ext/ogg/gstoggaviparse.c - imx-gst-plugins-base - Git at Google

 /* GStreamer
  * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
  *
  * gstoggaviparse.c: ogg avi stream parser
  *
  * 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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */

 /*
  * Ogg in AVI is mostly done for vorbis audio. In the codec_data we receive the
  * first 3 packets of the raw vorbis data. On the sinkpad we receive full-blown Ogg
  * pages.
  * Before extracting the packets out of the ogg pages, we push the raw vorbis
  * header packets to the decoder.
  * We don't use the incomming timestamps but use the ganulepos on the ogg pages
  * directly.
  * This parser only does ogg/vorbis for now.
  */

 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
 #include <gst/gst.h>
 #include <ogg/ogg.h>
 #include <string.h>

 #include "gstogg.h"

 GST_DEBUG_CATEGORY_STATIC (gst_ogg_avi_parse_debug);
 #define GST_CAT_DEFAULT gst_ogg_avi_parse_debug

 #define GST_TYPE_OGG_AVI_PARSE (gst_ogg_avi_parse_get_type())
 #define GST_OGG_AVI_PARSE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OGG_AVI_PARSE, GstOggAviParse))
 #define GST_OGG_AVI_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OGG_AVI_PARSE, GstOggAviParse))
 #define GST_IS_OGG_AVI_PARSE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OGG_AVI_PARSE))
 #define GST_IS_OGG_AVI_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OGG_AVI_PARSE))

 static GType gst_ogg_avi_parse_get_type (void);

 typedef struct _GstOggAviParse GstOggAviParse;
 typedef struct _GstOggAviParseClass GstOggAviParseClass;

 struct _GstOggAviParse
 {
   GstElement element;

   GstPad *sinkpad;
   GstPad *srcpad;

   gboolean discont;
   gint serial;

   ogg_sync_state sync;
   ogg_stream_state stream;
 };

 struct _GstOggAviParseClass
 {
   GstElementClass parent_class;
 };

 static void gst_ogg_avi_parse_base_init (gpointer g_class);
 static void gst_ogg_avi_parse_class_init (GstOggAviParseClass * klass);
 static void gst_ogg_avi_parse_init (GstOggAviParse * ogg);
 static GstElementClass *parent_class = NULL;

 static GType
 gst_ogg_avi_parse_get_type (void)
 {
   static GType ogg_avi_parse_type = 0;

   if (!ogg_avi_parse_type) {
     static const GTypeInfo ogg_avi_parse_info = {
       sizeof (GstOggAviParseClass),
       gst_ogg_avi_parse_base_init,
       NULL,
       (GClassInitFunc) gst_ogg_avi_parse_class_init,
       NULL,
       NULL,
       sizeof (GstOggAviParse),
       0,
       (GInstanceInitFunc) gst_ogg_avi_parse_init,
     };

     ogg_avi_parse_type =
         g_type_register_static (GST_TYPE_ELEMENT, "GstOggAviParse",
         &ogg_avi_parse_info, 0);
   }
   return ogg_avi_parse_type;
 }

 enum
 {
   PROP_0
 };

 static GstStaticPadTemplate ogg_avi_parse_src_template_factory =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-vorbis")
     );

 static GstStaticPadTemplate ogg_avi_parse_sink_template_factory =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-ogg-avi")
     );

 static void gst_ogg_avi_parse_finalize (GObject * object);
 static GstStateChangeReturn gst_ogg_avi_parse_change_state (GstElement *
     element, GstStateChange transition);
 static gboolean gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent,
     GstEvent * event);
 static GstFlowReturn gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent,
     GstBuffer * buffer);
 static gboolean gst_ogg_avi_parse_setcaps (GstPad * pad, GstCaps * caps);

 static void
 gst_ogg_avi_parse_base_init (gpointer g_class)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

   gst_element_class_set_static_metadata (element_class,
       "Ogg AVI parser", "Codec/Parser",
       "parse an ogg avi stream into pages (info about ogg: http://xiph.org)",
       "Wim Taymans <wim@fluendo.com>");

   gst_element_class_add_static_pad_template (element_class,
       &ogg_avi_parse_sink_template_factory);
   gst_element_class_add_static_pad_template (element_class,
       &ogg_avi_parse_src_template_factory);
 }

 static void
 gst_ogg_avi_parse_class_init (GstOggAviParseClass * klass)
 {
   GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
   GObjectClass *gobject_class = G_OBJECT_CLASS (klass);

   parent_class = g_type_class_peek_parent (klass);

   gstelement_class->change_state = gst_ogg_avi_parse_change_state;

   gobject_class->finalize = gst_ogg_avi_parse_finalize;
 }

 static void
 gst_ogg_avi_parse_init (GstOggAviParse * ogg)
 {
   /* create the sink and source pads */
   ogg->sinkpad =
       gst_pad_new_from_static_template (&ogg_avi_parse_sink_template_factory,
       "sink");
   gst_pad_set_event_function (ogg->sinkpad, gst_ogg_avi_parse_event);
   gst_pad_set_chain_function (ogg->sinkpad, gst_ogg_avi_parse_chain);
   gst_element_add_pad (GST_ELEMENT (ogg), ogg->sinkpad);

   ogg->srcpad =
       gst_pad_new_from_static_template (&ogg_avi_parse_src_template_factory,
       "src");
   gst_pad_use_fixed_caps (ogg->srcpad);
   gst_element_add_pad (GST_ELEMENT (ogg), ogg->srcpad);
 }

 static void
 gst_ogg_avi_parse_finalize (GObject * object)
 {
   GstOggAviParse *ogg = GST_OGG_AVI_PARSE (object);

   GST_LOG_OBJECT (ogg, "Disposing of object %p", ogg);

   ogg_sync_clear (&ogg->sync);
   ogg_stream_clear (&ogg->stream);

   G_OBJECT_CLASS (parent_class)->finalize (object);
 }

 static gboolean
 gst_ogg_avi_parse_setcaps (GstPad * pad, GstCaps * caps)
 {
   GstOggAviParse *ogg;
   GstStructure *structure;
   const GValue *codec_data;
   GstBuffer *buffer;
   GstMapInfo map;
   guint8 *ptr;
   gsize left;
   guint32 sizes[3];
   GstCaps *outcaps;
   gint i, offs;

   ogg = GST_OGG_AVI_PARSE (GST_OBJECT_PARENT (pad));

   structure = gst_caps_get_structure (caps, 0);

   /* take codec data */
   codec_data = gst_structure_get_value (structure, "codec_data");
   if (codec_data == NULL)
     goto no_data;

   /* only buffers are valid */
   if (G_VALUE_TYPE (codec_data) != GST_TYPE_BUFFER)
     goto wrong_format;

   /* Now parse the data */
   buffer = gst_value_get_buffer (codec_data);

   /* first 22 bytes are bits_per_sample, channel_mask, GUID
    * Then we get 3 LE guint32 with the 3 header sizes
    * then we get the bytes of the 3 headers. */
   gst_buffer_map (buffer, &map, GST_MAP_READ);

   ptr = map.data;
   left = map.size;

   GST_LOG_OBJECT (ogg, "configuring codec_data of size %" G_GSIZE_FORMAT, left);

   /* skip headers */
   ptr += 22;
   left -= 22;

   /* we need at least 12 bytes for the packet sizes of the 3 headers */
   if (left < 12)
     goto buffer_too_small;

   /* read sizes of the 3 headers */
   sizes[0] = GST_READ_UINT32_LE (ptr);
   sizes[1] = GST_READ_UINT32_LE (ptr + 4);
   sizes[2] = GST_READ_UINT32_LE (ptr + 8);

   GST_DEBUG_OBJECT (ogg, "header sizes: %u %u %u", sizes[0], sizes[1],
       sizes[2]);

   left -= 12;

   /* and we need at least enough data for all the headers */
   if (left < sizes[0] + sizes[1] + sizes[2])
     goto buffer_too_small;

   /* set caps */
   outcaps = gst_caps_new_empty_simple ("audio/x-vorbis");
   gst_pad_set_caps (ogg->srcpad, outcaps);
   gst_caps_unref (outcaps);

   /* copy header data */
   offs = 34;
   for (i = 0; i < 3; i++) {
     GstBuffer *out;

     /* now output the raw vorbis header packets */
     out = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offs, sizes[i]);
     gst_pad_push (ogg->srcpad, out);

     offs += sizes[i];
   }
   gst_buffer_unmap (buffer, &map);

   return TRUE;

   /* ERRORS */
 no_data:
   {
     GST_DEBUG_OBJECT (ogg, "no codec_data found in caps");
     return FALSE;
   }
 wrong_format:
   {
     GST_DEBUG_OBJECT (ogg, "codec_data is not a buffer");
     return FALSE;
   }
 buffer_too_small:
   {
     GST_DEBUG_OBJECT (ogg, "codec_data is too small");
     gst_buffer_unmap (buffer, &map);
     return FALSE;
   }
 }

 static gboolean
 gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent, GstEvent * event)
 {
   GstOggAviParse *ogg;
   gboolean ret;

   ogg = GST_OGG_AVI_PARSE (parent);

   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_CAPS:
     {
       GstCaps *caps;

       gst_event_parse_caps (event, &caps);
       ret = gst_ogg_avi_parse_setcaps (pad, caps);
       gst_event_unref (event);
       break;
     }
     case GST_EVENT_FLUSH_START:
       ret = gst_pad_push_event (ogg->srcpad, event);
       break;
     case GST_EVENT_FLUSH_STOP:
       ogg_sync_reset (&ogg->sync);
       ogg_stream_reset (&ogg->stream);
       ogg->discont = TRUE;
       ret = gst_pad_push_event (ogg->srcpad, event);
       break;
     default:
       ret = gst_pad_push_event (ogg->srcpad, event);
       break;
   }
   return ret;
 }

 static GstFlowReturn
 gst_ogg_avi_parse_push_packet (GstOggAviParse * ogg, ogg_packet * packet)
 {
   GstBuffer *buffer;
   GstFlowReturn result;

   /* allocate space for header and body */
   buffer = gst_buffer_new_and_alloc (packet->bytes);
   gst_buffer_fill (buffer, 0, packet->packet, packet->bytes);

   GST_LOG_OBJECT (ogg, "created buffer %p from page", buffer);

   GST_BUFFER_OFFSET_END (buffer) = packet->granulepos;

   if (ogg->discont) {
     GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
     ogg->discont = FALSE;
   }

   result = gst_pad_push (ogg->srcpad, buffer);

   return result;
 }

 static GstFlowReturn
 gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
 {
   GstFlowReturn result = GST_FLOW_OK;
   GstOggAviParse *ogg;
   guint size;
   gchar *oggbuf;
   gint ret = -1;

   ogg = GST_OGG_AVI_PARSE (parent);

   size = gst_buffer_get_size (buffer);

   GST_LOG_OBJECT (ogg, "Chain function received buffer of size %d", size);

   if (GST_BUFFER_IS_DISCONT (buffer)) {
     ogg_sync_reset (&ogg->sync);
     ogg->discont = TRUE;
   }

   /* write data to sync layer */
   oggbuf = ogg_sync_buffer (&ogg->sync, size);
   gst_buffer_extract (buffer, 0, oggbuf, size);
   ogg_sync_wrote (&ogg->sync, size);
   gst_buffer_unref (buffer);

   /* try to get as many packets out of the stream as possible */
   do {
     ogg_page page;

     /* try to swap out a page */
     ret = ogg_sync_pageout (&ogg->sync, &page);
     if (ret == 0) {
       GST_DEBUG_OBJECT (ogg, "need more data");
       break;
     } else if (ret == -1) {
       GST_DEBUG_OBJECT (ogg, "discont in pages");
       ogg->discont = TRUE;
     } else {
       /* new unknown stream, init the ogg stream with the serial number of the
        * page. */
       if (ogg->serial == -1) {
         ogg->serial = ogg_page_serialno (&page);
         ogg_stream_init (&ogg->stream, ogg->serial);
       }

       /* submit page */
       if (ogg_stream_pagein (&ogg->stream, &page) != 0) {
         GST_WARNING_OBJECT (ogg, "ogg stream choked on page resetting stream");
         ogg_sync_reset (&ogg->sync);
         ogg->discont = TRUE;
         continue;
       }

       /* try to get as many packets as possible out of the page */
       do {
         ogg_packet packet;

         ret = ogg_stream_packetout (&ogg->stream, &packet);
         GST_LOG_OBJECT (ogg, "packetout gave %d", ret);
         switch (ret) {
           case 0:
             break;
           case -1:
             /* out of sync, We mark a DISCONT. */
             ogg->discont = TRUE;
             break;
           case 1:
             result = gst_ogg_avi_parse_push_packet (ogg, &packet);
             if (result != GST_FLOW_OK)
               goto done;
             break;
           default:
             GST_WARNING_OBJECT (ogg,
                 "invalid return value %d for ogg_stream_packetout, resetting stream",
                 ret);
             break;
         }
       }
       while (ret != 0);
     }
   }
   while (ret != 0);

 done:
   return result;
 }

 static GstStateChangeReturn
 gst_ogg_avi_parse_change_state (GstElement * element, GstStateChange transition)
 {
   GstOggAviParse *ogg;
   GstStateChangeReturn result = GST_STATE_CHANGE_FAILURE;

   ogg = GST_OGG_AVI_PARSE (element);

   switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
       ogg_sync_init (&ogg->sync);
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       ogg_sync_reset (&ogg->sync);
       ogg_stream_reset (&ogg->stream);
       ogg->serial = -1;
       ogg->discont = TRUE;
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
     default:
       break;
   }

   result = parent_class->change_state (element, transition);

   switch (transition) {
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       ogg_sync_clear (&ogg->sync);
       break;
     default:
       break;
   }
   return result;
 }

 gboolean
 gst_ogg_avi_parse_plugin_init (GstPlugin * plugin)
 {
   GST_DEBUG_CATEGORY_INIT (gst_ogg_avi_parse_debug, "oggaviparse", 0,
       "ogg avi parser");

   return gst_element_register (plugin, "oggaviparse", GST_RANK_PRIMARY,
       GST_TYPE_OGG_AVI_PARSE);
 }
	/* GStreamer
	* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
	*
	* gstoggaviparse.c: ogg avi stream parser
	*
	* 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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, USA.
	*/

	/*
	* Ogg in AVI is mostly done for vorbis audio. In the codec_data we receive the
	* first 3 packets of the raw vorbis data. On the sinkpad we receive full-blown Ogg
	* pages.
	* Before extracting the packets out of the ogg pages, we push the raw vorbis
	* header packets to the decoder.
	* We don't use the incomming timestamps but use the ganulepos on the ogg pages
	* directly.
	* This parser only does ogg/vorbis for now.
	*/

	#ifdef HAVE_CONFIG_H
	#include "config.h"
	#endif
	#include <gst/gst.h>
	#include <ogg/ogg.h>
	#include <string.h>

	#include "gstogg.h"

	GST_DEBUG_CATEGORY_STATIC (gst_ogg_avi_parse_debug);
	#define GST_CAT_DEFAULT gst_ogg_avi_parse_debug

	#define GST_TYPE_OGG_AVI_PARSE (gst_ogg_avi_parse_get_type())
	#define GST_OGG_AVI_PARSE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OGG_AVI_PARSE, GstOggAviParse))
	#define GST_OGG_AVI_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OGG_AVI_PARSE, GstOggAviParse))
	#define GST_IS_OGG_AVI_PARSE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OGG_AVI_PARSE))
	#define GST_IS_OGG_AVI_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OGG_AVI_PARSE))

	static GType gst_ogg_avi_parse_get_type (void);

	typedef struct _GstOggAviParse GstOggAviParse;
	typedef struct _GstOggAviParseClass GstOggAviParseClass;

	struct _GstOggAviParse
	{
	GstElement element;

	GstPad *sinkpad;
	GstPad *srcpad;

	gboolean discont;
	gint serial;

	ogg_sync_state sync;
	ogg_stream_state stream;
	};

	struct _GstOggAviParseClass
	{
	GstElementClass parent_class;
	};

	static void gst_ogg_avi_parse_base_init (gpointer g_class);
	static void gst_ogg_avi_parse_class_init (GstOggAviParseClass * klass);
	static void gst_ogg_avi_parse_init (GstOggAviParse * ogg);
	static GstElementClass *parent_class = NULL;

	static GType
	gst_ogg_avi_parse_get_type (void)
	{
	static GType ogg_avi_parse_type = 0;

	if (!ogg_avi_parse_type) {
	static const GTypeInfo ogg_avi_parse_info = {
	sizeof (GstOggAviParseClass),
	gst_ogg_avi_parse_base_init,
	NULL,
	(GClassInitFunc) gst_ogg_avi_parse_class_init,
	NULL,
	NULL,
	sizeof (GstOggAviParse),
	0,
	(GInstanceInitFunc) gst_ogg_avi_parse_init,
	};

	ogg_avi_parse_type =
	g_type_register_static (GST_TYPE_ELEMENT, "GstOggAviParse",
	&ogg_avi_parse_info, 0);
	}
	return ogg_avi_parse_type;
	}

	enum
	{
	PROP_0
	};

	static GstStaticPadTemplate ogg_avi_parse_src_template_factory =
	GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/x-vorbis")
	);

	static GstStaticPadTemplate ogg_avi_parse_sink_template_factory =
	GST_STATIC_PAD_TEMPLATE ("sink",
	GST_PAD_SINK,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("application/x-ogg-avi")
	);

	static void gst_ogg_avi_parse_finalize (GObject * object);
	static GstStateChangeReturn gst_ogg_avi_parse_change_state (GstElement *
	element, GstStateChange transition);
	static gboolean gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent,
	GstEvent * event);
	static GstFlowReturn gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent,
	GstBuffer * buffer);
	static gboolean gst_ogg_avi_parse_setcaps (GstPad * pad, GstCaps * caps);

	static void
	gst_ogg_avi_parse_base_init (gpointer g_class)
	{
	GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

	gst_element_class_set_static_metadata (element_class,
	"Ogg AVI parser", "Codec/Parser",
	"parse an ogg avi stream into pages (info about ogg: http://xiph.org)",
	"Wim Taymans <wim@fluendo.com>");

	gst_element_class_add_static_pad_template (element_class,
	&ogg_avi_parse_sink_template_factory);
	gst_element_class_add_static_pad_template (element_class,
	&ogg_avi_parse_src_template_factory);
	}

	static void
	gst_ogg_avi_parse_class_init (GstOggAviParseClass * klass)
	{
	GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
	GObjectClass *gobject_class = G_OBJECT_CLASS (klass);

	parent_class = g_type_class_peek_parent (klass);

	gstelement_class->change_state = gst_ogg_avi_parse_change_state;

	gobject_class->finalize = gst_ogg_avi_parse_finalize;
	}

	static void
	gst_ogg_avi_parse_init (GstOggAviParse * ogg)
	{
	/* create the sink and source pads */
	ogg->sinkpad =
	gst_pad_new_from_static_template (&ogg_avi_parse_sink_template_factory,
	"sink");
	gst_pad_set_event_function (ogg->sinkpad, gst_ogg_avi_parse_event);
	gst_pad_set_chain_function (ogg->sinkpad, gst_ogg_avi_parse_chain);
	gst_element_add_pad (GST_ELEMENT (ogg), ogg->sinkpad);

	ogg->srcpad =
	gst_pad_new_from_static_template (&ogg_avi_parse_src_template_factory,
	"src");
	gst_pad_use_fixed_caps (ogg->srcpad);
	gst_element_add_pad (GST_ELEMENT (ogg), ogg->srcpad);
	}

	static void
	gst_ogg_avi_parse_finalize (GObject * object)
	{
	GstOggAviParse *ogg = GST_OGG_AVI_PARSE (object);

	GST_LOG_OBJECT (ogg, "Disposing of object %p", ogg);

	ogg_sync_clear (&ogg->sync);
	ogg_stream_clear (&ogg->stream);

	G_OBJECT_CLASS (parent_class)->finalize (object);
	}

	static gboolean
	gst_ogg_avi_parse_setcaps (GstPad * pad, GstCaps * caps)
	{
	GstOggAviParse *ogg;
	GstStructure *structure;
	const GValue *codec_data;
	GstBuffer *buffer;
	GstMapInfo map;
	guint8 *ptr;
	gsize left;
	guint32 sizes[3];
	GstCaps *outcaps;
	gint i, offs;

	ogg = GST_OGG_AVI_PARSE (GST_OBJECT_PARENT (pad));

	structure = gst_caps_get_structure (caps, 0);

	/* take codec data */
	codec_data = gst_structure_get_value (structure, "codec_data");
	if (codec_data == NULL)
	goto no_data;

	/* only buffers are valid */
	if (G_VALUE_TYPE (codec_data) != GST_TYPE_BUFFER)
	goto wrong_format;

	/* Now parse the data */
	buffer = gst_value_get_buffer (codec_data);

	/* first 22 bytes are bits_per_sample, channel_mask, GUID
	* Then we get 3 LE guint32 with the 3 header sizes
	* then we get the bytes of the 3 headers. */
	gst_buffer_map (buffer, &map, GST_MAP_READ);

	ptr = map.data;
	left = map.size;

	GST_LOG_OBJECT (ogg, "configuring codec_data of size %" G_GSIZE_FORMAT, left);

	/* skip headers */
	ptr += 22;
	left -= 22;

	/* we need at least 12 bytes for the packet sizes of the 3 headers */
	if (left < 12)
	goto buffer_too_small;

	/* read sizes of the 3 headers */
	sizes[0] = GST_READ_UINT32_LE (ptr);
	sizes[1] = GST_READ_UINT32_LE (ptr + 4);
	sizes[2] = GST_READ_UINT32_LE (ptr + 8);

	GST_DEBUG_OBJECT (ogg, "header sizes: %u %u %u", sizes[0], sizes[1],
	sizes[2]);

	left -= 12;

	/* and we need at least enough data for all the headers */
	if (left < sizes[0] + sizes[1] + sizes[2])
	goto buffer_too_small;

	/* set caps */
	outcaps = gst_caps_new_empty_simple ("audio/x-vorbis");
	gst_pad_set_caps (ogg->srcpad, outcaps);
	gst_caps_unref (outcaps);

	/* copy header data */
	offs = 34;
	for (i = 0; i < 3; i++) {
	GstBuffer *out;

	/* now output the raw vorbis header packets */
	out = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offs, sizes[i]);
	gst_pad_push (ogg->srcpad, out);

	offs += sizes[i];
	}
	gst_buffer_unmap (buffer, &map);

	return TRUE;

	/* ERRORS */
	no_data:
	{
	GST_DEBUG_OBJECT (ogg, "no codec_data found in caps");
	return FALSE;
	}
	wrong_format:
	{
	GST_DEBUG_OBJECT (ogg, "codec_data is not a buffer");
	return FALSE;
	}
	buffer_too_small:
	{
	GST_DEBUG_OBJECT (ogg, "codec_data is too small");
	gst_buffer_unmap (buffer, &map);
	return FALSE;
	}
	}

	static gboolean
	gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent, GstEvent * event)
	{
	GstOggAviParse *ogg;
	gboolean ret;

	ogg = GST_OGG_AVI_PARSE (parent);

	switch (GST_EVENT_TYPE (event)) {
	case GST_EVENT_CAPS:
	{
	GstCaps *caps;

	gst_event_parse_caps (event, &caps);
	ret = gst_ogg_avi_parse_setcaps (pad, caps);
	gst_event_unref (event);
	break;
	}
	case GST_EVENT_FLUSH_START:
	ret = gst_pad_push_event (ogg->srcpad, event);
	break;
	case GST_EVENT_FLUSH_STOP:
	ogg_sync_reset (&ogg->sync);
	ogg_stream_reset (&ogg->stream);
	ogg->discont = TRUE;
	ret = gst_pad_push_event (ogg->srcpad, event);
	break;
	default:
	ret = gst_pad_push_event (ogg->srcpad, event);
	break;
	}
	return ret;
	}

	static GstFlowReturn
	gst_ogg_avi_parse_push_packet (GstOggAviParse * ogg, ogg_packet * packet)
	{
	GstBuffer *buffer;
	GstFlowReturn result;

	/* allocate space for header and body */
	buffer = gst_buffer_new_and_alloc (packet->bytes);
	gst_buffer_fill (buffer, 0, packet->packet, packet->bytes);

	GST_LOG_OBJECT (ogg, "created buffer %p from page", buffer);

	GST_BUFFER_OFFSET_END (buffer) = packet->granulepos;

	if (ogg->discont) {
	GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
	ogg->discont = FALSE;
	}

	result = gst_pad_push (ogg->srcpad, buffer);

	return result;
	}

	static GstFlowReturn
	gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
	{
	GstFlowReturn result = GST_FLOW_OK;
	GstOggAviParse *ogg;
	guint size;
	gchar *oggbuf;
	gint ret = -1;

	ogg = GST_OGG_AVI_PARSE (parent);

	size = gst_buffer_get_size (buffer);

	GST_LOG_OBJECT (ogg, "Chain function received buffer of size %d", size);

	if (GST_BUFFER_IS_DISCONT (buffer)) {
	ogg_sync_reset (&ogg->sync);
	ogg->discont = TRUE;
	}

	/* write data to sync layer */
	oggbuf = ogg_sync_buffer (&ogg->sync, size);
	gst_buffer_extract (buffer, 0, oggbuf, size);
	ogg_sync_wrote (&ogg->sync, size);
	gst_buffer_unref (buffer);

	/* try to get as many packets out of the stream as possible */
	do {
	ogg_page page;

	/* try to swap out a page */
	ret = ogg_sync_pageout (&ogg->sync, &page);
	if (ret == 0) {
	GST_DEBUG_OBJECT (ogg, "need more data");
	break;
	} else if (ret == -1) {
	GST_DEBUG_OBJECT (ogg, "discont in pages");
	ogg->discont = TRUE;
	} else {
	/* new unknown stream, init the ogg stream with the serial number of the
	* page. */
	if (ogg->serial == -1) {
	ogg->serial = ogg_page_serialno (&page);
	ogg_stream_init (&ogg->stream, ogg->serial);
	}

	/* submit page */
	if (ogg_stream_pagein (&ogg->stream, &page) != 0) {
	GST_WARNING_OBJECT (ogg, "ogg stream choked on page resetting stream");
	ogg_sync_reset (&ogg->sync);
	ogg->discont = TRUE;
	continue;
	}

	/* try to get as many packets as possible out of the page */
	do {
	ogg_packet packet;

	ret = ogg_stream_packetout (&ogg->stream, &packet);
	GST_LOG_OBJECT (ogg, "packetout gave %d", ret);
	switch (ret) {
	case 0:
	break;
	case -1:
	/* out of sync, We mark a DISCONT. */
	ogg->discont = TRUE;
	break;
	case 1:
	result = gst_ogg_avi_parse_push_packet (ogg, &packet);
	if (result != GST_FLOW_OK)
	goto done;
	break;
	default:
	GST_WARNING_OBJECT (ogg,
	"invalid return value %d for ogg_stream_packetout, resetting stream",
	ret);
	break;
	}
	}
	while (ret != 0);
	}
	}
	while (ret != 0);

	done:
	return result;
	}

	static GstStateChangeReturn
	gst_ogg_avi_parse_change_state (GstElement * element, GstStateChange transition)
	{
	GstOggAviParse *ogg;
	GstStateChangeReturn result = GST_STATE_CHANGE_FAILURE;

	ogg = GST_OGG_AVI_PARSE (element);

	switch (transition) {
	case GST_STATE_CHANGE_NULL_TO_READY:
	ogg_sync_init (&ogg->sync);
	break;
	case GST_STATE_CHANGE_READY_TO_PAUSED:
	ogg_sync_reset (&ogg->sync);
	ogg_stream_reset (&ogg->stream);
	ogg->serial = -1;
	ogg->discont = TRUE;
	break;
	case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
	break;
	default:
	break;
	}

	result = parent_class->change_state (element, transition);

	switch (transition) {
	case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
	break;
	case GST_STATE_CHANGE_PAUSED_TO_READY:
	break;
	case GST_STATE_CHANGE_READY_TO_NULL:
	ogg_sync_clear (&ogg->sync);
	break;
	default:
	break;
	}
	return result;
	}

	gboolean
	gst_ogg_avi_parse_plugin_init (GstPlugin * plugin)
	{
	GST_DEBUG_CATEGORY_INIT (gst_ogg_avi_parse_debug, "oggaviparse", 0,
	"ogg avi parser");

	return gst_element_register (plugin, "oggaviparse", GST_RANK_PRIMARY,
	GST_TYPE_OGG_AVI_PARSE);
	}