 (&reader, &block_size, 8))
-      goto need_more_data;
-    block_size++;
-  } else if (block_size == 7) {
-    if (!gst_bit_reader_get_bits_uint16 (&reader, &block_size, 16))
-      goto need_more_data;
-    block_size++;
-  } else {
-    block_size = blocksize_table[block_size];
-  }
-
-  /* calculate the real samplerate from the samplerate index */
-  if (samplerate == 0 && flacparse->samplerate == 0) {
-    goto need_streaminfo;
-  } else if (samplerate < 12) {
-    samplerate = sample_rate_table[samplerate];
-  } else if (samplerate == 12) {
-    if (!gst_bit_reader_get_bits_uint32 (&reader, &samplerate, 8))
-      goto need_more_data;
-    samplerate *= 1000;
-  } else if (samplerate == 13) {
-    if (!gst_bit_reader_get_bits_uint32 (&reader, &samplerate, 16))
-      goto need_more_data;
-  } else if (samplerate == 14) {
-    if (!gst_bit_reader_get_bits_uint32 (&reader, &samplerate, 16))
-      goto need_more_data;
-    samplerate *= 10;
-  }
-
-  if (flacparse->samplerate && flacparse->samplerate != samplerate)
-    goto error;
-
-  /* check crc-8 for the header */
-  if (!gst_bit_reader_get_bits_uint8 (&reader, &expected_crc, 8))
-    goto need_more_data;
-
-  actual_crc =
-      gst_flac_calculate_crc8 (data,
-      (gst_bit_reader_get_pos (&reader) / 8) - 1);
-  if (actual_crc != expected_crc)
-    goto error;
-
-  if (set) {
-    flacparse->block_size = block_size;
-    if (!flacparse->samplerate)
-      flacparse->samplerate = samplerate;
-    if (!flacparse->bps)
-      flacparse->bps = bps;
-    if (!flacparse->blocking_strategy)
-      flacparse->blocking_strategy = blocking_strategy;
-    if (!flacparse->channels)
-      flacparse->channels = channels;
-    if (!flacparse->sample_number)
-      flacparse->sample_number = sample_number;
-
-    GST_DEBUG_OBJECT (flacparse,
-        "Parsed frame at offset %" G_GUINT64_FORMAT ":\n" "Block size: %u\n"
-        "Sample/Frame number: %" G_GUINT64_FORMAT, flacparse->offset,
-        flacparse->block_size, flacparse->sample_number);
-  }
-
-  if (block_size_ret)
-    *block_size_ret = block_size;
-
-  return FRAME_HEADER_VALID;
-
-need_streaminfo:
-  GST_ERROR_OBJECT (flacparse, "Need STREAMINFO");
-  return FRAME_HEADER_INVALID;
-error:
-  return FRAME_HEADER_INVALID;
-
-need_more_data:
-  return FRAME_HEADER_MORE_DATA;
-}
-
-static gboolean
-gst_flac_parse_frame_is_valid (GstFlacParse * flacparse,
-    GstBaseParseFrame * frame, guint * ret)
-{
-  GstBuffer *buffer;
-  const guint8 *data;
-  guint max, size, remaining;
-  guint i, search_start, search_end;
-  FrameHeaderCheckReturn header_ret;
-  guint16 block_size;
-
-  buffer = frame->buffer;
-  data = GST_BUFFER_DATA (buffer);
-  size = GST_BUFFER_SIZE (buffer);
-
-  if (size <= flacparse->min_framesize)
-    goto need_more;
-
-  header_ret =
-      gst_flac_parse_frame_header_is_valid (flacparse, data, size, TRUE,
-      &block_size);
-  if (header_ret == FRAME_HEADER_INVALID) {
-    *ret = 0;
-    return FALSE;
-  } else if (header_ret == FRAME_HEADER_MORE_DATA) {
-    goto need_more;
-  }
-
-  /* mind unknown framesize */
-  search_start = MAX (2, flacparse->min_framesize);
-  if (flacparse->max_framesize)
-    search_end = MIN (size, flacparse->max_framesize + 9 + 2);
-  else
-    search_end = size;
-  search_end -= 2;
-
-  remaining = size;
-
-  for (i = search_start; i < search_end; i++, remaining--) {
-    if ((GST_READ_UINT16_BE (data + i) & 0xfffe) == 0xfff8) {
-      header_ret =
-          gst_flac_parse_frame_header_is_valid (flacparse, data + i, remaining,
-          FALSE, NULL);
-      if (header_ret == FRAME_HEADER_VALID) {
-        if (flacparse->check_frame_checksums) {
-          guint16 actual_crc = gst_flac_calculate_crc16 (data, i - 2);
-          guint16 expected_crc = GST_READ_UINT16_BE (data + i - 2);
-
-          if (actual_crc != expected_crc)
-            continue;
-        }
-        *ret = i;
-        flacparse->block_size = block_size;
-        return TRUE;
-      } else if (header_ret == FRAME_HEADER_MORE_DATA) {
-        goto need_more;
-      }
-    }
-  }
-
-  /* For the last frame output everything to the end */
-  if (G_UNLIKELY (GST_BASE_PARSE_FRAME_DRAIN (frame))) {
-    if (flacparse->check_frame_checksums) {
-      guint16 actual_crc = gst_flac_calculate_crc16 (data, size - 2);
-      guint16 expected_crc = GST_READ_UINT16_BE (data + size - 2);
-
-      if (actual_crc == expected_crc) {
-        *ret = size;
-        flacparse->block_size = block_size;
-        return TRUE;
-      }
-    } else {
-      *ret = size;
-      flacparse->block_size = block_size;
-      return TRUE;
-    }
-  }
-
-need_more:
-  max = flacparse->max_framesize + 16;
-  if (max == 16)
-    max = 1 << 24;
-  *ret = MIN (size + 4096, max);
-  return FALSE;
-}
-
-static gboolean
-gst_flac_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (parse);
-  GstBuffer *buffer = frame->buffer;
-  const guint8 *data = GST_BUFFER_DATA (buffer);
-
-  if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 4))
-    return FALSE;
-
-  if (flacparse->state == GST_FLAC_PARSE_STATE_INIT) {
-    if (memcmp (GST_BUFFER_DATA (buffer), "fLaC", 4) == 0) {
-      GST_DEBUG_OBJECT (flacparse, "fLaC marker found");
-      *framesize = 4;
-      return TRUE;
-    } else if (data[0] == 0xff && (data[1] >> 2) == 0x3e) {
-      GST_DEBUG_OBJECT (flacparse, "Found headerless FLAC");
-      /* Minimal size of a frame header */
-      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), 9);
-      flacparse->state = GST_FLAC_PARSE_STATE_GENERATE_HEADERS;
-      *skipsize = 0;
-      return FALSE;
-    } else {
-      GST_DEBUG_OBJECT (flacparse, "fLaC marker not found");
-      return FALSE;
-    }
-  } else if (flacparse->state == GST_FLAC_PARSE_STATE_HEADERS) {
-    guint size = 4 + ((data[1] << 16) | (data[2] << 8) | (data[3]));
-
-    GST_DEBUG_OBJECT (flacparse, "Found metadata block of size %u", size);
-    *framesize = size;
-    return TRUE;
-  } else {
-    if ((GST_READ_UINT16_BE (data) & 0xfffe) == 0xfff8) {
-      gboolean ret;
-      guint next;
-
-      flacparse->offset = GST_BUFFER_OFFSET (buffer);
-      flacparse->blocking_strategy = 0;
-      flacparse->block_size = 0;
-      flacparse->sample_number = 0;
-
-      GST_DEBUG_OBJECT (flacparse, "Found sync code");
-      ret = gst_flac_parse_frame_is_valid (flacparse, frame, &next);
-      if (ret) {
-        *framesize = next;
-        return TRUE;
-      } else {
-        /* If we're at EOS and the frame was not valid, drop it! */
-        if (G_UNLIKELY (GST_BASE_PARSE_FRAME_DRAIN (frame))) {
-          GST_WARNING_OBJECT (flacparse, "EOS");
-          return FALSE;
-        }
-
-        if (next == 0) {
-        } else if (next > GST_BUFFER_SIZE (buffer)) {
-          GST_DEBUG_OBJECT (flacparse, "Requesting %u bytes", next);
-          *skipsize = 0;
-          gst_base_parse_set_min_frame_size (parse, next);
-          return FALSE;
-        } else {
-          GST_ERROR_OBJECT (flacparse,
-              "Giving up on invalid frame (%d bytes)",
-              GST_BUFFER_SIZE (buffer));
-          return FALSE;
-        }
-      }
-    } else {
-      GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buffer);
-      gint off;
-
-      off =
-          gst_byte_reader_masked_scan_uint32 (&reader, 0xfffc0000, 0xfff80000,
-          0, GST_BUFFER_SIZE (buffer));
-
-      if (off > 0) {
-        GST_DEBUG_OBJECT (parse, "Possible sync at buffer offset %d", off);
-        *skipsize = off;
-        return FALSE;
-      } else {
-        GST_DEBUG_OBJECT (flacparse, "Sync code not found");
-        *skipsize = GST_BUFFER_SIZE (buffer) - 3;
-        return FALSE;
-      }
-    }
-  }
-
-  return FALSE;
-}
-
-static gboolean
-gst_flac_parse_handle_streaminfo (GstFlacParse * flacparse, GstBuffer * buffer)
-{
-  GstBitReader reader = GST_BIT_READER_INIT_FROM_BUFFER (buffer);
-
-  if (GST_BUFFER_SIZE (buffer) != 4 + 34) {
-    GST_ERROR_OBJECT (flacparse, "Invalid metablock size for STREAMINFO: %u",
-        GST_BUFFER_SIZE (buffer));
-    return FALSE;
-  }
-
-  /* Skip metadata block header */
-  gst_bit_reader_skip (&reader, 32);
-
-  if (!gst_bit_reader_get_bits_uint16 (&reader, &flacparse->min_blocksize, 16))
-    goto error;
-  if (flacparse->min_blocksize < 16) {
-    GST_ERROR_OBJECT (flacparse, "Invalid minimum block size: %u",
-        flacparse->min_blocksize);
-    return FALSE;
-  }
-
-  if (!gst_bit_reader_get_bits_uint16 (&reader, &flacparse->max_blocksize, 16))
-    goto error;
-  if (flacparse->max_blocksize < 16) {
-    GST_ERROR_OBJECT (flacparse, "Invalid maximum block size: %u",
-        flacparse->max_blocksize);
-    return FALSE;
-  }
-
-  if (!gst_bit_reader_get_bits_uint32 (&reader, &flacparse->min_framesize, 24))
-    goto error;
-  if (!gst_bit_reader_get_bits_uint32 (&reader, &flacparse->max_framesize, 24))
-    goto error;
-
-  if (!gst_bit_reader_get_bits_uint32 (&reader, &flacparse->samplerate, 20))
-    goto error;
-  if (flacparse->samplerate == 0) {
-    GST_ERROR_OBJECT (flacparse, "Invalid sample rate 0");
-    return FALSE;
-  }
-
-  if (!gst_bit_reader_get_bits_uint8 (&reader, &flacparse->channels, 3))
-    goto error;
-  flacparse->channels++;
-  if (flacparse->channels > 8) {
-    GST_ERROR_OBJECT (flacparse, "Invalid number of channels %u",
-        flacparse->channels);
-    return FALSE;
-  }
-
-  if (!gst_bit_reader_get_bits_uint8 (&reader, &flacparse->bps, 5))
-    goto error;
-  flacparse->bps++;
-
-  if (!gst_bit_reader_get_bits_uint64 (&reader, &flacparse->total_samples, 36))
-    goto error;
-  if (flacparse->total_samples)
-    gst_base_parse_set_duration (GST_BASE_PARSE (flacparse), GST_FORMAT_TIME,
-        GST_FRAMES_TO_CLOCK_TIME (flacparse->total_samples,
-            flacparse->samplerate), 0);
-
-  GST_DEBUG_OBJECT (flacparse, "STREAMINFO:\n"
-      "\tmin/max blocksize: %u/%u,\n"
-      "\tmin/max framesize: %u/%u,\n"
-      "\tsamplerate: %u,\n"
-      "\tchannels: %u,\n"
-      "\tbits per sample: %u,\n"
-      "\ttotal samples: %" G_GUINT64_FORMAT,
-      flacparse->min_blocksize, flacparse->max_blocksize,
-      flacparse->min_framesize, flacparse->max_framesize,
-      flacparse->samplerate,
-      flacparse->channels, flacparse->bps, flacparse->total_samples);
-
-  return TRUE;
-
-error:
-  GST_ERROR_OBJECT (flacparse, "Failed to read data");
-  return FALSE;
-}
-
-static gboolean
-gst_flac_parse_handle_vorbiscomment (GstFlacParse * flacparse,
-    GstBuffer * buffer)
-{
-  flacparse->tags = gst_tag_list_from_vorbiscomment_buffer (buffer,
-      GST_BUFFER_DATA (buffer), 4, NULL);
-
-  if (flacparse->tags == NULL) {
-    GST_ERROR_OBJECT (flacparse, "Invalid vorbiscomment block");
-  } else if (gst_tag_list_is_empty (flacparse->tags)) {
-    gst_tag_list_free (flacparse->tags);
-    flacparse->tags = NULL;
-  }
-
-  return TRUE;
-}
-
-static gboolean
-gst_flac_parse_handle_picture (GstFlacParse * flacparse, GstBuffer * buffer)
-{
-  GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buffer);
-  const guint8 *data = GST_BUFFER_DATA (buffer);
-  guint32 img_len = 0, img_type = 0;
-  guint32 img_mimetype_len = 0, img_description_len = 0;
-
-  if (!gst_byte_reader_skip (&reader, 4))
-    goto error;
-
-  if (!gst_byte_reader_get_uint32_be (&reader, &img_type))
-    goto error;
-
-  if (!gst_byte_reader_get_uint32_be (&reader, &img_mimetype_len))
-    goto error;
-  if (!gst_byte_reader_skip (&reader, img_mimetype_len))
-    goto error;
-
-  if (!gst_byte_reader_get_uint32_be (&reader, &img_description_len))
-    goto error;
-  if (!gst_byte_reader_skip (&reader, img_description_len))
-    goto error;
-
-  if (!gst_byte_reader_skip (&reader, 4 * 4))
-    goto error;
-
-  if (!gst_byte_reader_get_uint32_be (&reader, &img_len))
-    goto error;
-
-  if (!flacparse->tags)
-    flacparse->tags = gst_tag_list_new ();
-
-  gst_tag_list_add_id3_image (flacparse->tags,
-      data + gst_byte_reader_get_pos (&reader), img_len, img_type);
-
-  if (gst_tag_list_is_empty (flacparse->tags)) {
-    gst_tag_list_free (flacparse->tags);
-    flacparse->tags = NULL;
-  }
-
-  return TRUE;
-
-error:
-  GST_ERROR_OBJECT (flacparse, "Error reading data");
-  return FALSE;
-}
-
-static gboolean
-gst_flac_parse_handle_seektable (GstFlacParse * flacparse, GstBuffer * buffer)
-{
-
-  GST_DEBUG_OBJECT (flacparse, "storing seektable");
-  /* only store for now;
-   * offset of the first frame is needed to get real info */
-  flacparse->seektable = gst_buffer_ref (buffer);
-
-  return TRUE;
-}
-
-static void
-gst_flac_parse_process_seektable (GstFlacParse * flacparse, gint64 boffset)
-{
-  GstByteReader br;
-  gint64 offset = 0, samples = 0;
-
-  GST_DEBUG_OBJECT (flacparse,
-      "parsing seektable; base offset %" G_GINT64_FORMAT, boffset);
-
-  if (boffset <= 0)
-    goto done;
-
-  gst_byte_reader_init_from_buffer (&br, flacparse->seektable);
-  /* skip header */
-  if (!gst_byte_reader_skip (&br, 4))
-    goto done;
-
-  /* seekpoints */
-  while (gst_byte_reader_get_remaining (&br)) {
-    if (!gst_byte_reader_get_int64_be (&br, &samples))
-      break;
-    if (!gst_byte_reader_get_int64_be (&br, &offset))
-      break;
-    if (!gst_byte_reader_skip (&br, 2))
-      break;
-
-    GST_LOG_OBJECT (flacparse, "samples %" G_GINT64_FORMAT " -> offset %"
-        G_GINT64_FORMAT, samples, offset);
-
-    /* sanity check */
-    if (G_LIKELY (offset > 0 && samples > 0)) {
-      gst_base_parse_add_index_entry (GST_BASE_PARSE (flacparse),
-          boffset + offset, gst_util_uint64_scale (samples, GST_SECOND,
-              flacparse->samplerate), TRUE, FALSE);
-    }
-  }
-
-done:
-  gst_buffer_unref (flacparse->seektable);
-  flacparse->seektable = NULL;
-}
-
-static void
-_value_array_append_buffer (GValue * array_val, GstBuffer * buf)
-{
-  GValue value = { 0, };
-
-  g_value_init (&value, GST_TYPE_BUFFER);
-  /* copy buffer to avoid problems with circular refcounts */
-  buf = gst_buffer_copy (buf);
-  /* again, for good measure */
-  GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
-  gst_value_set_buffer (&value, buf);
-  gst_buffer_unref (buf);
-  gst_value_array_append_value (array_val, &value);
-  g_value_unset (&value);
-}
-
-static gboolean
-gst_flac_parse_handle_headers (GstFlacParse * flacparse)
-{
-  GstBuffer *vorbiscomment = NULL;
-  GstBuffer *streaminfo = NULL;
-  GstBuffer *marker = NULL;
-  GValue array = { 0, };
-  GstCaps *caps;
-  GList *l;
-  gboolean res = TRUE;
-
-  caps = gst_caps_new_simple ("audio/x-flac",
-      "channels", G_TYPE_INT, flacparse->channels,
-      "framed", G_TYPE_BOOLEAN, TRUE,
-      "rate", G_TYPE_INT, flacparse->samplerate, NULL);
-
-  if (!flacparse->headers)
-    goto push_headers;
-
-  for (l = flacparse->headers; l; l = l->next) {
-    GstBuffer *header = l->data;
-    const guint8 *data = GST_BUFFER_DATA (header);
-    guint size = GST_BUFFER_SIZE (header);
-
-    GST_BUFFER_FLAG_SET (header, GST_BUFFER_FLAG_IN_CAPS);
-
-    if (size == 4 && memcmp (data, "fLaC", 4) == 0) {
-      marker = header;
-    } else if (size > 1 && (data[0] & 0x7f) == 0) {
-      streaminfo = header;
-    } else if (size > 1 && (data[0] & 0x7f) == 4) {
-      vorbiscomment = header;
-    }
-  }
-
-  if (marker == NULL || streaminfo == NULL || vorbiscomment == NULL) {
-    GST_WARNING_OBJECT (flacparse,
-        "missing header %p %p %p, muxing into container "
-        "formats may be broken", marker, streaminfo, vorbiscomment);
-    goto push_headers;
-  }
-
-  g_value_init (&array, GST_TYPE_ARRAY);
-
-  /* add marker including STREAMINFO header */
-  {
-    GstBuffer *buf;
-    guint16 num;
-
-    /* minus one for the marker that is merged with streaminfo here */
-    num = g_list_length (flacparse->headers) - 1;
-
-    buf = gst_buffer_new_and_alloc (13 + GST_BUFFER_SIZE (streaminfo));
-    GST_BUFFER_DATA (buf)[0] = 0x7f;
-    memcpy (GST_BUFFER_DATA (buf) + 1, "FLAC", 4);
-    GST_BUFFER_DATA (buf)[5] = 0x01;    /* mapping version major */
-    GST_BUFFER_DATA (buf)[6] = 0x00;    /* mapping version minor */
-    GST_BUFFER_DATA (buf)[7] = (num & 0xFF00) >> 8;
-    GST_BUFFER_DATA (buf)[8] = (num & 0x00FF) >> 0;
-    memcpy (GST_BUFFER_DATA (buf) + 9, "fLaC", 4);
-    memcpy (GST_BUFFER_DATA (buf) + 13, GST_BUFFER_DATA (streaminfo),
-        GST_BUFFER_SIZE (streaminfo));
-    _value_array_append_buffer (&array, buf);
-    gst_buffer_unref (buf);
-  }
-
-  /* add VORBISCOMMENT header */
-  _value_array_append_buffer (&array, vorbiscomment);
-
-  /* add other headers, if there are any */
-  for (l = flacparse->headers; l; l = l->next) {
-    if (GST_BUFFER_CAST (l->data) != marker &&
-        GST_BUFFER_CAST (l->data) != streaminfo &&
-        GST_BUFFER_CAST (l->data) != vorbiscomment) {
-      _value_array_append_buffer (&array, GST_BUFFER_CAST (l->data));
-    }
-  }
-
-  gst_structure_set_value (gst_caps_get_structure (caps, 0),
-      "streamheader", &array);
-  g_value_unset (&array);
-
-push_headers:
-
-  gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (GST_BASE_PARSE (flacparse)), caps);
-  gst_caps_unref (caps);
-
-  /* push header buffers; update caps, so when we push the first buffer the
-   * negotiated caps will change to caps that include the streamheader field */
-  while (flacparse->headers) {
-    GstBuffer *buf = GST_BUFFER (flacparse->headers->data);
-    GstFlowReturn ret;
-    GstBaseParseFrame frame;
-
-    flacparse->headers =
-        g_list_delete_link (flacparse->headers, flacparse->headers);
-    buf = gst_buffer_make_metadata_writable (buf);
-    gst_buffer_set_caps (buf,
-        GST_PAD_CAPS (GST_BASE_PARSE_SRC_PAD (GST_BASE_PARSE (flacparse))));
-
-    /* init, set and give away frame */
-    gst_base_parse_frame_init (GST_BASE_PARSE (flacparse), &frame);
-    frame.buffer = buf;
-    frame.overhead = -1;
-    ret = gst_base_parse_push_frame (GST_BASE_PARSE (flacparse), &frame);
-    if (ret != GST_FLOW_OK) {
-      res = FALSE;
-      break;
-    }
-  }
-  g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL);
-  g_list_free (flacparse->headers);
-  flacparse->headers = NULL;
-
-  return res;
-}
-
-static gboolean
-gst_flac_parse_generate_headers (GstFlacParse * flacparse)
-{
-  GstBuffer *marker, *streaminfo, *vorbiscomment;
-  guint8 *data;
-
-  marker = gst_buffer_new_and_alloc (4);
-  memcpy (GST_BUFFER_DATA (marker), "fLaC", 4);
-  GST_BUFFER_TIMESTAMP (marker) = GST_CLOCK_TIME_NONE;
-  GST_BUFFER_DURATION (marker) = GST_CLOCK_TIME_NONE;
-  GST_BUFFER_OFFSET (marker) = 0;
-  GST_BUFFER_OFFSET_END (marker) = 0;
-  flacparse->headers = g_list_append (flacparse->headers, marker);
-
-  streaminfo = gst_buffer_new_and_alloc (4 + 34);
-  data = GST_BUFFER_DATA (streaminfo);
-  memset (data, 0, 4 + 34);
-
-  /* metadata block header */
-  data[0] = 0x00;               /* is_last = 0; type = 0; */
-  data[1] = 0x00;               /* length = 34; */
-  data[2] = 0x00;
-  data[3] = 0x22;
-
-  /* streaminfo */
-
-  data[4] = (flacparse->block_size >> 8) & 0xff;        /* min blocksize = blocksize; */
-  data[5] = (flacparse->block_size) & 0xff;
-  data[6] = (flacparse->block_size >> 8) & 0xff;        /* max blocksize = blocksize; */
-  data[7] = (flacparse->block_size) & 0xff;
-
-  data[8] = 0x00;               /* min framesize = 0; */
-  data[9] = 0x00;
-  data[10] = 0x00;
-  data[11] = 0x00;              /* max framesize = 0; */
-  data[12] = 0x00;
-  data[13] = 0x00;
-
-  data[14] = (flacparse->samplerate >> 12) & 0xff;
-  data[15] = (flacparse->samplerate >> 4) & 0xff;
-  data[16] = (flacparse->samplerate >> 0) & 0xf0;
-
-  data[16] |= (flacparse->channels - 1) << 1;
-
-  data[16] |= ((flacparse->bps - 1) >> 4) & 0x01;
-  data[17] = (((flacparse->bps - 1)) & 0x0f) << 4;
-
-  {
-    gint64 duration;
-    GstFormat fmt = GST_FORMAT_TIME;
-
-    if (gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE
-                (flacparse)), &fmt, &duration) && fmt == GST_FORMAT_TIME) {
-      duration = GST_CLOCK_TIME_TO_FRAMES (duration, flacparse->samplerate);
-
-      data[17] |= (duration >> 32) & 0xff;
-      data[18] |= (duration >> 24) & 0xff;
-      data[19] |= (duration >> 16) & 0xff;
-      data[20] |= (duration >> 8) & 0xff;
-      data[21] |= (duration >> 0) & 0xff;
-    }
-  }
-  /* MD5 = 0; */
-
-  GST_BUFFER_TIMESTAMP (streaminfo) = GST_CLOCK_TIME_NONE;
-  GST_BUFFER_DURATION (streaminfo) = GST_CLOCK_TIME_NONE;
-  GST_BUFFER_OFFSET (streaminfo) = 0;
-  GST_BUFFER_OFFSET_END (streaminfo) = 0;
-  flacparse->headers = g_list_append (flacparse->headers, streaminfo);
-
-  /* empty vorbiscomment */
-  {
-    GstTagList *taglist = gst_tag_list_new ();
-    guchar header[4];
-    guint size;
-
-    header[0] = 0x84;           /* is_last = 1; type = 4; */
-
-    vorbiscomment =
-        gst_tag_list_to_vorbiscomment_buffer (taglist, header,
-        sizeof (header), NULL);
-    gst_tag_list_free (taglist);
-
-    /* Get rid of framing bit */
-    if (GST_BUFFER_DATA (vorbiscomment)[GST_BUFFER_SIZE (vorbiscomment) -
-            1] == 1) {
-      GstBuffer *sub;
-
-      sub =
-          gst_buffer_create_sub (vorbiscomment, 0,
-          GST_BUFFER_SIZE (vorbiscomment) - 1);
-      gst_buffer_unref (vorbiscomment);
-      vorbiscomment = sub;
-    }
-
-    size = GST_BUFFER_SIZE (vorbiscomment) - 4;
-    GST_BUFFER_DATA (vorbiscomment)[1] = ((size & 0xFF0000) >> 16);
-    GST_BUFFER_DATA (vorbiscomment)[2] = ((size & 0x00FF00) >> 8);
-    GST_BUFFER_DATA (vorbiscomment)[3] = (size & 0x0000FF);
-
-    GST_BUFFER_TIMESTAMP (vorbiscomment) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_DURATION (vorbiscomment) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_OFFSET (vorbiscomment) = 0;
-    GST_BUFFER_OFFSET_END (vorbiscomment) = 0;
-    flacparse->headers = g_list_append (flacparse->headers, vorbiscomment);
-  }
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_flac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (parse);
-  GstBuffer *buffer = frame->buffer;
-  const guint8 *data = GST_BUFFER_DATA (buffer);
-
-  if (flacparse->state == GST_FLAC_PARSE_STATE_INIT) {
-    GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_OFFSET (buffer) = 0;
-    GST_BUFFER_OFFSET_END (buffer) = 0;
-
-    /* 32 bits metadata block */
-    gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), 4);
-    flacparse->state = GST_FLAC_PARSE_STATE_HEADERS;
-
-    flacparse->headers =
-        g_list_append (flacparse->headers, gst_buffer_ref (buffer));
-
-    return GST_BASE_PARSE_FLOW_DROPPED;
-  } else if (flacparse->state == GST_FLAC_PARSE_STATE_HEADERS) {
-    gboolean is_last = ((data[0] & 0x80) == 0x80);
-    guint type = (data[0] & 0x7F);
-
-    if (type == 127) {
-      GST_WARNING_OBJECT (flacparse, "Invalid metadata block type");
-      return GST_BASE_PARSE_FLOW_DROPPED;
-    }
-
-    GST_DEBUG_OBJECT (flacparse, "Handling metadata block of type %u", type);
-
-    switch (type) {
-      case 0:                  /* STREAMINFO */
-        if (!gst_flac_parse_handle_streaminfo (flacparse, buffer))
-          return GST_FLOW_ERROR;
-        break;
-      case 3:                  /* SEEKTABLE */
-        if (!gst_flac_parse_handle_seektable (flacparse, buffer))
-          return GST_FLOW_ERROR;
-        break;
-      case 4:                  /* VORBIS_COMMENT */
-        if (!gst_flac_parse_handle_vorbiscomment (flacparse, buffer))
-          return GST_FLOW_ERROR;
-        break;
-      case 6:                  /* PICTURE */
-        if (!gst_flac_parse_handle_picture (flacparse, buffer))
-          return GST_FLOW_ERROR;
-        break;
-      case 1:                  /* PADDING */
-      case 2:                  /* APPLICATION */
-      case 5:                  /* CUESHEET */
-      default:                 /* RESERVED */
-        break;
-    }
-
-    GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
-    GST_BUFFER_OFFSET (buffer) = 0;
-    GST_BUFFER_OFFSET_END (buffer) = 0;
-
-    flacparse->headers =
-        g_list_append (flacparse->headers, gst_buffer_ref (buffer));
-
-    if (is_last) {
-      if (!gst_flac_parse_handle_headers (flacparse))
-        return GST_FLOW_ERROR;
-
-      /* Minimal size of a frame header */
-      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), MAX (9,
-              flacparse->min_framesize));
-      flacparse->state = GST_FLAC_PARSE_STATE_DATA;
-    }
-
-    /* DROPPED because we pushed already or will push all headers manually */
-    return GST_BASE_PARSE_FLOW_DROPPED;
-  } else {
-    if (flacparse->offset != GST_BUFFER_OFFSET (buffer)) {
-      FrameHeaderCheckReturn ret;
-
-      flacparse->offset = GST_BUFFER_OFFSET (buffer);
-      ret =
-          gst_flac_parse_frame_header_is_valid (flacparse,
-          GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), TRUE, NULL);
-      if (ret != FRAME_HEADER_VALID) {
-        GST_ERROR_OBJECT (flacparse,
-            "Baseclass didn't provide a complete frame");
-        return GST_FLOW_ERROR;
-      }
-    }
-
-    if (flacparse->block_size == 0) {
-      GST_ERROR_OBJECT (flacparse, "Unparsed frame");
-      return GST_FLOW_ERROR;
-    }
-
-    if (flacparse->seektable)
-      gst_flac_parse_process_seektable (flacparse, GST_BUFFER_OFFSET (buffer));
-
-    if (flacparse->state == GST_FLAC_PARSE_STATE_GENERATE_HEADERS) {
-      if (flacparse->blocking_strategy == 1) {
-        GST_WARNING_OBJECT (flacparse,
-            "Generating headers for variable blocksize streams not supported");
-
-        if (!gst_flac_parse_handle_headers (flacparse))
-          return GST_FLOW_ERROR;
-      } else {
-        GST_DEBUG_OBJECT (flacparse, "Generating headers");
-
-        if (!gst_flac_parse_generate_headers (flacparse))
-          return GST_FLOW_ERROR;
-
-        if (!gst_flac_parse_handle_headers (flacparse))
-          return GST_FLOW_ERROR;
-      }
-      flacparse->state = GST_FLAC_PARSE_STATE_DATA;
-    }
-
-    /* also cater for oggmux metadata */
-    if (flacparse->blocking_strategy == 0) {
-      GST_BUFFER_TIMESTAMP (buffer) =
-          gst_util_uint64_scale (flacparse->sample_number,
-          flacparse->block_size * GST_SECOND, flacparse->samplerate);
-      GST_BUFFER_OFFSET_END (buffer) =
-          flacparse->sample_number * flacparse->block_size +
-          flacparse->block_size;
-    } else {
-      GST_BUFFER_TIMESTAMP (buffer) =
-          gst_util_uint64_scale (flacparse->sample_number, GST_SECOND,
-          flacparse->samplerate);
-      GST_BUFFER_OFFSET_END (buffer) =
-          flacparse->sample_number + flacparse->block_size;
-    }
-    GST_BUFFER_OFFSET (buffer) =
-        gst_util_uint64_scale (GST_BUFFER_OFFSET_END (buffer), GST_SECOND,
-        flacparse->samplerate);
-    GST_BUFFER_DURATION (buffer) =
-        GST_BUFFER_OFFSET (buffer) - GST_BUFFER_TIMESTAMP (buffer);
-
-    /* To simplify, we just assume that it's a fixed size header and ignore
-     * subframe headers. The first could lead us to being off by 88 bits and
-     * the second even less, so the total inaccuracy is negligible. */
-    frame->overhead = 7;
-
-    /* Minimal size of a frame header */
-    gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), MAX (9,
-            flacparse->min_framesize));
-
-    flacparse->offset = -1;
-    flacparse->blocking_strategy = 0;
-    flacparse->block_size = 0;
-    flacparse->sample_number = 0;
-    return GST_FLOW_OK;
-  }
-}
-
-static GstFlowReturn
-gst_flac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
-  GstFlacParse *flacparse = GST_FLAC_PARSE (parse);
-
-  /* Push tags */
-  if (flacparse->tags) {
-    gst_element_found_tags (GST_ELEMENT (flacparse), flacparse->tags);
-    flacparse->tags = NULL;
-  }
-
-  frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
-
-  return GST_FLOW_OK;
-}
diff --git a/gst/audioparsers/gstflacparse.h b/gst/audioparsers/gstflacparse.h
deleted file mode 100644
index 664b2a6bc..000000000
--- a/gst/audioparsers/gstflacparse.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/* GStreamer
- *
- * Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>.
- * Copyright (C) 2009 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * Copyright (C) 2009 Nokia Corporation. All rights reserved.
- *   Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * 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_FLAC_PARSE_H__
-#define __GST_FLAC_PARSE_H__
-
-#include <gst/gst.h>
-#include <gst/baseparse/gstbaseparse.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_FLAC_PARSE		   (gst_flac_parse_get_type())
-#define GST_FLAC_PARSE(obj)		   (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_FLAC_PARSE,GstFlacParse))
-#define GST_FLAC_PARSE_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_FLAC_PARSE,GstFlacParseClass))
-#define GST_FLAC_PARSE_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_FLAC_PARSE,GstFlacParseClass))
-#define GST_IS_FLAC_PARSE(obj)	   (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_FLAC_PARSE))
-#define GST_IS_FLAC_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_FLAC_PARSE))
-#define GST_FLAC_PARSE_CAST(obj)	((GstFlacParse *)(obj))
-
-typedef struct _GstFlacParse GstFlacParse;
-typedef struct _GstFlacParseClass GstFlacParseClass;
-
-typedef enum {
-  GST_FLAC_PARSE_STATE_INIT,
-  GST_FLAC_PARSE_STATE_HEADERS,
-  GST_FLAC_PARSE_STATE_GENERATE_HEADERS,
-  GST_FLAC_PARSE_STATE_DATA
-} GstFlacParseState;
-
-typedef struct {
-  guint8 type;
-} GstFlacParseSubFrame;
-
-struct _GstFlacParse {
-  GstBaseParse parent;
-
-  /* Properties */
-  gboolean check_frame_checksums;
-
-  GstFlacParseState state;
-
-  gint64 upstream_length;
-
-  /* STREAMINFO content */
-  guint16 min_blocksize, max_blocksize;
-  guint32 min_framesize, max_framesize;
-  guint32 samplerate;
-  guint8 channels;
-  guint8 bps;
-  guint64 total_samples;
-
-  /* Current frame */
-  guint64 offset;
-  guint8 blocking_strategy;
-  guint16 block_size;
-  guint64 sample_number;
-
-  GstTagList *tags;
-
-  GList *headers;
-  GstBuffer *seektable;
-};
-
-struct _GstFlacParseClass {
-  GstBaseParseClass parent_class;
-};
-
-GType gst_flac_parse_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_FLAC_PARSE_H__ */
diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c
deleted file mode 100644
index a9eabdcb3..000000000
--- a/gst/audioparsers/gstmpegaudioparse.c
+++ /dev/null
@@ -1,1252 +0,0 @@
-/* GStreamer MPEG audio parser
- * Copyright (C) 2006-2007 Jan Schmidt <thaytan@mad.scientist.com>
- * Copyright (C) 2010 Mark Nauwelaerts <mnauw users sf net>
- * Copyright (C) 2010 Nokia Corporation. All rights reserved.
- *   Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-/**
- * SECTION:element-mpegaudioparse
- * @short_description: MPEG audio parser
- * @see_also: #GstAmrParse, #GstAACParse
- *
- * Parses and frames mpeg1 audio streams. Provides seeking.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=test.mp3 ! mpegaudioparse ! mad ! autoaudiosink
- * ]|
- * </refsect2>
- */
-
-/* FIXME: we should make the base class (GstBaseParse) aware of the
- * XING seek table somehow, so it can use it properly for things like
- * accurate seeks. Currently it can only do a lookup via the convert function,
- * but then doesn't know what the result represents exactly. One could either
- * add a vfunc for index lookup, or just make mpegaudioparse populate the
- * base class's index via the API provided.
- */
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-
-#include "gstmpegaudioparse.h"
-#include <gst/base/gstbytereader.h>
-
-GST_DEBUG_CATEGORY_STATIC (mpeg_audio_parse_debug);
-#define GST_CAT_DEFAULT mpeg_audio_parse_debug
-
-#define MPEG_AUDIO_CHANNEL_MODE_UNKNOWN -1
-#define MPEG_AUDIO_CHANNEL_MODE_STEREO 0
-#define MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO 1
-#define MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL 2
-#define MPEG_AUDIO_CHANNEL_MODE_MONO 3
-
-#define CRC_UNKNOWN -1
-#define CRC_PROTECTED 0
-#define CRC_NOT_PROTECTED 1
-
-#define XING_FRAMES_FLAG     0x0001
-#define XING_BYTES_FLAG      0x0002
-#define XING_TOC_FLAG        0x0004
-#define XING_VBR_SCALE_FLAG  0x0008
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/mpeg, "
-        "mpegversion = (int) 1, "
-        "layer = (int) [ 1, 3 ], "
-        "rate = (int) [ 8000, 48000 ], channels = (int) [ 1, 2 ],"
-        "parsed=(boolean) true")
-    );
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false")
-    );
-
-static void gst_mpeg_audio_parse_finalize (GObject * object);
-
-static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse);
-static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse);
-static gboolean gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * size, gint * skipsize);
-static GstFlowReturn gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame);
-static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
-    GstFormat src_format, gint64 src_value,
-    GstFormat dest_format, gint64 * dest_value);
-
-GST_BOILERPLATE (GstMpegAudioParse, gst_mpeg_audio_parse, GstBaseParse,
-    GST_TYPE_BASE_PARSE);
-
-#define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE  \
-    (gst_mpeg_audio_channel_mode_get_type())
-
-static const GEnumValue mpeg_audio_channel_mode[] = {
-  {MPEG_AUDIO_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
-  {MPEG_AUDIO_CHANNEL_MODE_MONO, "Mono", "mono"},
-  {MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
-  {MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
-  {MPEG_AUDIO_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
-  {0, NULL, NULL},
-};
-
-static GType
-gst_mpeg_audio_channel_mode_get_type (void)
-{
-  static GType mpeg_audio_channel_mode_type = 0;
-
-  if (!mpeg_audio_channel_mode_type) {
-    mpeg_audio_channel_mode_type =
-        g_enum_register_static ("GstMpegAudioChannelMode",
-        mpeg_audio_channel_mode);
-  }
-  return mpeg_audio_channel_mode_type;
-}
-
-static const gchar *
-gst_mpeg_audio_channel_mode_get_nick (gint mode)
-{
-  guint i;
-  for (i = 0; i < G_N_ELEMENTS (mpeg_audio_channel_mode); i++) {
-    if (mpeg_audio_channel_mode[i].value == mode)
-      return mpeg_audio_channel_mode[i].value_nick;
-  }
-  return NULL;
-}
-
-static void
-gst_mpeg_audio_parse_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_template));
-
-  gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser",
-      "Codec/Parser/Audio",
-      "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
-      "Jan Schmidt <thaytan@mad.scientist.com>,"
-      "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
-}
-
-static void
-gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
-{
-  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
-  GObjectClass *object_class = G_OBJECT_CLASS (klass);
-
-  GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0,
-      "MPEG1 audio stream parser");
-
-  object_class->finalize = gst_mpeg_audio_parse_finalize;
-
-  parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start);
-  parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop);
-  parse_class->check_valid_frame =
-      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_check_valid_frame);
-  parse_class->parse_frame =
-      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_parse_frame);
-  parse_class->pre_push_frame =
-      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
-  parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
-
-  /* register tags */
-#define GST_TAG_CRC      "has-crc"
-#define GST_TAG_MODE     "channel-mode"
-
-  gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
-      "has crc", "Using CRC", NULL);
-  gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
-      "channel mode", "MPEG audio channel mode", NULL);
-
-  g_type_class_ref (GST_TYPE_MPEG_AUDIO_CHANNEL_MODE);
-}
-
-static void
-gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
-{
-  mp3parse->channels = -1;
-  mp3parse->rate = -1;
-  mp3parse->sent_codec_tag = FALSE;
-  mp3parse->last_posted_crc = CRC_UNKNOWN;
-  mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
-
-  mp3parse->hdr_bitrate = 0;
-
-  mp3parse->xing_flags = 0;
-  mp3parse->xing_bitrate = 0;
-  mp3parse->xing_frames = 0;
-  mp3parse->xing_total_time = 0;
-  mp3parse->xing_bytes = 0;
-  mp3parse->xing_vbr_scale = 0;
-  memset (mp3parse->xing_seek_table, 0, 100);
-  memset (mp3parse->xing_seek_table_inverse, 0, 256);
-
-  mp3parse->vbri_bitrate = 0;
-  mp3parse->vbri_frames = 0;
-  mp3parse->vbri_total_time = 0;
-  mp3parse->vbri_bytes = 0;
-  mp3parse->vbri_seek_points = 0;
-  g_free (mp3parse->vbri_seek_table);
-  mp3parse->vbri_seek_table = NULL;
-
-  mp3parse->encoder_delay = 0;
-  mp3parse->encoder_padding = 0;
-}
-
-static void
-gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse,
-    GstMpegAudioParseClass * klass)
-{
-  gst_mpeg_audio_parse_reset (mp3parse);
-}
-
-static void
-gst_mpeg_audio_parse_finalize (GObject * object)
-{
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_mpeg_audio_parse_start (GstBaseParse * parse)
-{
-  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
-
-  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (mp3parse), 1024);
-  GST_DEBUG_OBJECT (parse, "starting");
-
-  gst_mpeg_audio_parse_reset (mp3parse);
-
-  return TRUE;
-}
-
-static gboolean
-gst_mpeg_audio_parse_stop (GstBaseParse * parse)
-{
-  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
-
-  GST_DEBUG_OBJECT (parse, "stopping");
-
-  gst_mpeg_audio_parse_reset (mp3parse);
-
-  return TRUE;
-}
-
-static const guint mp3types_bitrates[2][3][16] = {
-  {
-        {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
-        {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
-        {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
-      },
-  {
-        {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
-        {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
-        {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
-      },
-};
-
-static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
-{22050, 24000, 16000},
-{11025, 12000, 8000}
-};
-
-static inline guint
-mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
-    guint * put_version, guint * put_layer, guint * put_channels,
-    guint * put_bitrate, guint * put_samplerate, guint * put_mode,
-    guint * put_crc)
-{
-  guint length;
-  gulong mode, samplerate, bitrate, layer, channels, padding, crc;
-  gulong version;
-  gint lsf, mpg25;
-
-  if (header & (1 << 20)) {
-    lsf = (header & (1 << 19)) ? 0 : 1;
-    mpg25 = 0;
-  } else {
-    lsf = 1;
-    mpg25 = 1;
-  }
-
-  version = 1 + lsf + mpg25;
-
-  layer = 4 - ((header >> 17) & 0x3);
-
-  crc = (header >> 16) & 0x1;
-
-  bitrate = (header >> 12) & 0xF;
-  bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
-  /* The caller has ensured we have a valid header, so bitrate can't be
-     zero here. */
-  g_assert (bitrate != 0);
-
-  samplerate = (header >> 10) & 0x3;
-  samplerate = mp3types_freqs[lsf + mpg25][samplerate];
-
-  padding = (header >> 9) & 0x1;
-
-  mode = (header >> 6) & 0x3;
-  channels = (mode == 3) ? 1 : 2;
-
-  switch (layer) {
-    case 1:
-      length = 4 * ((bitrate * 12) / samplerate + padding);
-      break;
-    case 2:
-      length = (bitrate * 144) / samplerate + padding;
-      break;
-    default:
-    case 3:
-      length = (bitrate * 144) / (samplerate << lsf) + padding;
-      break;
-  }
-
-  GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
-      length);
-  GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
-      "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
-      layer, channels, gst_mpeg_audio_channel_mode_get_nick (mode));
-
-  if (put_version)
-    *put_version = version;
-  if (put_layer)
-    *put_layer = layer;
-  if (put_channels)
-    *put_channels = channels;
-  if (put_bitrate)
-    *put_bitrate = bitrate;
-  if (put_samplerate)
-    *put_samplerate = samplerate;
-  if (put_mode)
-    *put_mode = mode;
-  if (put_crc)
-    *put_crc = crc;
-
-  return length;
-}
-
-/* Minimum number of consecutive, valid-looking frames to consider
- * for resyncing */
-#define MIN_RESYNC_FRAMES 3
-
-/* Perform extended validation to check that subsequent headers match
- * the first header given here in important characteristics, to avoid
- * false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive
- * frames to match their major characteristics.
- *
- * If at_eos is set to TRUE, we just check that we don't find any invalid
- * frames in whatever data is available, rather than requiring a full
- * MIN_RESYNC_FRAMES of data.
- *
- * Returns TRUE if we've seen enough data to validate or reject the frame.
- * If TRUE is returned, then *valid contains TRUE if it validated, or false
- * if we decided it was false sync.
- * If FALSE is returned, then *valid contains minimum needed data.
- */
-static gboolean
-gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
-    guint32 header, int bpf, gboolean at_eos, gint * valid)
-{
-  guint32 next_header;
-  const guint8 *data;
-  guint available;
-  int frames_found = 1;
-  int offset = bpf;
-
-  available = GST_BUFFER_SIZE (buf);
-  data = GST_BUFFER_DATA (buf);
-
-  while (frames_found < MIN_RESYNC_FRAMES) {
-    /* Check if we have enough data for all these frames, plus the next
-       frame header. */
-    if (available < offset + 4) {
-      if (at_eos) {
-        /* Running out of data at EOS is fine; just accept it */
-        *valid = TRUE;
-        return TRUE;
-      } else {
-        *valid = offset + 4;
-        return FALSE;
-      }
-    }
-
-    next_header = GST_READ_UINT32_BE (data + offset);
-    GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
-        offset, (unsigned int) header, (unsigned int) next_header, bpf);
-
-/* mask the bits which are allowed to differ between frames */
-#define HDRMASK ~((0xF << 12)  /* bitrate */ | \
-                  (0x1 <<  9)  /* padding */ | \
-                  (0xf <<  4)  /* mode|mode extension */ | \
-                  (0xf))        /* copyright|emphasis */
-
-    if ((next_header & HDRMASK) != (header & HDRMASK)) {
-      /* If any of the unmasked bits don't match, then it's not valid */
-      GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
-          "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)",
-          (guint) header, (guint) header & HDRMASK, (guint) next_header,
-          (guint) next_header & HDRMASK, bpf);
-      *valid = FALSE;
-      return TRUE;
-    } else if ((((next_header >> 12) & 0xf) == 0) ||
-        (((next_header >> 12) & 0xf) == 0xf)) {
-      /* The essential parts were the same, but the bitrate held an
-         invalid value - also reject */
-      GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
-      *valid = FALSE;
-      return TRUE;
-    }
-
-    bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
-        NULL, NULL, NULL, NULL, NULL, NULL, NULL);
-
-    offset += bpf;
-    frames_found++;
-  }
-
-  *valid = TRUE;
-  return TRUE;
-}
-
-static gboolean
-gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
-    unsigned long head)
-{
-  GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
-  /* if it's not a valid sync */
-  if ((head & 0xffe00000) != 0xffe00000) {
-    GST_WARNING_OBJECT (mp3parse, "invalid sync");
-    return FALSE;
-  }
-  /* if it's an invalid MPEG version */
-  if (((head >> 19) & 3) == 0x1) {
-    GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx",
-        (head >> 19) & 3);
-    return FALSE;
-  }
-  /* if it's an invalid layer */
-  if (!((head >> 17) & 3)) {
-    GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3);
-    return FALSE;
-  }
-  /* if it's an invalid bitrate */
-  if (((head >> 12) & 0xf) == 0x0) {
-    GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx."
-        "Free format files are not supported yet", (head >> 12) & 0xf);
-    return FALSE;
-  }
-  if (((head >> 12) & 0xf) == 0xf) {
-    GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
-    return FALSE;
-  }
-  /* if it's an invalid samplerate */
-  if (((head >> 10) & 0x3) == 0x3) {
-    GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx",
-        (head >> 10) & 0x3);
-    return FALSE;
-  }
-
-  if ((head & 0x3) == 0x2) {
-    /* Ignore this as there are some files with emphasis 0x2 that can
-     * be played fine. See BGO #537235 */
-    GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3);
-  }
-
-  return TRUE;
-}
-
-static gboolean
-gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
-{
-  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
-  GstBuffer *buf = frame->buffer;
-  GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
-  gint off, bpf;
-  gboolean sync, drain, valid, caps_change;
-  guint32 header;
-  guint bitrate, layer, rate, channels, version, mode, crc;
-
-  if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6))
-    return FALSE;
-
-  off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
-      0, GST_BUFFER_SIZE (buf));
-
-  GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
-
-  /* didn't find anything that looks like a sync word, skip */
-  if (off < 0) {
-    *skipsize = GST_BUFFER_SIZE (buf) - 3;
-    return FALSE;
-  }
-
-  /* possible frame header, but not at offset 0? skip bytes before sync */
-  if (off > 0) {
-    *skipsize = off;
-    return FALSE;
-  }
-
-  /* make sure the values in the frame header look sane */
-  header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf));
-  if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
-    *skipsize = 1;
-    return FALSE;
-  }
-
-  GST_LOG_OBJECT (parse, "got frame");
-
-  bpf = mp3_type_frame_length_from_header (mp3parse, header,
-      &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
-  g_assert (bpf != 0);
-
-  if (channels != mp3parse->channels || rate != mp3parse->rate ||
-      layer != mp3parse->layer || version != mp3parse->version)
-    caps_change = TRUE;
-  else
-    caps_change = FALSE;
-
-  sync = GST_BASE_PARSE_FRAME_SYNC (frame);
-  drain = GST_BASE_PARSE_FRAME_DRAIN (frame);
-
-  if (!drain && (!sync || caps_change)) {
-    if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, drain,
-            &valid)) {
-      /* not enough data */
-      gst_base_parse_set_min_frame_size (parse, valid);
-      *skipsize = 0;
-      return FALSE;
-    } else {
-      if (!valid) {
-        *skipsize = off + 2;
-        return FALSE;
-      }
-    }
-  } else if (drain && !sync && caps_change && mp3parse->rate > 0) {
-    /* avoid caps jitter that we can't be sure of */
-    *skipsize = off + 2;
-    return FALSE;
-  }
-
-  *framesize = bpf;
-  return TRUE;
-}
-
-static void
-gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse * mp3parse,
-    GstBuffer * buf)
-{
-  const guint32 xing_id = 0x58696e67;   /* 'Xing' in hex */
-  const guint32 info_id = 0x496e666f;   /* 'Info' in hex - found in LAME CBR files */
-  const guint32 vbri_id = 0x56425249;   /* 'VBRI' in hex */
-  const guint32 lame_id = 0x4c414d45;   /* 'LAME' in hex */
-  gint offset;
-  guint64 avail;
-  gint64 upstream_total_bytes = 0;
-  GstFormat fmt = GST_FORMAT_BYTES;
-  guint32 read_id;
-  const guint8 *data;
-  GstBaseParseSeekable seekable;
-  guint bitrate;
-
-  if (mp3parse->sent_codec_tag)
-    return;
-
-  /* Check first frame for Xing info */
-  if (mp3parse->version == 1) { /* MPEG-1 file */
-    if (mp3parse->channels == 1)
-      offset = 0x11;
-    else
-      offset = 0x20;
-  } else {                      /* MPEG-2 header */
-    if (mp3parse->channels == 1)
-      offset = 0x09;
-    else
-      offset = 0x11;
-  }
-  /* Skip the 4 bytes of the MP3 header too */
-  offset += 4;
-
-  /* Check if we have enough data to read the Xing header */
-  avail = GST_BUFFER_SIZE (buf);
-  data = GST_BUFFER_DATA (buf);
-  if (avail < offset + 8)
-    return;
-
-  /* The header starts at the provided offset */
-  data += offset;
-
-  /* obtain real upstream total bytes */
-  fmt = GST_FORMAT_BYTES;
-  if (!gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE
-              (mp3parse)), &fmt, &upstream_total_bytes))
-    upstream_total_bytes = 0;
-
-  read_id = GST_READ_UINT32_BE (data);
-  if (read_id == xing_id || read_id == info_id) {
-    guint32 xing_flags;
-    guint bytes_needed = offset + 8;
-    gint64 total_bytes;
-    GstClockTime total_time;
-
-    GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
-
-    /* Read 4 base bytes of flags, big-endian */
-    xing_flags = GST_READ_UINT32_BE (data + 4);
-    if (xing_flags & XING_FRAMES_FLAG)
-      bytes_needed += 4;
-    if (xing_flags & XING_BYTES_FLAG)
-      bytes_needed += 4;
-    if (xing_flags & XING_TOC_FLAG)
-      bytes_needed += 100;
-    if (xing_flags & XING_VBR_SCALE_FLAG)
-      bytes_needed += 4;
-    if (avail < bytes_needed) {
-      GST_DEBUG_OBJECT (mp3parse,
-          "Not enough data to read Xing header (need %d)", bytes_needed);
-      return;
-    }
-
-    GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
-    mp3parse->xing_flags = xing_flags;
-
-    data = GST_BUFFER_DATA (buf);
-    data += offset + 8;
-
-    if (xing_flags & XING_FRAMES_FLAG) {
-      mp3parse->xing_frames = GST_READ_UINT32_BE (data);
-      if (mp3parse->xing_frames == 0) {
-        GST_WARNING_OBJECT (mp3parse,
-            "Invalid number of frames in Xing header");
-        mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
-      } else {
-        mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
-            (guint64) (mp3parse->xing_frames) * (mp3parse->spf),
-            mp3parse->rate);
-      }
-
-      data += 4;
-    } else {
-      mp3parse->xing_frames = 0;
-      mp3parse->xing_total_time = 0;
-    }
-
-    if (xing_flags & XING_BYTES_FLAG) {
-      mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
-      if (mp3parse->xing_bytes == 0) {
-        GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header");
-        mp3parse->xing_flags &= ~XING_BYTES_FLAG;
-      }
-      data += 4;
-    } else {
-      mp3parse->xing_bytes = 0;
-    }
-
-    /* If we know the upstream size and duration, compute the
-     * total bitrate, rounded up to the nearest kbit/sec */
-    if ((total_time = mp3parse->xing_total_time) &&
-        (total_bytes = mp3parse->xing_bytes)) {
-      mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
-          8 * GST_SECOND, total_time);
-      mp3parse->xing_bitrate += 500;
-      mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
-    }
-
-    if (xing_flags & XING_TOC_FLAG) {
-      int i, percent = 0;
-      guchar *table = mp3parse->xing_seek_table;
-      guchar old = 0, new;
-      guint first;
-
-      first = data[0];
-      GST_DEBUG_OBJECT (mp3parse,
-          "Subtracting initial offset of %d bytes from Xing TOC", first);
-
-      /* xing seek table: percent time -> 1/256 bytepos */
-      for (i = 0; i < 100; i++) {
-        new = data[i] - first;
-        if (old > new) {
-          GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
-          mp3parse->xing_flags &= ~XING_TOC_FLAG;
-          goto skip_toc;
-        }
-        mp3parse->xing_seek_table[i] = old = new;
-      }
-
-      /* build inverse table: 1/256 bytepos -> 1/100 percent time */
-      for (i = 0; i < 256; i++) {
-        while (percent < 99 && table[percent + 1] <= i)
-          percent++;
-
-        if (table[percent] == i) {
-          mp3parse->xing_seek_table_inverse[i] = percent * 100;
-        } else if (table[percent] < i && percent < 99) {
-          gdouble fa, fb, fx;
-          gint a = percent, b = percent + 1;
-
-          fa = table[a];
-          fb = table[b];
-          fx = (b - a) / (fb - fa) * (i - fa) + a;
-          mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
-        } else if (percent == 99) {
-          gdouble fa, fb, fx;
-          gint a = percent, b = 100;
-
-          fa = table[a];
-          fb = 256.0;
-          fx = (b - a) / (fb - fa) * (i - fa) + a;
-          mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
-        }
-      }
-    skip_toc:
-      data += 100;
-    } else {
-      memset (mp3parse->xing_seek_table, 0, 100);
-      memset (mp3parse->xing_seek_table_inverse, 0, 256);
-    }
-
-    if (xing_flags & XING_VBR_SCALE_FLAG) {
-      mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
-      data += 4;
-    } else
-      mp3parse->xing_vbr_scale = 0;
-
-    GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
-        GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames,
-        GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes,
-        mp3parse->xing_vbr_scale);
-
-    /* check for truncated file */
-    if (upstream_total_bytes && mp3parse->xing_bytes &&
-        mp3parse->xing_bytes * 0.8 > upstream_total_bytes) {
-      GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
-          "invalidating Xing header duration and size");
-      mp3parse->xing_flags &= ~XING_BYTES_FLAG;
-      mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
-    }
-
-    /* Optional LAME tag? */
-    if (avail - bytes_needed >= 36 && GST_READ_UINT32_BE (data) == lame_id) {
-      gchar lame_version[10] = { 0, };
-      guint tag_rev;
-      guint32 encoder_delay, encoder_padding;
-
-      memcpy (lame_version, data, 9);
-      data += 9;
-      tag_rev = data[0] >> 4;
-      GST_DEBUG_OBJECT (mp3parse, "Found LAME tag revision %d created by '%s'",
-          tag_rev, lame_version);
-
-      /* Skip all the information we're not interested in */
-      data += 12;
-      /* Encoder delay and end padding */
-      encoder_delay = GST_READ_UINT24_BE (data);
-      encoder_delay >>= 12;
-      encoder_padding = GST_READ_UINT24_BE (data);
-      encoder_padding &= 0x000fff;
-
-      mp3parse->encoder_delay = encoder_delay;
-      mp3parse->encoder_padding = encoder_padding;
-
-      GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u",
-          encoder_delay, encoder_padding);
-    }
-  } else if (read_id == vbri_id) {
-    gint64 total_bytes, total_frames;
-    GstClockTime total_time;
-    guint16 nseek_points;
-
-    GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
-    if (avail < offset + 26) {
-      GST_DEBUG_OBJECT (mp3parse,
-          "Not enough data to read VBRI header (need %d)", offset + 26);
-      return;
-    }
-
-    GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
-    data = GST_BUFFER_DATA (buf);
-    data += offset + 4;
-
-    if (GST_READ_UINT16_BE (data) != 0x0001) {
-      GST_WARNING_OBJECT (mp3parse,
-          "Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
-      return;
-    }
-    data += 2;
-
-    /* Skip encoder delay */
-    data += 2;
-
-    /* Skip quality */
-    data += 2;
-
-    total_bytes = GST_READ_UINT32_BE (data);
-    if (total_bytes != 0)
-      mp3parse->vbri_bytes = total_bytes;
-    data += 4;
-
-    total_frames = GST_READ_UINT32_BE (data);
-    if (total_frames != 0) {
-      mp3parse->vbri_frames = total_frames;
-      mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND,
-          (guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate);
-    }
-    data += 4;
-
-    /* If we know the upstream size and duration, compute the
-     * total bitrate, rounded up to the nearest kbit/sec */
-    if ((total_time = mp3parse->vbri_total_time) &&
-        (total_bytes = mp3parse->vbri_bytes)) {
-      mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes,
-          8 * GST_SECOND, total_time);
-      mp3parse->vbri_bitrate += 500;
-      mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000;
-    }
-
-    nseek_points = GST_READ_UINT16_BE (data);
-    data += 2;
-
-    if (nseek_points > 0) {
-      guint scale, seek_bytes, seek_frames;
-      gint i;
-
-      mp3parse->vbri_seek_points = nseek_points;
-
-      scale = GST_READ_UINT16_BE (data);
-      data += 2;
-
-      seek_bytes = GST_READ_UINT16_BE (data);
-      data += 2;
-
-      seek_frames = GST_READ_UINT16_BE (data);
-
-      if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) {
-        GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table");
-        goto out_vbri;
-      }
-
-      if (avail < offset + 26 + nseek_points * seek_bytes) {
-        GST_WARNING_OBJECT (mp3parse,
-            "Not enough data to read VBRI seek table (need %d)",
-            offset + 26 + nseek_points * seek_bytes);
-        goto out_vbri;
-      }
-
-      if (seek_frames * nseek_points < total_frames - seek_frames ||
-          seek_frames * nseek_points > total_frames + seek_frames) {
-        GST_WARNING_OBJECT (mp3parse,
-            "VBRI seek table doesn't cover the complete file");
-        goto out_vbri;
-      }
-
-      if (avail < offset + 26) {
-        GST_DEBUG_OBJECT (mp3parse,
-            "Not enough data to read VBRI header (need %d)",
-            offset + 26 + nseek_points * seek_bytes);
-        return;
-      }
-
-      data = GST_BUFFER_DATA (buf);
-      data += offset + 26;
-
-      /* VBRI seek table: frame/seek_frames -> byte */
-      mp3parse->vbri_seek_table = g_new (guint32, nseek_points);
-      if (seek_bytes == 4)
-        for (i = 0; i < nseek_points; i++) {
-          mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale;
-          data += 4;
-      } else if (seek_bytes == 3)
-        for (i = 0; i < nseek_points; i++) {
-          mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale;
-          data += 3;
-      } else if (seek_bytes == 2)
-        for (i = 0; i < nseek_points; i++) {
-          mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale;
-          data += 2;
-      } else                    /* seek_bytes == 1 */
-        for (i = 0; i < nseek_points; i++) {
-          mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale;
-          data += 1;
-        }
-    }
-  out_vbri:
-
-    GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %"
-        GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames,
-        GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes);
-
-    /* check for truncated file */
-    if (upstream_total_bytes && mp3parse->vbri_bytes &&
-        mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) {
-      GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
-          "invalidating VBRI header duration and size");
-      mp3parse->vbri_valid = FALSE;
-    } else {
-      mp3parse->vbri_valid = TRUE;
-    }
-  } else {
-    GST_DEBUG_OBJECT (mp3parse,
-        "Xing, LAME or VBRI header not found in first frame");
-  }
-
-  /* set duration if tables provided a valid one */
-  if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
-    gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
-        mp3parse->xing_total_time, 0);
-  }
-  if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) {
-    gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
-        mp3parse->vbri_total_time, 0);
-  }
-
-  /* tell baseclass how nicely we can seek, and a bitrate if one found */
-  seekable = GST_BASE_PARSE_SEEK_DEFAULT;
-  if ((mp3parse->xing_flags & XING_TOC_FLAG) && mp3parse->xing_bytes &&
-      mp3parse->xing_total_time)
-    seekable = GST_BASE_PARSE_SEEK_TABLE;
-
-  if (mp3parse->vbri_seek_table && mp3parse->vbri_bytes &&
-      mp3parse->vbri_total_time)
-    seekable = GST_BASE_PARSE_SEEK_TABLE;
-
-  if (mp3parse->xing_bitrate)
-    bitrate = mp3parse->xing_bitrate;
-  else if (mp3parse->vbri_bitrate)
-    bitrate = mp3parse->vbri_bitrate;
-  else
-    bitrate = 0;
-
-  gst_base_parse_set_seek (GST_BASE_PARSE (mp3parse), seekable, bitrate);
-}
-
-static GstFlowReturn
-gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame)
-{
-  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
-  GstBuffer *buf = frame->buffer;
-  guint bitrate, layer, rate, channels, version, mode, crc;
-
-  g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 4, GST_FLOW_ERROR);
-
-  if (!mp3_type_frame_length_from_header (mp3parse,
-          GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)),
-          &version, &layer, &channels, &bitrate, &rate, &mode, &crc))
-    goto broken_header;
-
-  if (G_UNLIKELY (channels != mp3parse->channels || rate != mp3parse->rate ||
-          layer != mp3parse->layer || version != mp3parse->version)) {
-    GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
-        "mpegversion", G_TYPE_INT, 1,
-        "mpegaudioversion", G_TYPE_INT, version,
-        "layer", G_TYPE_INT, layer,
-        "rate", G_TYPE_INT, rate,
-        "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
-    gst_buffer_set_caps (buf, caps);
-    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
-    gst_caps_unref (caps);
-
-    mp3parse->rate = rate;
-    mp3parse->channels = channels;
-    mp3parse->layer = layer;
-    mp3parse->version = version;
-
-    /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
-    if (mp3parse->layer == 1)
-      mp3parse->spf = 384;
-    else if (mp3parse->layer == 2)
-      mp3parse->spf = 1152;
-    else if (mp3parse->version == 1) {
-      mp3parse->spf = 1152;
-    } else {
-      /* MPEG-2 or "2.5" */
-      mp3parse->spf = 576;
-    }
-
-    /* lead_in:
-     * We start pushing 9 frames earlier (29 frames for MPEG2) than
-     * segment start to be able to decode the first frame we want.
-     * 9 (29) frames are the theoretical maximum of frames that contain
-     * data for the current frame (bit reservoir).
-     *
-     * lead_out:
-     * Some mp3 streams have an offset in the timestamps, for which we have to
-     * push the frame *after* the end position in order for the decoder to be
-     * able to decode everything up until the segment.stop position. */
-    gst_base_parse_set_frame_props (parse, mp3parse->rate, mp3parse->spf,
-        (version == 1) ? 10 : 30, 2);
-  }
-
-  mp3parse->hdr_bitrate = bitrate;
-
-  /* For first frame; check for seek tables and output a codec tag */
-  gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
-
-  /* store some frame info for later processing */
-  mp3parse->last_crc = crc;
-  mp3parse->last_mode = mode;
-
-  return GST_FLOW_OK;
-
-/* ERRORS */
-broken_header:
-  {
-    /* this really shouldn't ever happen */
-    GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL));
-    return GST_FLOW_ERROR;
-  }
-}
-
-static gboolean
-gst_mpeg_audio_parse_time_to_bytepos (GstMpegAudioParse * mp3parse,
-    GstClockTime ts, gint64 * bytepos)
-{
-  gint64 total_bytes;
-  GstClockTime total_time;
-
-  /* If XING seek table exists use this for time->byte conversion */
-  if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
-      (total_bytes = mp3parse->xing_bytes) &&
-      (total_time = mp3parse->xing_total_time)) {
-    gdouble fa, fb, fx;
-    gdouble percent =
-        CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) /
-        gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
-    gint index = CLAMP (percent, 0, 99);
-
-    fa = mp3parse->xing_seek_table[index];
-    if (index < 99)
-      fb = mp3parse->xing_seek_table[index + 1];
-    else
-      fb = 256.0;
-
-    fx = fa + (fb - fa) * (percent - index);
-
-    *bytepos = (1.0 / 256.0) * fx * total_bytes;
-
-    return TRUE;
-  }
-
-  if (mp3parse->vbri_seek_table && (total_bytes = mp3parse->vbri_bytes) &&
-      (total_time = mp3parse->vbri_total_time)) {
-    gint i, j;
-    gdouble a, b, fa, fb;
-
-    i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
-    i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1);
-
-    a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
-            mp3parse->vbri_seek_points));
-    fa = 0.0;
-    for (j = i; j >= 0; j--)
-      fa += mp3parse->vbri_seek_table[j];
-
-    if (i + 1 < mp3parse->vbri_seek_points) {
-      b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
-              mp3parse->vbri_seek_points));
-      fb = fa + mp3parse->vbri_seek_table[i + 1];
-    } else {
-      b = gst_guint64_to_gdouble (total_time);
-      fb = total_bytes;
-    }
-
-    *bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a);
-
-    return TRUE;
-  }
-
-  return FALSE;
-}
-
-static gboolean
-gst_mpeg_audio_parse_bytepos_to_time (GstMpegAudioParse * mp3parse,
-    gint64 bytepos, GstClockTime * ts)
-{
-  gint64 total_bytes;
-  GstClockTime total_time;
-
-  /* If XING seek table exists use this for byte->time conversion */
-  if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
-      (total_bytes = mp3parse->xing_bytes) &&
-      (total_time = mp3parse->xing_total_time)) {
-    gdouble fa, fb, fx;
-    gdouble pos;
-    gint index;
-
-    pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
-    index = CLAMP (pos, 0, 255);
-    fa = mp3parse->xing_seek_table_inverse[index];
-    if (index < 255)
-      fb = mp3parse->xing_seek_table_inverse[index + 1];
-    else
-      fb = 10000.0;
-
-    fx = fa + (fb - fa) * (pos - index);
-
-    *ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time);
-
-    return TRUE;
-  }
-
-  if (mp3parse->vbri_seek_table &&
-      (total_bytes = mp3parse->vbri_bytes) &&
-      (total_time = mp3parse->vbri_total_time)) {
-    gint i = 0;
-    guint64 sum = 0;
-    gdouble a, b, fa, fb;
-
-    do {
-      sum += mp3parse->vbri_seek_table[i];
-      i++;
-    } while (i + 1 < mp3parse->vbri_seek_points
-        && sum + mp3parse->vbri_seek_table[i] < bytepos);
-    i--;
-
-    a = gst_guint64_to_gdouble (sum);
-    fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
-            mp3parse->vbri_seek_points));
-
-    if (i + 1 < mp3parse->vbri_seek_points) {
-      b = a + mp3parse->vbri_seek_table[i + 1];
-      fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
-              mp3parse->vbri_seek_points));
-    } else {
-      b = total_bytes;
-      fb = gst_guint64_to_gdouble (total_time);
-    }
-
-    *ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a));
-
-    return TRUE;
-  }
-
-  return FALSE;
-}
-
-static gboolean
-gst_mpeg_audio_parse_convert (GstBaseParse * parse, GstFormat src_format,
-    gint64 src_value, GstFormat dest_format, gint64 * dest_value)
-{
-  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
-  gboolean res = FALSE;
-
-  if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES)
-    res =
-        gst_mpeg_audio_parse_time_to_bytepos (mp3parse, src_value, dest_value);
-  else if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME)
-    res = gst_mpeg_audio_parse_bytepos_to_time (mp3parse, src_value,
-        (GstClockTime *) dest_value);
-
-  /* if no tables, fall back to default estimated rate based conversion */
-  if (!res)
-    return gst_base_parse_convert_default (parse, src_format, src_value,
-        dest_format, dest_value);
-
-  return res;
-}
-
-static GstFlowReturn
-gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
-    GstBaseParseFrame * frame)
-{
-  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
-  GstTagList *taglist;
-
-  /* tag sending done late enough in hook to ensure pending events
-   * have already been sent */
-
-  if (!mp3parse->sent_codec_tag) {
-    gchar *codec;
-
-    /* codec tag */
-    if (mp3parse->layer == 3) {
-      codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)",
-          mp3parse->version, mp3parse->layer);
-    } else {
-      codec = g_strdup_printf ("MPEG %d Audio, Layer %d",
-          mp3parse->version, mp3parse->layer);
-    }
-    taglist = gst_tag_list_new ();
-    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
-        GST_TAG_AUDIO_CODEC, codec, NULL);
-    if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 &&
-        mp3parse->vbri_bitrate == 0) {
-      /* We don't have a VBR bitrate, so post the available bitrate as
-       * nominal and let baseparse calculate the real bitrate */
-      gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
-          GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL);
-    }
-    gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
-        GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
-    g_free (codec);
-
-    /* also signals the end of first-frame processing */
-    mp3parse->sent_codec_tag = TRUE;
-  }
-
-  /* we will create a taglist (if any of the parameters has changed)
-   * to add the tags that changed */
-  taglist = NULL;
-  if (mp3parse->last_posted_crc != mp3parse->last_crc) {
-    gboolean using_crc;
-
-    if (!taglist) {
-      taglist = gst_tag_list_new ();
-    }
-    mp3parse->last_posted_crc = mp3parse->last_crc;
-    if (mp3parse->last_posted_crc == CRC_PROTECTED) {
-      using_crc = TRUE;
-    } else {
-      using_crc = FALSE;
-    }
-    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC,
-        using_crc, NULL);
-  }
-
-  if (mp3parse->last_posted_channel_mode != mp3parse->last_mode) {
-    if (!taglist) {
-      taglist = gst_tag_list_new ();
-    }
-    mp3parse->last_posted_channel_mode = mp3parse->last_mode;
-
-    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
-        gst_mpeg_audio_channel_mode_get_nick (mp3parse->last_mode), NULL);
-  }
-
-  /* if the taglist exists, we need to send it */
-  if (taglist) {
-    gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
-        GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
-  }
-
-  /* usual clipping applies */
-  frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
-
-  return GST_FLOW_OK;
-}
diff --git a/gst/audioparsers/gstmpegaudioparse.h b/gst/audioparsers/gstmpegaudioparse.h
deleted file mode 100644
index 68b259751..000000000
--- a/gst/audioparsers/gstmpegaudioparse.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/* GStreamer MPEG audio parser
- * Copyright (C) 2006-2007 Jan Schmidt <thaytan@mad.scientist.com>
- * Copyright (C) 2010 Mark Nauwelaerts <mnauw users sf net>
- * Copyright (C) 2010 Nokia Corporation. All rights reserved.
- *   Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * 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_MPEG_AUDIO_PARSE_H__
-#define __GST_MPEG_AUDIO_PARSE_H__
-
-#include <gst/gst.h>
-#include <gst/baseparse/gstbaseparse.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_MPEG_AUDIO_PARSE \
-  (gst_mpeg_audio_parse_get_type())
-#define GST_MPEG_AUDIO_PARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_MPEG_AUDIO_PARSE, GstMpegAudioParse))
-#define GST_MPEG_AUDIO_PARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_MPEG_AUDIO_PARSE, GstMpegAudioParseClass))
-#define GST_IS_MPEG_AUDIO_PARSE(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_MPEG_AUDIO_PARSE))
-#define GST_IS_MPEG_AUDIO_PARSE_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_MPEG_AUDIO_PARSE))
-
-typedef struct _GstMpegAudioParse GstMpegAudioParse;
-typedef struct _GstMpegAudioParseClass GstMpegAudioParseClass;
-
-/**
- * GstMpegAudioParse:
- *
- * The opaque GstMpegAudioParse object
- */
-struct _GstMpegAudioParse {
-  GstBaseParse baseparse;
-
-  /*< private >*/
-  gint         rate;
-  gint         channels;
-  gint         layer;
-  gint         version;
-
-  GstClockTime max_bitreservoir;
-  /* samples per frame */
-  gint        spf;
-
-  gboolean     sent_codec_tag;
-  guint        last_posted_bitrate;
-  gint         last_posted_crc, last_crc;
-  guint        last_posted_channel_mode, last_mode;
-
-  /* Bitrate from non-vbr headers */
-  guint32      hdr_bitrate;
-
-  /* Xing info */
-  guint32      xing_flags;
-  guint32      xing_frames;
-  GstClockTime xing_total_time;
-  guint32      xing_bytes;
-  /* percent -> filepos mapping */
-  guchar       xing_seek_table[100];
-  /* filepos -> percent mapping */
-  guint16      xing_seek_table_inverse[256];
-  guint32      xing_vbr_scale;
-  guint        xing_bitrate;
-
-  /* VBRI info */
-  guint32      vbri_frames;
-  GstClockTime vbri_total_time;
-  guint32      vbri_bytes;
-  guint        vbri_bitrate;
-  guint        vbri_seek_points;
-  guint32     *vbri_seek_table;
-  gboolean     vbri_valid;
-
-  /* LAME info */
-  guint32      encoder_delay;
-  guint32      encoder_padding;
-};
-
-/**
- * GstMpegAudioParseClass:
- * @parent_class: Element parent class.
- *
- * The opaque GstMpegAudioParseClass data structure.
- */
-struct _GstMpegAudioParseClass {
-  GstBaseParseClass baseparse_class;
-};
-
-GType gst_mpeg_audio_parse_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_MPEG_AUDIO_PARSE_H__ */
diff --git a/gst/audioparsers/plugin.c b/gst/audioparsers/plugin.c
deleted file mode 100644
index 7d6d2f373..000000000
--- a/gst/audioparsers/plugin.c
+++ /dev/null
@@ -1,57 +0,0 @@
-/* GStreamer audio parsers
- * Copyright (C) 2009 Tim-Philipp Müller <tim centricular 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 "gstaacparse.h"
-#include "gstamrparse.h"
-#include "gstac3parse.h"
-#include "gstdcaparse.h"
-#include "gstflacparse.h"
-#include "gstmpegaudioparse.h"
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  gboolean ret;
-
-  ret = gst_element_register (plugin, "aacparse",
-      GST_RANK_PRIMARY + 1, GST_TYPE_AACPARSE);
-  ret &= gst_element_register (plugin, "amrparse",
-      GST_RANK_PRIMARY + 1, GST_TYPE_AMRPARSE);
-  ret &= gst_element_register (plugin, "ac3parse",
-      GST_RANK_PRIMARY + 1, GST_TYPE_AC3_PARSE);
-  ret &= gst_element_register (plugin, "dcaparse",
-      GST_RANK_PRIMARY + 1, GST_TYPE_DCA_PARSE);
-  ret &= gst_element_register (plugin, "flacparse",
-      GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE);
-  ret &= gst_element_register (plugin, "mpegaudioparse",
-      GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE);
-
-  return ret;
-}
-
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    "audioparsersbad",
-    "audioparsers",
-    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 293e97147..2f8207a8f 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -154,15 +154,11 @@ check_PROGRAMS = \
 	$(check_ofa)        \
 	$(check_timidity)  \
 	$(check_kate)  \
-	elements/aacparse \
-	elements/ac3parse \
-	elements/amrparse \
 	elements/autoconvert \
 	elements/autovideoconvert \
 	elements/asfmux \
 	elements/camerabin \
 	elements/dataurisrc \
-	elements/flacparse \
 	elements/legacyresample \
         $(check_jifmux) \
 	elements/jpegparse \
@@ -171,7 +167,6 @@ check_PROGRAMS = \
 	elements/mxfdemux \
 	elements/mxfmux \
 	elements/id3mux \
-	elements/mpegaudioparse \
 	pipelines/mxf \
 	$(check_mimic) \
 	elements/rtpmux \
@@ -237,23 +232,6 @@ elements_rtpmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-0.10 $(GST_BASE_LIBS)
 elements_assrender_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
 elements_assrender_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-0.10 -lgstapp-0.10 $(GST_BASE_LIBS) $(LDADD)
 
-# parser unit test convenience lib
-noinst_LTLIBRARIES = libparser.la
-libparser_la_SOURCES = elements/parser.c elements/parser.h
-libparser_la_CFLAGS = \
-	-I$(top_srcdir)/tests/check \
-	$(GST_CHECK_CFLAGS) $(GST_OPTION_CFLAGS)
-
-elements_aacparse_LDADD = libparser.la $(LDADD)
-
-elements_ac3parse_LDADD = libparser.la $(LDADD)
-
-elements_amrparse_LDADD = libparser.la $(LDADD)
-
-elements_flacparse_LDADD = libparser.la $(LDADD)
-
-elements_mpegaudioparse_LDADD = libparser.la $(LDADD)
-
 EXTRA_DIST = gst-plugins-bad.supp
 
 orc_cog_CFLAGS = $(ORC_CFLAGS)
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index 8a748233c..df8ab1761 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -1,7 +1,4 @@
 .dirstamp
-aacparse
-ac3parse
-amrparse
 asfmux
 assrender
 autoconvert
@@ -12,7 +9,6 @@ deinterleave
 dataurisrc
 faac
 faad
-flacparse
 gdpdepay
 gdppay
 id3mux
@@ -22,8 +18,8 @@ jifmux
 jpegparse
 kate
 legacyresample
+logoinsert
 mpeg2enc
-mpegaudioparse
 mplex
 mxfdemux
 mxfmux
diff --git a/tests/check/elements/aacparse.c b/tests/check/elements/aacparse.c
deleted file mode 100644
index af10a2779..000000000
--- a/tests/check/elements/aacparse.c
+++ /dev/null
@@ -1,240 +0,0 @@
-/*
- * GStreamer
- *
- * unit test for aacparse
- *
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-#include "parser.h"
-
-#define SRC_CAPS_CDATA "audio/mpeg, framed=(boolean)false, codec_data=(buffer)1190"
-#define SRC_CAPS_TMPL  "audio/mpeg, framed=(boolean)false, mpegversion=(int){2,4}"
-
-#define SINK_CAPS \
-    "audio/mpeg, framed=(boolean)true"
-#define SINK_CAPS_MPEG2 \
-    "audio/mpeg, framed=(boolean)true, mpegversion=2, rate=48000, channels=2"
-#define SINK_CAPS_MPEG4 \
-    "audio/mpeg, framed=(boolean)true, mpegversion=4, rate=96000, channels=2"
-#define SINK_CAPS_TMPL  "audio/mpeg, framed=(boolean)true, mpegversion=(int){2,4}"
-
-GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS_TMPL)
-    );
-
-GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS_TMPL)
-    );
-
-/* some data */
-static guint8 adif_header[] = {
-  'A', 'D', 'I', 'F'
-};
-
-static guint8 adts_frame_mpeg2[] = {
-  0xff, 0xf9, 0x4c, 0x80, 0x01, 0xff, 0xfc, 0x21, 0x10, 0xd3, 0x20, 0x0c,
-  0x32, 0x00, 0xc7
-};
-
-static guint8 adts_frame_mpeg4[] = {
-  0xff, 0xf1, 0x4c, 0x80, 0x01, 0xff, 0xfc, 0x21, 0x10, 0xd3, 0x20, 0x0c,
-  0x32, 0x00, 0xc7
-};
-
-static guint8 garbage_frame[] = {
-  0xff, 0xff, 0xff, 0xff, 0xff
-};
-
-/*
- * Test if the parser pushes data with ADIF header properly and detects the
- * stream to MPEG4 properly.
- */
-GST_START_TEST (test_parse_adif_normal)
-{
-  GstParserTest ptest;
-
-  /* ADIF header */
-  gst_parser_test_init (&ptest, adif_header, sizeof (adif_header), 1);
-  /* well, no garbage, followed by random data */
-  ptest.series[2].size = 100;
-  ptest.series[2].num = 3;
-  /* and we do not really expect output frames */
-  ptest.framed = FALSE;
-  /* Check that the negotiated caps are as expected */
-  /* For ADIF parser assumes that data is always version 4 */
-  ptest.sink_caps =
-      gst_caps_from_string (SINK_CAPS_MPEG4 ", stream-format=(string)adif");
-
-  gst_parser_test_run (&ptest, NULL);
-
-  gst_caps_unref (ptest.sink_caps);
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_adts_normal)
-{
-  gst_parser_test_normal (adts_frame_mpeg4, sizeof (adts_frame_mpeg4));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_adts_drain_single)
-{
-  gst_parser_test_drain_single (adts_frame_mpeg4, sizeof (adts_frame_mpeg4));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_adts_drain_garbage)
-{
-  gst_parser_test_drain_garbage (adts_frame_mpeg4, sizeof (adts_frame_mpeg4),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_adts_split)
-{
-  gst_parser_test_split (adts_frame_mpeg4, sizeof (adts_frame_mpeg4));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_adts_skip_garbage)
-{
-  gst_parser_test_skip_garbage (adts_frame_mpeg4, sizeof (adts_frame_mpeg4),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-/*
- * Test if the src caps are set according to stream format (MPEG version).
- */
-GST_START_TEST (test_parse_adts_detect_mpeg_version)
-{
-  gst_parser_test_output_caps (adts_frame_mpeg2, sizeof (adts_frame_mpeg2),
-      NULL, SINK_CAPS_MPEG2 ", stream-format=(string)adts");
-}
-
-GST_END_TEST;
-
-#define structure_get_int(s,f) \
-    (g_value_get_int(gst_structure_get_value(s,f)))
-#define fail_unless_structure_field_int_equals(s,field,num) \
-    fail_unless_equals_int (structure_get_int(s,field), num)
-/*
- * Test if the parser handles raw stream and codec_data info properly.
- */
-GST_START_TEST (test_parse_handle_codec_data)
-{
-  GstCaps *caps;
-  GstStructure *s;
-  const gchar *stream_format;
-
-  /* Push random data. It should get through since the parser should be
-   * initialized because it got codec_data in the caps */
-  caps = gst_parser_test_get_output_caps (NULL, 100, SRC_CAPS_CDATA);
-  fail_unless (caps != NULL);
-
-  /* Check that the negotiated caps are as expected */
-  /* When codec_data is present, parser assumes that data is version 4 */
-  GST_LOG ("aac output caps: %" GST_PTR_FORMAT, caps);
-  s = gst_caps_get_structure (caps, 0);
-  fail_unless (gst_structure_has_name (s, "audio/mpeg"));
-  fail_unless_structure_field_int_equals (s, "mpegversion", 4);
-  fail_unless_structure_field_int_equals (s, "channels", 2);
-  fail_unless_structure_field_int_equals (s, "rate", 48000);
-  fail_unless (gst_structure_has_field (s, "codec_data"));
-  fail_unless (gst_structure_has_field (s, "stream-format"));
-  stream_format = gst_structure_get_string (s, "stream-format");
-  fail_unless (strcmp (stream_format, "raw") == 0);
-
-  gst_caps_unref (caps);
-}
-
-GST_END_TEST;
-
-
-static Suite *
-aacparse_suite (void)
-{
-  Suite *s = suite_create ("aacparse");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  /* ADIF tests */
-  tcase_add_test (tc_chain, test_parse_adif_normal);
-
-  /* ADTS tests */
-  tcase_add_test (tc_chain, test_parse_adts_normal);
-  tcase_add_test (tc_chain, test_parse_adts_drain_single);
-  tcase_add_test (tc_chain, test_parse_adts_drain_garbage);
-  tcase_add_test (tc_chain, test_parse_adts_split);
-  tcase_add_test (tc_chain, test_parse_adts_skip_garbage);
-  tcase_add_test (tc_chain, test_parse_adts_detect_mpeg_version);
-
-  /* Other tests */
-  tcase_add_test (tc_chain, test_parse_handle_codec_data);
-
-  return s;
-}
-
-
-/*
- * TODO:
- *   - Both push- and pull-modes need to be tested
- *      * Pull-mode & EOS
- */
-
-int
-main (int argc, char **argv)
-{
-  int nf;
-
-  Suite *s = aacparse_suite ();
-  SRunner *sr = srunner_create (s);
-
-  gst_check_init (&argc, &argv);
-
-  /* init test context */
-  ctx_factory = "aacparse";
-  ctx_sink_template = &sinktemplate;
-  ctx_src_template = &srctemplate;
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf = srunner_ntests_failed (sr);
-  srunner_free (sr);
-
-  return nf;
-}
diff --git a/tests/check/elements/ac3parse.c b/tests/check/elements/ac3parse.c
deleted file mode 100644
index 03e8e1dc8..000000000
--- a/tests/check/elements/ac3parse.c
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- * GStreamer
- *
- * unit test for ac3parse
- *
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-#include "parser.h"
-
-#define SRC_CAPS_TMPL   "audio/x-ac3, framed=(boolean)false"
-#define SINK_CAPS_TMPL  "audio/x-ac3, framed=(boolean)true"
-
-GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS_TMPL)
-    );
-
-GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS_TMPL)
-    );
-
-/* some data */
-
-static guint8 ac3_frame[512] = {
-  0x0b, 0x77, 0xb6, 0xa8, 0x10, 0x40, 0x2f, 0x84,
-  0x29, 0xcb, 0xfe, 0x75, 0x7c, 0xf9, 0xf3, 0xe7,
-  0xcf, 0x9f, 0x3e, 0x7c, 0xf9, 0xf3, 0xe7, 0xcf,
-  0x9f, 0x3e, 0x7c, 0xf9, 0xf3, 0xe7, 0xcf, 0x9f,
-  0x3e, 0x7c, 0xf9, 0xf3, 0xe7, 0xcf, 0x9f, 0x3e,
-  0x7c, 0xf9, 0xf3, 0xe7, 0xcf, 0x9f, 0x3e, 0x7c,
-  0xf9, 0xf3, 0xe7, 0xcf, 0x9f, 0x3e, 0x7c, 0xf9,
-  0xf3, 0xe7, 0xcf, 0x9f, 0x3e, 0x7c, 0xf9, 0xf3,
-  0xe7, 0xcf, 0x9f, 0x3e, 0x7c, 0xf9, 0xf3, 0xe7,
-  0xcf, 0x9f, 0x3e, 0x32, 0xd3, 0xff, 0xc0, 0x06,
-  0xe9, 0x40, 0x00, 0x6e, 0x94, 0x00, 0x06, 0xe9,
-  0x40, 0x00, 0x6e, 0x94, 0x00, 0x06, 0xe9, 0x40,
-  0x00, 0x6e, 0x90, 0x00, 0x00, 0x00, 0x00, 0x00,
-  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-};
-
-static guint8 garbage_frame[] = {
-  0xff, 0xff, 0xff, 0xff, 0xff
-};
-
-
-GST_START_TEST (test_parse_normal)
-{
-  gst_parser_test_normal (ac3_frame, sizeof (ac3_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_drain_single)
-{
-  gst_parser_test_drain_single (ac3_frame, sizeof (ac3_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_drain_garbage)
-{
-  gst_parser_test_drain_garbage (ac3_frame, sizeof (ac3_frame),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_split)
-{
-  gst_parser_test_split (ac3_frame, sizeof (ac3_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_skip_garbage)
-{
-  gst_parser_test_skip_garbage (ac3_frame, sizeof (ac3_frame),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_detect_stream)
-{
-  gst_parser_test_output_caps (ac3_frame, sizeof (ac3_frame),
-      NULL, SINK_CAPS_TMPL ",channels=1,rate=48000");
-}
-
-GST_END_TEST;
-
-
-static Suite *
-ac3parse_suite (void)
-{
-  Suite *s = suite_create ("ac3parse");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_parse_normal);
-  tcase_add_test (tc_chain, test_parse_drain_single);
-  tcase_add_test (tc_chain, test_parse_drain_garbage);
-  tcase_add_test (tc_chain, test_parse_split);
-  tcase_add_test (tc_chain, test_parse_skip_garbage);
-  tcase_add_test (tc_chain, test_parse_detect_stream);
-
-  return s;
-}
-
-
-/*
- * TODO:
- *   - Both push- and pull-modes need to be tested
- *      * Pull-mode & EOS
- */
-
-int
-main (int argc, char **argv)
-{
-  int nf;
-
-  Suite *s = ac3parse_suite ();
-  SRunner *sr = srunner_create (s);
-
-  gst_check_init (&argc, &argv);
-
-  /* init test context */
-  ctx_factory = "ac3parse";
-  ctx_sink_template = &sinktemplate;
-  ctx_src_template = &srctemplate;
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf = srunner_ntests_failed (sr);
-  srunner_free (sr);
-
-  return nf;
-}
diff --git a/tests/check/elements/amrparse.c b/tests/check/elements/amrparse.c
deleted file mode 100644
index e5d64ca57..000000000
--- a/tests/check/elements/amrparse.c
+++ /dev/null
@@ -1,327 +0,0 @@
-/*
- * GStreamer
- *
- * unit test for amrparse
- *
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-#include "parser.h"
-
-#define SRC_CAPS_NB  "audio/x-amr-nb-sh"
-#define SRC_CAPS_WB  "audio/x-amr-wb-sh"
-#define SRC_CAPS_ANY "ANY"
-
-#define SINK_CAPS_NB  "audio/AMR, rate=8000 , channels=1"
-#define SINK_CAPS_WB  "audio/AMR-WB, rate=16000 , channels=1"
-#define SINK_CAPS_ANY "ANY"
-
-#define AMR_FRAME_DURATION (GST_SECOND/50)
-
-static GstStaticPadTemplate sinktemplate_nb = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS_NB)
-    );
-
-static GstStaticPadTemplate sinktemplate_wb = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS_WB)
-    );
-
-static GstStaticPadTemplate srctemplate_nb = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS_NB)
-    );
-
-static GstStaticPadTemplate srctemplate_wb = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS_WB)
-    );
-
-
-/* some data */
-
-static guint8 frame_data_nb[] = {
-  0x0c, 0x56, 0x3c, 0x52, 0xe0, 0x61, 0xbc, 0x45,
-  0x0f, 0x98, 0x2e, 0x01, 0x42, 0x02
-};
-
-static guint8 frame_data_wb[] = {
-  0x08, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06,
-  0x07, 0x08, 0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
-  0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16
-};
-
-static guint8 frame_hdr_nb[] = {
-  '#', '!', 'A', 'M', 'R', '\n'
-};
-
-static guint8 frame_hdr_wb[] = {
-  '#', '!', 'A', 'M', 'R', '-', 'W', 'B', '\n'
-};
-
-static guint8 garbage_frame[] = {
-  0xff, 0xff, 0xff, 0xff, 0xff
-};
-
-
-GST_START_TEST (test_parse_nb_normal)
-{
-  gst_parser_test_normal (frame_data_nb, sizeof (frame_data_nb));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_nb_drain_single)
-{
-  gst_parser_test_drain_single (frame_data_nb, sizeof (frame_data_nb));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_nb_drain_garbage)
-{
-  gst_parser_test_drain_garbage (frame_data_nb, sizeof (frame_data_nb),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_nb_split)
-{
-  gst_parser_test_split (frame_data_nb, sizeof (frame_data_nb));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_nb_skip_garbage)
-{
-  gst_parser_test_skip_garbage (frame_data_nb, sizeof (frame_data_nb),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_nb_detect_stream)
-{
-  GstParserTest ptest;
-  GstCaps *old_ctx_caps;
-
-  /* no input caps, override ctx */
-  old_ctx_caps = ctx_input_caps;
-  ctx_input_caps = NULL;
-
-  /* AMR-NB header */
-  gst_parser_test_init (&ptest, frame_hdr_nb, sizeof (frame_hdr_nb), 1);
-  /* well, no garbage, followed by real data */
-  ptest.series[2].data = frame_data_nb;
-  ptest.series[2].size = sizeof (frame_data_nb);
-  ptest.series[2].num = 10;
-  /* header gets dropped, so ... */
-  /* buffer count will not match */
-  ptest.framed = FALSE;
-  /* total size a bit less */
-  ptest.dropped = sizeof (frame_hdr_nb);
-
-  /* Check that the negotiated caps are as expected */
-  ptest.sink_caps = gst_caps_from_string (SINK_CAPS_NB);
-
-  gst_parser_test_run (&ptest, NULL);
-
-  gst_caps_unref (ptest.sink_caps);
-
-  ctx_input_caps = old_ctx_caps;
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_wb_normal)
-{
-  gst_parser_test_normal (frame_data_wb, sizeof (frame_data_wb));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_wb_drain_single)
-{
-  gst_parser_test_drain_single (frame_data_wb, sizeof (frame_data_wb));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_wb_drain_garbage)
-{
-  gst_parser_test_drain_garbage (frame_data_wb, sizeof (frame_data_wb),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_wb_split)
-{
-  gst_parser_test_split (frame_data_wb, sizeof (frame_data_wb));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_wb_skip_garbage)
-{
-  gst_parser_test_skip_garbage (frame_data_wb, sizeof (frame_data_wb),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_wb_detect_stream)
-{
-  GstParserTest ptest;
-  GstCaps *old_ctx_caps;
-
-  /* no input caps, override ctx */
-  old_ctx_caps = ctx_input_caps;
-  ctx_input_caps = NULL;
-
-  /* AMR-WB header */
-  gst_parser_test_init (&ptest, frame_hdr_wb, sizeof (frame_hdr_wb), 1);
-  /* well, no garbage, followed by real data */
-  ptest.series[2].data = frame_data_wb;
-  ptest.series[2].size = sizeof (frame_data_wb);
-  ptest.series[2].num = 10;
-  /* header gets dropped, so ... */
-  /* buffer count will not match */
-  ptest.framed = FALSE;
-  /* total size a bit less */
-  ptest.dropped = sizeof (frame_hdr_wb);
-
-  /* Check that the negotiated caps are as expected */
-  ptest.sink_caps = gst_caps_from_string (SINK_CAPS_WB);
-
-  gst_parser_test_run (&ptest, NULL);
-
-  gst_caps_unref (ptest.sink_caps);
-
-  ctx_input_caps = old_ctx_caps;
-}
-
-GST_END_TEST;
-
-
-
-/*
- * Create test suite.
- */
-static Suite *
-amrnb_parse_suite (void)
-{
-  Suite *s = suite_create ("amrwb_parse");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  /* AMR-NB tests */
-  tcase_add_test (tc_chain, test_parse_nb_normal);
-  tcase_add_test (tc_chain, test_parse_nb_drain_single);
-  tcase_add_test (tc_chain, test_parse_nb_drain_garbage);
-  tcase_add_test (tc_chain, test_parse_nb_split);
-  tcase_add_test (tc_chain, test_parse_nb_detect_stream);
-  tcase_add_test (tc_chain, test_parse_nb_skip_garbage);
-
-  return s;
-}
-
-static Suite *
-amrwb_parse_suite (void)
-{
-  Suite *s = suite_create ("amrnb_parse");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  /* AMR-WB tests */
-  tcase_add_test (tc_chain, test_parse_wb_normal);
-  tcase_add_test (tc_chain, test_parse_wb_drain_single);
-  tcase_add_test (tc_chain, test_parse_wb_drain_garbage);
-  tcase_add_test (tc_chain, test_parse_wb_split);
-  tcase_add_test (tc_chain, test_parse_wb_detect_stream);
-  tcase_add_test (tc_chain, test_parse_wb_skip_garbage);
-
-  return s;
-}
-
-/*
- * TODO:
- *   - Both push- and pull-modes need to be tested
- *      * Pull-mode & EOS
- */
-
-int
-main (int argc, char **argv)
-{
-  int nf;
-  GstCaps *caps;
-
-  Suite *s = amrnb_parse_suite ();
-  SRunner *sr = srunner_create (s);
-
-  gst_check_init (&argc, &argv);
-
-  /* init test context */
-  ctx_factory = "amrparse";
-  ctx_sink_template = &sinktemplate_nb;
-  ctx_src_template = &srctemplate_nb;
-  caps = gst_caps_from_string (SRC_CAPS_NB);
-  g_assert (caps);
-  ctx_input_caps = caps;
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf = srunner_ntests_failed (sr);
-  srunner_free (sr);
-  gst_caps_unref (caps);
-
-  s = amrwb_parse_suite ();
-  sr = srunner_create (s);
-
-  ctx_sink_template = &sinktemplate_wb;
-  ctx_src_template = &srctemplate_wb;
-  caps = gst_caps_from_string (SRC_CAPS_WB);
-  g_assert (caps);
-  ctx_input_caps = caps;
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf += srunner_ntests_failed (sr);
-  srunner_free (sr);
-  gst_caps_unref (caps);
-
-  return nf;
-}
diff --git a/tests/check/elements/flacparse.c b/tests/check/elements/flacparse.c
deleted file mode 100644
index 0c25bc6f5..000000000
--- a/tests/check/elements/flacparse.c
+++ /dev/null
@@ -1,299 +0,0 @@
-/*
- * GStreamer
- *
- * unit test for flacparse
- *
- * Copyright (C) 2010 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-#include "parser.h"
-
-#define SRC_CAPS_TMPL  "audio/x-flac, framed=(boolean)false"
-#define SINK_CAPS_TMPL  "audio/x-flac, framed=(boolean)true"
-
-GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS_TMPL)
-    );
-
-GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS_TMPL)
-    );
-
-/* some data */
-static guint8 streaminfo_header[] = {
-  0x7f, 0x46, 0x4c, 0x41, 0x43, 0x01, 0x00, 0x00,
-  0x02, 0x66, 0x4c, 0x61, 0x43, 0x00, 0x00, 0x00,
-  0x22, 0x12, 0x00, 0x12, 0x00, 0x00, 0x00, 0x00,
-  0x00, 0x00, 0x00, 0x0a, 0xc4, 0x40, 0xf0, 0x00,
-  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-  0x00, 0x00, 0x00
-};
-
-static guint8 comment_header[] = {
-  0x84, 0x00, 0x00, 0x08, 0x00, 0x00, 0x00, 0x00,
-  0x00, 0x00, 0x00, 0x00
-};
-
-static guint8 flac_frame[] = {
-  0xff, 0xf8, 0xa9, 0x08, 0x00, 0x50, 0x18, 0x06,
-  0x6a, 0x0c, 0xce, 0x13, 0x24, 0x19, 0x68, 0x00,
-  0x46, 0x23, 0x08, 0xca, 0xcb, 0x58, 0x9c, 0x26,
-  0x92, 0x30, 0xa6, 0x29, 0x8a, 0xca, 0xd1, 0x18,
-  0xae, 0x26, 0x5c, 0x90, 0x60, 0xbf, 0x11, 0xad,
-  0x43, 0x02, 0x06, 0x26, 0xbd, 0x35, 0xdd, 0xa3,
-  0x11, 0xa6, 0x4d, 0x18, 0x8c, 0x9a, 0xe4, 0x62,
-  0xd9, 0x23, 0x11, 0x8b, 0xcb, 0x56, 0x55, 0x45,
-  0xc2, 0x18, 0x56, 0xa2, 0xe2, 0xe1, 0x18, 0x99,
-  0x54, 0x98, 0x46, 0x4d, 0x08, 0x70, 0x9a, 0x64,
-  0xc4, 0x18, 0x4f, 0x27, 0x64, 0x31, 0x66, 0x27,
-  0x79, 0x19, 0x3c, 0x8c, 0x8c, 0xa3, 0x44, 0x18,
-  0x23, 0xd2, 0x6b, 0x8b, 0x64, 0x8c, 0x21, 0x84,
-  0xd6, 0x23, 0x13, 0x13, 0x2d, 0x44, 0xca, 0x5a,
-  0x23, 0x09, 0x93, 0x25, 0x18, 0x10, 0x61, 0x38,
-  0xb4, 0x60, 0x8f, 0x2c, 0x8d, 0x26, 0xb4, 0xc9,
-  0xd9, 0x19, 0x19, 0x34, 0xd7, 0x31, 0x06, 0x10,
-  0xc4, 0x30, 0x83, 0x17, 0xe2, 0x0c, 0x2c, 0xc4,
-  0xc8, 0xc9, 0x3c, 0x5e, 0x93, 0x11, 0x8a, 0x62,
-  0x64, 0x8c, 0x26, 0x23, 0x22, 0x30, 0x9a, 0x58,
-  0x86, 0x04, 0x18, 0x4c, 0xab, 0x2b, 0x26, 0x5c,
-  0x46, 0x88, 0xcb, 0xb1, 0x0d, 0x26, 0xbb, 0x5e,
-  0x8c, 0xa7, 0x64, 0x31, 0x3d, 0x31, 0x06, 0x26,
-  0x43, 0x17, 0xa3, 0x08, 0x61, 0x06, 0x17, 0xc4,
-  0x62, 0xec, 0x4d, 0x4b, 0x2e, 0x2d, 0x4a, 0x94,
-  0xa4, 0xc2, 0x31, 0x4c, 0x4c, 0x20, 0xc0, 0x83,
-  0x14, 0x8c, 0x27, 0x8b, 0x31, 0x23, 0x2f, 0x23,
-  0x11, 0x91, 0x94, 0x65, 0x1a, 0x20, 0xc2, 0x18,
-  0x86, 0x51, 0x88, 0x62, 0x7c, 0x43, 0x2e, 0xa3,
-  0x04, 0x18, 0x8c, 0x20, 0xc2, 0xf5, 0xaa, 0x94,
-  0xc2, 0x31, 0x32, 0xd2, 0xb2, 0xa2, 0x30, 0xba,
-  0x10, 0xc2, 0xb5, 0x89, 0xa5, 0x18, 0x10, 0x62,
-  0x9a, 0x10, 0x61, 0x19, 0x72, 0x71, 0x1a, 0xb9,
-  0x0c, 0x23, 0x46, 0x10, 0x62, 0x78, 0x81, 0x82,
-  0x3d, 0x75, 0xea, 0x6b, 0x51, 0x8b, 0x61, 0x06,
-  0x08, 0x62, 0x32, 0x5e, 0x84, 0x18, 0x27, 0x25,
-  0xc2, 0x6a, 0x4b, 0x51, 0x31, 0x34, 0x5e, 0x29,
-  0xa1, 0x3c, 0x4d, 0x26, 0x23, 0x10, 0xc2, 0x6b,
-  0xb1, 0x0d, 0x11, 0xae, 0x46, 0x88, 0x31, 0x35,
-  0xb1, 0x06, 0x08, 0x79, 0x7e, 0x4f, 0x53, 0x23,
-  0x29, 0xa4, 0x30, 0x20, 0x30, 0x23, 0x5a, 0xb2,
-  0xc8, 0x60, 0x9c, 0x93, 0x13, 0x17, 0x92, 0x98,
-  0x46, 0x13, 0x54, 0x53, 0x08, 0xcb, 0x13, 0xa1,
-  0x1a, 0x89, 0xe5, 0x46, 0x08, 0x18, 0x10, 0x30,
-  0x9d, 0x68, 0xc2, 0x1c, 0x46, 0x46, 0xae, 0x62,
-  0x1a, 0x46, 0x4e, 0x4d, 0x34, 0x8c, 0xbd, 0x26,
-  0xc0, 0x40, 0x62, 0xc9, 0xa9, 0x31, 0x74, 0xa8,
-  0x99, 0x52, 0xb0, 0x8c, 0xa9, 0x29, 0x84, 0x61,
-  0x19, 0x54, 0x43, 0x02, 0x06, 0x04, 0x32, 0xe5,
-  0x18, 0x21, 0x91, 0x8b, 0xf2, 0xcc, 0x10, 0x30,
-  0x8e, 0x23, 0xc4, 0x76, 0x43, 0x08, 0x30, 0x83,
-  0x08, 0x62, 0x6c, 0x4e, 0xe2, 0x35, 0x96, 0xd0,
-  0x8e, 0x89, 0x97, 0x42, 0x18, 0x91, 0x84, 0x61,
-  0x3c, 0x26, 0xa5, 0x2c, 0x4e, 0x17, 0x94, 0xb8,
-  0xb5, 0xa4, 0xcb, 0x88, 0xc9, 0x84, 0x18, 0xb9,
-  0x84, 0x19, 0x23, 0x2d, 0xa4, 0x64, 0x62, 0x18,
-  0x86, 0x53, 0x93, 0xcb, 0x30, 0x8f, 0x2f, 0x93,
-  0x55, 0xc4, 0xd7, 0x08, 0x62, 0xb8, 0x46, 0x84,
-  0x68, 0xa3, 0x02, 0xaf, 0x33
-};
-
-static guint8 garbage_frame[] = {
-  0xff, 0xff, 0xff, 0xff, 0xff
-};
-
-
-GST_START_TEST (test_parse_flac_normal)
-{
-  gst_parser_test_normal (flac_frame, sizeof (flac_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_flac_drain_single)
-{
-  gst_parser_test_drain_single (flac_frame, sizeof (flac_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_flac_drain_garbage)
-{
-  /* We always output the after frame garbage too because we
-   * have no way of detecting it
-   */
-#if 0
-  gst_parser_test_drain_garbage (flac_frame, sizeof (flac_frame),
-      garbage_frame, sizeof (garbage_frame));
-#endif
-  guint8 frame[sizeof (flac_frame) + sizeof (garbage_frame)];
-
-  memcpy (frame, flac_frame, sizeof (flac_frame));
-  memcpy (frame + sizeof (flac_frame), garbage_frame, sizeof (garbage_frame));
-
-  gst_parser_test_drain_single (frame, sizeof (frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_flac_split)
-{
-  gst_parser_test_split (flac_frame, sizeof (flac_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_flac_skip_garbage)
-{
-  /* We always include the garbage into the frame because
-   * we have no easy way for finding the real end of the
-   * frame. The decoder will later skip the garbage
-   */
-#if 0
-  gst_parser_test_skip_garbage (flac_frame, sizeof (flac_frame),
-      garbage_frame, sizeof (garbage_frame));
-#endif
-  guint8 frame[sizeof (flac_frame) + sizeof (garbage_frame)];
-
-  memcpy (frame, flac_frame, sizeof (flac_frame));
-  memcpy (frame + sizeof (flac_frame), garbage_frame, sizeof (garbage_frame));
-
-  gst_parser_test_normal (frame, sizeof (frame));
-}
-
-GST_END_TEST;
-
-
-#define structure_get_int(s,f) \
-    (g_value_get_int(gst_structure_get_value(s,f)))
-#define fail_unless_structure_field_int_equals(s,field,num) \
-    fail_unless_equals_int (structure_get_int(s,field), num)
-/*
- * Test if the parser handles raw stream and codec_data info properly.
- */
-GST_START_TEST (test_parse_flac_detect_stream)
-{
-  GstCaps *caps;
-  GstStructure *s;
-  const GValue *streamheader;
-  GArray *bufarr;
-  gint i;
-
-  /* Push random data. It should get through since the parser should be
-   * initialized because it got codec_data in the caps */
-  caps = gst_parser_test_get_output_caps (flac_frame, sizeof (flac_frame),
-      SRC_CAPS_TMPL);
-  fail_unless (caps != NULL);
-
-  /* Check that the negotiated caps are as expected */
-  /* When codec_data is present, parser assumes that data is version 4 */
-  GST_LOG ("flac output caps: %" GST_PTR_FORMAT, caps);
-  s = gst_caps_get_structure (caps, 0);
-  fail_unless (gst_structure_has_name (s, "audio/x-flac"));
-  fail_unless_structure_field_int_equals (s, "channels", 1);
-  fail_unless_structure_field_int_equals (s, "rate", 44100);
-  fail_unless (gst_structure_has_field (s, "streamheader"));
-  streamheader = gst_structure_get_value (s, "streamheader");
-  fail_unless (G_VALUE_TYPE (streamheader) == GST_TYPE_ARRAY);
-  bufarr = g_value_peek_pointer (streamheader);
-  fail_unless (bufarr->len == 2);
-  for (i = 0; i < bufarr->len; i++) {
-    GstBuffer *buf;
-    GValue *bufval = &g_array_index (bufarr, GValue, i);
-
-    fail_unless (G_VALUE_TYPE (bufval) == GST_TYPE_BUFFER);
-    buf = g_value_peek_pointer (bufval);
-    if (i == 0) {
-      fail_unless (GST_BUFFER_SIZE (buf) == sizeof (streaminfo_header));
-      fail_unless (memcmp (buf, streaminfo_header, sizeof (streaminfo_header)));
-    } else if (i == 1) {
-      fail_unless (GST_BUFFER_SIZE (buf) == sizeof (comment_header));
-      fail_unless (memcmp (buf, comment_header, sizeof (comment_header)));
-    }
-  }
-
-  gst_caps_unref (caps);
-}
-
-GST_END_TEST;
-
-
-static Suite *
-flacparse_suite (void)
-{
-  Suite *s = suite_create ("flacparse");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_parse_flac_normal);
-  tcase_add_test (tc_chain, test_parse_flac_drain_single);
-  tcase_add_test (tc_chain, test_parse_flac_drain_garbage);
-  tcase_add_test (tc_chain, test_parse_flac_split);
-  tcase_add_test (tc_chain, test_parse_flac_skip_garbage);
-
-  /* Other tests */
-  tcase_add_test (tc_chain, test_parse_flac_detect_stream);
-
-  return s;
-}
-
-
-/*
- * TODO:
- *   - Both push- and pull-modes need to be tested
- *      * Pull-mode & EOS
- */
-
-int
-main (int argc, char **argv)
-{
-  int nf;
-
-  Suite *s = flacparse_suite ();
-  SRunner *sr = srunner_create (s);
-
-  gst_check_init (&argc, &argv);
-
-  /* init test context */
-  ctx_factory = "flacparse";
-  ctx_sink_template = &sinktemplate;
-  ctx_src_template = &srctemplate;
-  ctx_discard = 3;
-  ctx_headers[0].data = streaminfo_header;
-  ctx_headers[0].size = sizeof (streaminfo_header);
-  ctx_headers[1].data = comment_header;
-  ctx_headers[1].size = sizeof (comment_header);
-  /* custom offsets, and ts always repeatedly 0 */
-  ctx_no_metadata = TRUE;
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf = srunner_ntests_failed (sr);
-  srunner_free (sr);
-
-  return nf;
-}
diff --git a/tests/check/elements/mpegaudioparse.c b/tests/check/elements/mpegaudioparse.c
deleted file mode 100644
index 69a08640f..000000000
--- a/tests/check/elements/mpegaudioparse.c
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- * GStreamer
- *
- * unit test for aacparse
- *
- * Copyright (C) 2008 Nokia Corporation. All rights reserved.
- *
- * Contact: Stefan Kost <stefan.kost@nokia.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-#include "parser.h"
-
-#define SRC_CAPS_TMPL   "audio/mpeg, parsed=(boolean)false, mpegversion=(int)1"
-#define SINK_CAPS_TMPL  "audio/mpeg, parsed=(boolean)true, mpegversion=(int)1"
-
-GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SINK_CAPS_TMPL)
-    );
-
-GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (SRC_CAPS_TMPL)
-    );
-
-const gchar *factory = "aacparse";
-
-/* some data */
-static guint8 mp3_frame[384] = {
-  0xff, 0xfb, 0x94, 0xc4, 0xff, 0x83, 0xc0, 0x00,
-  0x01, 0xa4, 0x00, 0x00, 0x00, 0x20, 0x00, 0x00,
-  0x34, 0x80, 0x00, 0x00, 0x04, 0x00,
-};
-
-static guint8 garbage_frame[] = {
-  0xff, 0xff, 0xff, 0xff, 0xff
-};
-
-
-GST_START_TEST (test_parse_normal)
-{
-  gst_parser_test_normal (mp3_frame, sizeof (mp3_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_drain_single)
-{
-  gst_parser_test_drain_single (mp3_frame, sizeof (mp3_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_drain_garbage)
-{
-  gst_parser_test_drain_garbage (mp3_frame, sizeof (mp3_frame),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_split)
-{
-  gst_parser_test_split (mp3_frame, sizeof (mp3_frame));
-}
-
-GST_END_TEST;
-
-
-GST_START_TEST (test_parse_skip_garbage)
-{
-  gst_parser_test_skip_garbage (mp3_frame, sizeof (mp3_frame),
-      garbage_frame, sizeof (garbage_frame));
-}
-
-GST_END_TEST;
-
-
-#define structure_get_int(s,f) \
-    (g_value_get_int(gst_structure_get_value(s,f)))
-#define fail_unless_structure_field_int_equals(s,field,num) \
-    fail_unless_equals_int (structure_get_int(s,field), num)
-
-GST_START_TEST (test_parse_detect_stream)
-{
-  GstStructure *s;
-  GstCaps *caps;
-
-  caps = gst_parser_test_get_output_caps (mp3_frame, sizeof (mp3_frame), NULL);
-
-  fail_unless (caps != NULL);
-
-  GST_LOG ("mpegaudio output caps: %" GST_PTR_FORMAT, caps);
-  s = gst_caps_get_structure (caps, 0);
-  fail_unless (gst_structure_has_name (s, "audio/mpeg"));
-  fail_unless_structure_field_int_equals (s, "mpegversion", 1);
-  fail_unless_structure_field_int_equals (s, "layer", 3);
-  fail_unless_structure_field_int_equals (s, "channels", 1);
-  fail_unless_structure_field_int_equals (s, "rate", 48000);
-
-  gst_caps_unref (caps);
-}
-
-GST_END_TEST;
-
-
-static Suite *
-mpegaudioparse_suite (void)
-{
-  Suite *s = suite_create ("mpegaudioparse");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_parse_normal);
-  tcase_add_test (tc_chain, test_parse_drain_single);
-  tcase_add_test (tc_chain, test_parse_drain_garbage);
-  tcase_add_test (tc_chain, test_parse_split);
-  tcase_add_test (tc_chain, test_parse_skip_garbage);
-  tcase_add_test (tc_chain, test_parse_detect_stream);
-
-  return s;
-}
-
-
-/*
- * TODO:
- *   - Both push- and pull-modes need to be tested
- *      * Pull-mode & EOS
- */
-
-int
-main (int argc, char **argv)
-{
-  int nf;
-
-  Suite *s = mpegaudioparse_suite ();
-  SRunner *sr = srunner_create (s);
-
-  gst_check_init (&argc, &argv);
-
-  /* init test context */
-  ctx_factory = "mpegaudioparse";
-  ctx_sink_template = &sinktemplate;
-  ctx_src_template = &srctemplate;
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf = srunner_ntests_failed (sr);
-  srunner_free (sr);
-
-  return nf;
-}