gst/rtp/gstrtpmpapay.c - imx-gst-plugins-good - Git at Google

 /* GStreamer
  * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */

 #ifdef HAVE_CONFIG_H
 #  include "config.h"
 #endif

 #include <string.h>

 #include <gst/rtp/gstrtpbuffer.h>
 #include <gst/audio/audio.h>

 #include "gstrtpmpapay.h"
 #include "gstrtputils.h"

 GST_DEBUG_CATEGORY_STATIC (rtpmpapay_debug);
 #define GST_CAT_DEFAULT (rtpmpapay_debug)

 static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
     );

 static GstStaticPadTemplate gst_rtp_mpa_pay_src_template =
     GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
         "payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", "
         "clock-rate = (int) 90000; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
         "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
         "clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
     );

 static void gst_rtp_mpa_pay_finalize (GObject * object);

 static GstStateChangeReturn gst_rtp_mpa_pay_change_state (GstElement * element,
     GstStateChange transition);

 static gboolean gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload,
     GstCaps * caps);
 static gboolean gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload,
     GstEvent * event);
 static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay);
 static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * payload,
     GstBuffer * buffer);

 #define gst_rtp_mpa_pay_parent_class parent_class
 G_DEFINE_TYPE (GstRtpMPAPay, gst_rtp_mpa_pay, GST_TYPE_RTP_BASE_PAYLOAD);

 static void
 gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
   GstRTPBasePayloadClass *gstrtpbasepayload_class;

   GST_DEBUG_CATEGORY_INIT (rtpmpapay_debug, "rtpmpapay", 0,
       "MPEG Audio RTP Depayloader");

   gobject_class = (GObjectClass *) klass;
   gstelement_class = (GstElementClass *) klass;
   gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

   gobject_class->finalize = gst_rtp_mpa_pay_finalize;

   gstelement_class->change_state = gst_rtp_mpa_pay_change_state;

   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_mpa_pay_src_template);
   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_mpa_pay_sink_template);

   gst_element_class_set_static_metadata (gstelement_class,
       "RTP MPEG audio payloader", "Codec/Payloader/Network/RTP",
       "Payload MPEG audio as RTP packets (RFC 2038)",
       "Wim Taymans <wim.taymans@gmail.com>");

   gstrtpbasepayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
   gstrtpbasepayload_class->sink_event = gst_rtp_mpa_pay_sink_event;
   gstrtpbasepayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
 }

 static void
 gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
 {
   rtpmpapay->adapter = gst_adapter_new ();

   GST_RTP_BASE_PAYLOAD (rtpmpapay)->pt = GST_RTP_PAYLOAD_MPA;
 }

 static void
 gst_rtp_mpa_pay_finalize (GObject * object)
 {
   GstRtpMPAPay *rtpmpapay;

   rtpmpapay = GST_RTP_MPA_PAY (object);

   g_object_unref (rtpmpapay->adapter);
   rtpmpapay->adapter = NULL;

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

 static void
 gst_rtp_mpa_pay_reset (GstRtpMPAPay * pay)
 {
   pay->first_ts = -1;
   pay->duration = 0;
   gst_adapter_clear (pay->adapter);
   GST_DEBUG_OBJECT (pay, "reset depayloader");
 }

 static gboolean
 gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;

   gst_rtp_base_payload_set_options (payload, "audio",
       payload->pt != GST_RTP_PAYLOAD_MPA, "MPA", 90000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);

   return res;
 }

 static gboolean
 gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
 {
   gboolean ret;
   GstRtpMPAPay *rtpmpapay;

   rtpmpapay = GST_RTP_MPA_PAY (payload);

   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_EOS:
       /* make sure we push the last packets in the adapter on EOS */
       gst_rtp_mpa_pay_flush (rtpmpapay);
       break;
     case GST_EVENT_FLUSH_STOP:
       gst_rtp_mpa_pay_reset (rtpmpapay);
       break;
     default:
       break;
   }

   ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);

   return ret;
 }

 #define RTP_HEADER_LEN 12

 static GstFlowReturn
 gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
 {
   guint avail;
   GstBuffer *outbuf;
   GstFlowReturn ret;
   guint16 frag_offset;
   GstBufferList *list;

   /* the data available in the adapter is either smaller
    * than the MTU or bigger. In the case it is smaller, the complete
    * adapter contents can be put in one packet. In the case the
    * adapter has more than one MTU, we need to split the MPA data
    * over multiple packets. The frag_offset in each packet header
    * needs to be updated with the position in the MPA frame. */
   avail = gst_adapter_available (rtpmpapay->adapter);

   ret = GST_FLOW_OK;

   list =
       gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay) -
           RTP_HEADER_LEN) + 1);

   frag_offset = 0;
   while (avail > 0) {
     guint towrite;
     guint8 *payload;
     guint payload_len;
     guint packet_len;
     GstRTPBuffer rtp = { NULL };
     GstBuffer *paybuf;

     /* this will be the total length of the packet */
     packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);

     /* fill one MTU or all available bytes */
     towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay));

     /* this is the payload length */
     payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);

     /* create buffer to hold the payload */
     outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);

     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);

     payload_len -= 4;

     gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_PAYLOAD_MPA);

     /*
      *  0                   1                   2                   3
      *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      * |             MBZ               |          Frag_offset          |
      * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      */
     payload = gst_rtp_buffer_get_payload (&rtp);
     payload[0] = 0;
     payload[1] = 0;
     payload[2] = frag_offset >> 8;
     payload[3] = frag_offset & 0xff;

     avail -= payload_len;
     frag_offset += payload_len;

     if (avail == 0)
       gst_rtp_buffer_set_marker (&rtp, TRUE);

     gst_rtp_buffer_unmap (&rtp);

     paybuf = gst_adapter_take_buffer_fast (rtpmpapay->adapter, payload_len);
     gst_rtp_copy_audio_meta (rtpmpapay, outbuf, paybuf);
     outbuf = gst_buffer_append (outbuf, paybuf);

     GST_BUFFER_PTS (outbuf) = rtpmpapay->first_ts;
     GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
     gst_buffer_list_add (list, outbuf);
   }

   ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpapay), list);

   return ret;
 }

 static GstFlowReturn
 gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * basepayload,
     GstBuffer * buffer)
 {
   GstRtpMPAPay *rtpmpapay;
   GstFlowReturn ret;
   guint size, avail;
   guint packet_len;
   GstClockTime duration, timestamp;

   rtpmpapay = GST_RTP_MPA_PAY (basepayload);

   size = gst_buffer_get_size (buffer);
   duration = GST_BUFFER_DURATION (buffer);
   timestamp = GST_BUFFER_PTS (buffer);

   if (GST_BUFFER_IS_DISCONT (buffer)) {
     GST_DEBUG_OBJECT (rtpmpapay, "DISCONT");
     gst_rtp_mpa_pay_reset (rtpmpapay);
   }

   avail = gst_adapter_available (rtpmpapay->adapter);

   /* get packet length of previous data and this new data,
    * payload length includes a 4 byte header */
   packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);

   /* if this buffer is going to overflow the packet, flush what we
    * have. */
   if (gst_rtp_base_payload_is_filled (basepayload,
           packet_len, rtpmpapay->duration + duration)) {
     ret = gst_rtp_mpa_pay_flush (rtpmpapay);
     avail = 0;
   } else {
     ret = GST_FLOW_OK;
   }

   if (avail == 0) {
     GST_DEBUG_OBJECT (rtpmpapay,
         "first packet, save timestamp %" GST_TIME_FORMAT,
         GST_TIME_ARGS (timestamp));
     rtpmpapay->first_ts = timestamp;
     rtpmpapay->duration = 0;
   }

   gst_adapter_push (rtpmpapay->adapter, buffer);
   rtpmpapay->duration = duration;

   return ret;
 }

 static GstStateChangeReturn
 gst_rtp_mpa_pay_change_state (GstElement * element, GstStateChange transition)
 {
   GstRtpMPAPay *rtpmpapay;
   GstStateChangeReturn ret;

   rtpmpapay = GST_RTP_MPA_PAY (element);

   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       gst_rtp_mpa_pay_reset (rtpmpapay);
       break;
     default:
       break;
   }

   ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

   switch (transition) {
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       gst_rtp_mpa_pay_reset (rtpmpapay);
       break;
     default:
       break;
   }
   return ret;
 }

 gboolean
 gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin)
 {
   return gst_element_register (plugin, "rtpmpapay",
       GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_PAY);
 }
	/* GStreamer
	* Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, USA.
	*/

	#ifdef HAVE_CONFIG_H
	# include "config.h"
	#endif

	#include <string.h>

	#include <gst/rtp/gstrtpbuffer.h>
	#include <gst/audio/audio.h>

	#include "gstrtpmpapay.h"
	#include "gstrtputils.h"

	GST_DEBUG_CATEGORY_STATIC (rtpmpapay_debug);
	#define GST_CAT_DEFAULT (rtpmpapay_debug)

	static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template =
	GST_STATIC_PAD_TEMPLATE ("sink",
	GST_PAD_SINK,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
	);

	static GstStaticPadTemplate gst_rtp_mpa_pay_src_template =
	GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("application/x-rtp, "
	"media = (string) \"audio\", "
	"payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", "
	"clock-rate = (int) 90000; "
	"application/x-rtp, "
	"media = (string) \"audio\", "
	"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
	"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
	);

	static void gst_rtp_mpa_pay_finalize (GObject * object);

	static GstStateChangeReturn gst_rtp_mpa_pay_change_state (GstElement * element,
	GstStateChange transition);

	static gboolean gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload,
	GstCaps * caps);
	static gboolean gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload,
	GstEvent * event);
	static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay);
	static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * payload,
	GstBuffer * buffer);

	#define gst_rtp_mpa_pay_parent_class parent_class
	G_DEFINE_TYPE (GstRtpMPAPay, gst_rtp_mpa_pay, GST_TYPE_RTP_BASE_PAYLOAD);

	static void
	gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass)
	{
	GObjectClass *gobject_class;
	GstElementClass *gstelement_class;
	GstRTPBasePayloadClass *gstrtpbasepayload_class;

	GST_DEBUG_CATEGORY_INIT (rtpmpapay_debug, "rtpmpapay", 0,
	"MPEG Audio RTP Depayloader");

	gobject_class = (GObjectClass *) klass;
	gstelement_class = (GstElementClass *) klass;
	gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

	gobject_class->finalize = gst_rtp_mpa_pay_finalize;

	gstelement_class->change_state = gst_rtp_mpa_pay_change_state;

	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_mpa_pay_src_template);
	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_mpa_pay_sink_template);

	gst_element_class_set_static_metadata (gstelement_class,
	"RTP MPEG audio payloader", "Codec/Payloader/Network/RTP",
	"Payload MPEG audio as RTP packets (RFC 2038)",
	"Wim Taymans <wim.taymans@gmail.com>");

	gstrtpbasepayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
	gstrtpbasepayload_class->sink_event = gst_rtp_mpa_pay_sink_event;
	gstrtpbasepayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
	}

	static void
	gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
	{
	rtpmpapay->adapter = gst_adapter_new ();

	GST_RTP_BASE_PAYLOAD (rtpmpapay)->pt = GST_RTP_PAYLOAD_MPA;
	}

	static void
	gst_rtp_mpa_pay_finalize (GObject * object)
	{
	GstRtpMPAPay *rtpmpapay;

	rtpmpapay = GST_RTP_MPA_PAY (object);

	g_object_unref (rtpmpapay->adapter);
	rtpmpapay->adapter = NULL;

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

	static void
	gst_rtp_mpa_pay_reset (GstRtpMPAPay * pay)
	{
	pay->first_ts = -1;
	pay->duration = 0;
	gst_adapter_clear (pay->adapter);
	GST_DEBUG_OBJECT (pay, "reset depayloader");
	}

	static gboolean
	gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
	{
	gboolean res;

	gst_rtp_base_payload_set_options (payload, "audio",
	payload->pt != GST_RTP_PAYLOAD_MPA, "MPA", 90000);
	res = gst_rtp_base_payload_set_outcaps (payload, NULL);

	return res;
	}

	static gboolean
	gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
	{
	gboolean ret;
	GstRtpMPAPay *rtpmpapay;

	rtpmpapay = GST_RTP_MPA_PAY (payload);

	switch (GST_EVENT_TYPE (event)) {
	case GST_EVENT_EOS:
	/* make sure we push the last packets in the adapter on EOS */
	gst_rtp_mpa_pay_flush (rtpmpapay);
	break;
	case GST_EVENT_FLUSH_STOP:
	gst_rtp_mpa_pay_reset (rtpmpapay);
	break;
	default:
	break;
	}

	ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);

	return ret;
	}

	#define RTP_HEADER_LEN 12

	static GstFlowReturn
	gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
	{
	guint avail;
	GstBuffer *outbuf;
	GstFlowReturn ret;
	guint16 frag_offset;
	GstBufferList *list;

	/* the data available in the adapter is either smaller
	* than the MTU or bigger. In the case it is smaller, the complete
	* adapter contents can be put in one packet. In the case the
	* adapter has more than one MTU, we need to split the MPA data
	* over multiple packets. The frag_offset in each packet header
	* needs to be updated with the position in the MPA frame. */
	avail = gst_adapter_available (rtpmpapay->adapter);

	ret = GST_FLOW_OK;

	list =
	gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay) -
	RTP_HEADER_LEN) + 1);

	frag_offset = 0;
	while (avail > 0) {
	guint towrite;
	guint8 *payload;
	guint payload_len;
	guint packet_len;
	GstRTPBuffer rtp = { NULL };
	GstBuffer *paybuf;

	/* this will be the total length of the packet */
	packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);

	/* fill one MTU or all available bytes */
	towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay));

	/* this is the payload length */
	payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);

	/* create buffer to hold the payload */
	outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);

	gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);

	payload_len -= 4;

	gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_PAYLOAD_MPA);

	/*
	* 0 1 2 3
	* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
	* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	* \| MBZ \| Frag_offset \|
	* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	*/
	payload = gst_rtp_buffer_get_payload (&rtp);
	payload[0] = 0;
	payload[1] = 0;
	payload[2] = frag_offset >> 8;
	payload[3] = frag_offset & 0xff;

	avail -= payload_len;
	frag_offset += payload_len;

	if (avail == 0)
	gst_rtp_buffer_set_marker (&rtp, TRUE);

	gst_rtp_buffer_unmap (&rtp);

	paybuf = gst_adapter_take_buffer_fast (rtpmpapay->adapter, payload_len);
	gst_rtp_copy_audio_meta (rtpmpapay, outbuf, paybuf);
	outbuf = gst_buffer_append (outbuf, paybuf);

	GST_BUFFER_PTS (outbuf) = rtpmpapay->first_ts;
	GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
	gst_buffer_list_add (list, outbuf);
	}

	ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpapay), list);

	return ret;
	}

	static GstFlowReturn
	gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * basepayload,
	GstBuffer * buffer)
	{
	GstRtpMPAPay *rtpmpapay;
	GstFlowReturn ret;
	guint size, avail;
	guint packet_len;
	GstClockTime duration, timestamp;

	rtpmpapay = GST_RTP_MPA_PAY (basepayload);

	size = gst_buffer_get_size (buffer);
	duration = GST_BUFFER_DURATION (buffer);
	timestamp = GST_BUFFER_PTS (buffer);

	if (GST_BUFFER_IS_DISCONT (buffer)) {
	GST_DEBUG_OBJECT (rtpmpapay, "DISCONT");
	gst_rtp_mpa_pay_reset (rtpmpapay);
	}

	avail = gst_adapter_available (rtpmpapay->adapter);

	/* get packet length of previous data and this new data,
	* payload length includes a 4 byte header */
	packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);

	/* if this buffer is going to overflow the packet, flush what we
	* have. */
	if (gst_rtp_base_payload_is_filled (basepayload,
	packet_len, rtpmpapay->duration + duration)) {
	ret = gst_rtp_mpa_pay_flush (rtpmpapay);
	avail = 0;
	} else {
	ret = GST_FLOW_OK;
	}

	if (avail == 0) {
	GST_DEBUG_OBJECT (rtpmpapay,
	"first packet, save timestamp %" GST_TIME_FORMAT,
	GST_TIME_ARGS (timestamp));
	rtpmpapay->first_ts = timestamp;
	rtpmpapay->duration = 0;
	}

	gst_adapter_push (rtpmpapay->adapter, buffer);
	rtpmpapay->duration = duration;

	return ret;
	}

	static GstStateChangeReturn
	gst_rtp_mpa_pay_change_state (GstElement * element, GstStateChange transition)
	{
	GstRtpMPAPay *rtpmpapay;
	GstStateChangeReturn ret;

	rtpmpapay = GST_RTP_MPA_PAY (element);

	switch (transition) {
	case GST_STATE_CHANGE_READY_TO_PAUSED:
	gst_rtp_mpa_pay_reset (rtpmpapay);
	break;
	default:
	break;
	}

	ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

	switch (transition) {
	case GST_STATE_CHANGE_PAUSED_TO_READY:
	gst_rtp_mpa_pay_reset (rtpmpapay);
	break;
	default:
	break;
	}
	return ret;
	}

	gboolean
	gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin)
	{
	return gst_element_register (plugin, "rtpmpapay",
	GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_PAY);
	}