gst/rtpmanager/gstrtprtxqueue.c - imx-gst-plugins-good - Git at Google

 /* RTP Retransmission queue element for GStreamer
  *
  * gstrtprtxqueue.c:
  *
  * Copyright (C) 2013 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.
  */

 /**
  * SECTION:element-rtprtxqueue
  *
  * rtprtxqueue maintains a queue of transmitted RTP packets, up to a
  * configurable limit (see #GstRTPRtxQueue::max-size-time,
  * #GstRTPRtxQueue::max-size-packets), and retransmits them upon request
  * from the downstream rtpsession (GstRTPRetransmissionRequest event).
  *
  * This element is similar to rtprtxsend, but it has differences:
  * - Retransmission from rtprtxqueue is not RFC 4588 compliant. The
  * retransmitted packets have the same ssrc and payload type as the original
  * stream.
  * - As a side-effect of the above, rtprtxqueue does not require the use of
  * rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to
  * reconstruct the stream.
  * - Retransmission from rtprtxqueue happens as soon as the next regular flow
  * packet is chained, while rtprtxsend retransmits as soon as the retransmission
  * event is received, using a helper thread.
  * - rtprtxqueue can be used with rtpbin without the need of hooking to its
  * #GstRtpBin::request-aux-sender signal, which means it can be used with
  * rtpbin using gst-launch.
  *
  * See also #GstRtpRtxSend, #GstRtpRtxReceive
  *
  * # Example pipelines
  * |[
  * gst-launch-1.0 rtpbin name=b rtp-profile=avpf \
  *    audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \
  *    b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \
  *    udpsrc port=5001 ! b.recv_rtcp_sink_0 \
  *    b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false
  * ]| Sender pipeline
  * |[
  * gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \
  *    udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
  *        b.recv_rtp_sink_0 \
  *    b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
  *    udpsrc port=5002 ! b.recv_rtcp_sink_0 \
  *    b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false
  * ]| Receiver pipeline
  */

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

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

 #include "gstrtprtxqueue.h"

 GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_queue_debug);
 #define GST_CAT_DEFAULT gst_rtp_rtx_queue_debug

 #define DEFAULT_MAX_SIZE_TIME    0
 #define DEFAULT_MAX_SIZE_PACKETS 100

 enum
 {
   PROP_0,
   PROP_MAX_SIZE_TIME,
   PROP_MAX_SIZE_PACKETS,
   PROP_REQUESTS,
   PROP_FULFILLED_REQUESTS,
 };

 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp")
     );

 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp")
     );

 static gboolean gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent,
     GstEvent * event);
 static gboolean gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent,
     GstEvent * event);
 static GstFlowReturn gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent,
     GstBuffer * buffer);
 static GstFlowReturn gst_rtp_rtx_queue_chain_list (GstPad * pad,
     GstObject * parent, GstBufferList * list);

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

 static void gst_rtp_rtx_queue_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
 static void gst_rtp_rtx_queue_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 static void gst_rtp_rtx_queue_finalize (GObject * object);

 G_DEFINE_TYPE (GstRTPRtxQueue, gst_rtp_rtx_queue, GST_TYPE_ELEMENT);

 static void
 gst_rtp_rtx_queue_class_init (GstRTPRtxQueueClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;

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

   gobject_class->get_property = gst_rtp_rtx_queue_get_property;
   gobject_class->set_property = gst_rtp_rtx_queue_set_property;
   gobject_class->finalize = gst_rtp_rtx_queue_finalize;

   g_object_class_install_property (gobject_class, PROP_MAX_SIZE_TIME,
       g_param_spec_uint ("max-size-time", "Max Size Times",
           "Amount of ms to queue (0 = unlimited)", 0, G_MAXUINT,
           DEFAULT_MAX_SIZE_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

   g_object_class_install_property (gobject_class, PROP_MAX_SIZE_PACKETS,
       g_param_spec_uint ("max-size-packets", "Max Size Packets",
           "Amount of packets to queue (0 = unlimited)", 0, G_MAXUINT,
           DEFAULT_MAX_SIZE_PACKETS,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

   g_object_class_install_property (gobject_class, PROP_REQUESTS,
       g_param_spec_uint ("requests", "Requests",
           "Total number of retransmission requests", 0, G_MAXUINT,
           0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

   g_object_class_install_property (gobject_class, PROP_FULFILLED_REQUESTS,
       g_param_spec_uint ("fulfilled-requests", "Fulfilled Requests",
           "Number of fulfilled retransmission requests", 0, G_MAXUINT,
           0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

   gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
   gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);

   gst_element_class_set_static_metadata (gstelement_class,
       "RTP Retransmission Queue", "Codec",
       "Keep RTP packets in a queue for retransmission",
       "Wim Taymans <wim.taymans@gmail.com>");

   gstelement_class->change_state =
       GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_change_state);
 }

 static void
 gst_rtp_rtx_queue_reset (GstRTPRtxQueue * rtx, gboolean full)
 {
   g_mutex_lock (&rtx->lock);
   g_queue_foreach (rtx->queue, (GFunc) gst_buffer_unref, NULL);
   g_queue_clear (rtx->queue);
   g_list_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
   g_list_free (rtx->pending);
   rtx->pending = NULL;
   rtx->n_requests = 0;
   rtx->n_fulfilled_requests = 0;
   g_mutex_unlock (&rtx->lock);
 }

 static void
 gst_rtp_rtx_queue_finalize (GObject * object)
 {
   GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);

   gst_rtp_rtx_queue_reset (rtx, TRUE);
   g_queue_free (rtx->queue);
   g_mutex_clear (&rtx->lock);

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

 static void
 gst_rtp_rtx_queue_init (GstRTPRtxQueue * rtx)
 {
   GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx);

   rtx->srcpad =
       gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
           "src"), "src");
   GST_PAD_SET_PROXY_CAPS (rtx->srcpad);
   GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad);
   gst_pad_set_event_function (rtx->srcpad,
       GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_src_event));
   gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad);

   rtx->sinkpad =
       gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
           "sink"), "sink");
   GST_PAD_SET_PROXY_CAPS (rtx->sinkpad);
   GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
   gst_pad_set_event_function (rtx->sinkpad,
       GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_sink_event));
   gst_pad_set_chain_function (rtx->sinkpad,
       GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain));
   gst_pad_set_chain_list_function (rtx->sinkpad,
       GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain_list));
   gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);

   rtx->queue = g_queue_new ();
   g_mutex_init (&rtx->lock);

   rtx->max_size_time = DEFAULT_MAX_SIZE_TIME;
   rtx->max_size_packets = DEFAULT_MAX_SIZE_PACKETS;
 }

 typedef struct
 {
   GstRTPRtxQueue *rtx;
   guint seqnum;
   gboolean found;
 } RTXData;

 static void
 push_seqnum (GstBuffer * buffer, RTXData * data)
 {
   GstRTPRtxQueue *rtx = data->rtx;
   GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
   guint16 seqnum;

   if (data->found)
     return;

   if (!GST_IS_BUFFER (buffer) ||
       !gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer))
     return;

   seqnum = gst_rtp_buffer_get_seq (&rtpbuffer);
   gst_rtp_buffer_unmap (&rtpbuffer);

   if (seqnum == data->seqnum) {
     data->found = TRUE;
     GST_DEBUG_OBJECT (rtx, "found %d", seqnum);
     rtx->pending = g_list_prepend (rtx->pending, gst_buffer_ref (buffer));
   }
 }

 static gboolean
 gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
 {
   GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent);
   gboolean res;

   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_CUSTOM_UPSTREAM:
     {
       const GstStructure *s;

       s = gst_event_get_structure (event);
       if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
         guint seqnum;
         RTXData data;

         if (!gst_structure_get_uint (s, "seqnum", &seqnum))
           seqnum = -1;

         GST_DEBUG_OBJECT (rtx, "request %d", seqnum);

         g_mutex_lock (&rtx->lock);
         data.rtx = rtx;
         data.seqnum = seqnum;
         data.found = FALSE;
         rtx->n_requests += 1;
         g_queue_foreach (rtx->queue, (GFunc) push_seqnum, &data);
         g_mutex_unlock (&rtx->lock);

         gst_event_unref (event);
         res = TRUE;
       } else {
         res = gst_pad_event_default (pad, parent, event);
       }
       break;
     }
     default:
       res = gst_pad_event_default (pad, parent, event);
       break;
   }
   return res;
 }

 static gboolean
 gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent,
     GstEvent * event)
 {
   GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent);
   gboolean res;

   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_SEGMENT:
     {
       g_mutex_lock (&rtx->lock);
       gst_event_copy_segment (event, &rtx->head_segment);
       g_queue_push_head (rtx->queue, gst_event_ref (event));
       g_mutex_unlock (&rtx->lock);
       /* fall through */
     }
     default:
       res = gst_pad_event_default (pad, parent, event);
       break;
   }
   return res;
 }

 static void
 do_push (GstBuffer * buffer, GstRTPRtxQueue * rtx)
 {
   rtx->n_fulfilled_requests += 1;
   gst_pad_push (rtx->srcpad, buffer);
 }

 static guint32
 get_ts_diff (GstRTPRtxQueue * rtx)
 {
   GstClockTime high_ts, low_ts;
   GstClockTimeDiff result;
   GstBuffer *high_buf, *low_buf;

   high_buf = g_queue_peek_head (rtx->queue);

   while (GST_IS_EVENT ((low_buf = g_queue_peek_tail (rtx->queue)))) {
     GstEvent *event = g_queue_pop_tail (rtx->queue);
     gst_event_copy_segment (event, &rtx->tail_segment);
     gst_event_unref (event);
   }

   if (!high_buf || !low_buf || high_buf == low_buf)
     return 0;

   high_ts = GST_BUFFER_TIMESTAMP (high_buf);
   low_ts = GST_BUFFER_TIMESTAMP (low_buf);

   high_ts = gst_segment_to_running_time (&rtx->head_segment, GST_FORMAT_TIME,
       high_ts);
   low_ts = gst_segment_to_running_time (&rtx->tail_segment, GST_FORMAT_TIME,
       low_ts);

   result = high_ts - low_ts;

   /* return value in ms instead of ns */
   return (guint32) gst_util_uint64_scale_int (result, 1, GST_MSECOND);
 }

 /* Must be called with rtx->lock */
 static void
 shrink_queue (GstRTPRtxQueue * rtx)
 {
   if (rtx->max_size_packets) {
     while (g_queue_get_length (rtx->queue) > rtx->max_size_packets)
       gst_buffer_unref (g_queue_pop_tail (rtx->queue));
   }
   if (rtx->max_size_time) {
     while (get_ts_diff (rtx) > rtx->max_size_time)
       gst_buffer_unref (g_queue_pop_tail (rtx->queue));
   }
 }

 static GstFlowReturn
 gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
 {
   GstRTPRtxQueue *rtx;
   GstFlowReturn ret;
   GList *pending;

   rtx = GST_RTP_RTX_QUEUE (parent);

   g_mutex_lock (&rtx->lock);
   g_queue_push_head (rtx->queue, gst_buffer_ref (buffer));
   shrink_queue (rtx);

   pending = rtx->pending;
   rtx->pending = NULL;
   g_mutex_unlock (&rtx->lock);

   pending = g_list_reverse (pending);
   g_list_foreach (pending, (GFunc) do_push, rtx);
   g_list_free (pending);

   ret = gst_pad_push (rtx->srcpad, buffer);

   return ret;
 }

 static gboolean
 push_to_queue (GstBuffer ** buffer, guint idx, gpointer user_data)
 {
   GQueue *queue = user_data;

   g_queue_push_head (queue, gst_buffer_ref (*buffer));

   return TRUE;
 }

 static GstFlowReturn
 gst_rtp_rtx_queue_chain_list (GstPad * pad, GstObject * parent,
     GstBufferList * list)
 {
   GstRTPRtxQueue *rtx;
   GstFlowReturn ret;
   GList *pending;

   rtx = GST_RTP_RTX_QUEUE (parent);

   g_mutex_lock (&rtx->lock);
   gst_buffer_list_foreach (list, push_to_queue, rtx->queue);
   shrink_queue (rtx);

   pending = rtx->pending;
   rtx->pending = NULL;
   g_mutex_unlock (&rtx->lock);

   pending = g_list_reverse (pending);
   g_list_foreach (pending, (GFunc) do_push, rtx);
   g_list_free (pending);

   ret = gst_pad_push_list (rtx->srcpad, list);

   return ret;
 }

 static void
 gst_rtp_rtx_queue_get_property (GObject * object,
     guint prop_id, GValue * value, GParamSpec * pspec)
 {
   GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);

   switch (prop_id) {
     case PROP_MAX_SIZE_TIME:
       g_value_set_uint (value, rtx->max_size_time);
       break;
     case PROP_MAX_SIZE_PACKETS:
       g_value_set_uint (value, rtx->max_size_packets);
       break;
     case PROP_REQUESTS:
       g_value_set_uint (value, rtx->n_requests);
       break;
     case PROP_FULFILLED_REQUESTS:
       g_value_set_uint (value, rtx->n_fulfilled_requests);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
 }

 static void
 gst_rtp_rtx_queue_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec)
 {
   GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);

   switch (prop_id) {
     case PROP_MAX_SIZE_TIME:
       rtx->max_size_time = g_value_get_uint (value);
       break;
     case PROP_MAX_SIZE_PACKETS:
       rtx->max_size_packets = g_value_get_uint (value);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
 }

 static GstStateChangeReturn
 gst_rtp_rtx_queue_change_state (GstElement * element, GstStateChange transition)
 {
   GstStateChangeReturn ret;
   GstRTPRtxQueue *rtx;

   rtx = GST_RTP_RTX_QUEUE (element);

   switch (transition) {
     default:
       break;
   }

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

   switch (transition) {
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       gst_rtp_rtx_queue_reset (rtx, TRUE);
       break;
     default:
       break;
   }

   return ret;
 }

 gboolean
 gst_rtp_rtx_queue_plugin_init (GstPlugin * plugin)
 {
   GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_queue_debug, "rtprtxqueue", 0,
       "rtp retransmission queue");

   return gst_element_register (plugin, "rtprtxqueue", GST_RANK_NONE,
       GST_TYPE_RTP_RTX_QUEUE);
 }
	/* RTP Retransmission queue element for GStreamer
	*
	* gstrtprtxqueue.c:
	*
	* Copyright (C) 2013 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.
	*/

	/**
	* SECTION:element-rtprtxqueue
	*
	* rtprtxqueue maintains a queue of transmitted RTP packets, up to a
	* configurable limit (see #GstRTPRtxQueue::max-size-time,
	* #GstRTPRtxQueue::max-size-packets), and retransmits them upon request
	* from the downstream rtpsession (GstRTPRetransmissionRequest event).
	*
	* This element is similar to rtprtxsend, but it has differences:
	* - Retransmission from rtprtxqueue is not RFC 4588 compliant. The
	* retransmitted packets have the same ssrc and payload type as the original
	* stream.
	* - As a side-effect of the above, rtprtxqueue does not require the use of
	* rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to
	* reconstruct the stream.
	* - Retransmission from rtprtxqueue happens as soon as the next regular flow
	* packet is chained, while rtprtxsend retransmits as soon as the retransmission
	* event is received, using a helper thread.
	* - rtprtxqueue can be used with rtpbin without the need of hooking to its
	* #GstRtpBin::request-aux-sender signal, which means it can be used with
	* rtpbin using gst-launch.
	*
	* See also #GstRtpRtxSend, #GstRtpRtxReceive
	*
	* # Example pipelines
	* \|[
	* gst-launch-1.0 rtpbin name=b rtp-profile=avpf \
	* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \
	* b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \
	* udpsrc port=5001 ! b.recv_rtcp_sink_0 \
	* b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false
	* ]\| Sender pipeline
	* \|[
	* gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \
	* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
	* b.recv_rtp_sink_0 \
	* b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
	* udpsrc port=5002 ! b.recv_rtcp_sink_0 \
	* b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false
	* ]\| Receiver pipeline
	*/

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

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

	#include "gstrtprtxqueue.h"

	GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_queue_debug);
	#define GST_CAT_DEFAULT gst_rtp_rtx_queue_debug

	#define DEFAULT_MAX_SIZE_TIME 0
	#define DEFAULT_MAX_SIZE_PACKETS 100

	enum
	{
	PROP_0,
	PROP_MAX_SIZE_TIME,
	PROP_MAX_SIZE_PACKETS,
	PROP_REQUESTS,
	PROP_FULFILLED_REQUESTS,
	};

	static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("application/x-rtp")
	);

	static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
	GST_PAD_SINK,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("application/x-rtp")
	);

	static gboolean gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent,
	GstEvent * event);
	static gboolean gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent,
	GstEvent * event);
	static GstFlowReturn gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent,
	GstBuffer * buffer);
	static GstFlowReturn gst_rtp_rtx_queue_chain_list (GstPad * pad,
	GstObject * parent, GstBufferList * list);

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

	static void gst_rtp_rtx_queue_set_property (GObject * object, guint prop_id,
	const GValue * value, GParamSpec * pspec);
	static void gst_rtp_rtx_queue_get_property (GObject * object, guint prop_id,
	GValue * value, GParamSpec * pspec);
	static void gst_rtp_rtx_queue_finalize (GObject * object);

	G_DEFINE_TYPE (GstRTPRtxQueue, gst_rtp_rtx_queue, GST_TYPE_ELEMENT);

	static void
	gst_rtp_rtx_queue_class_init (GstRTPRtxQueueClass * klass)
	{
	GObjectClass *gobject_class;
	GstElementClass *gstelement_class;

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

	gobject_class->get_property = gst_rtp_rtx_queue_get_property;
	gobject_class->set_property = gst_rtp_rtx_queue_set_property;
	gobject_class->finalize = gst_rtp_rtx_queue_finalize;

	g_object_class_install_property (gobject_class, PROP_MAX_SIZE_TIME,
	g_param_spec_uint ("max-size-time", "Max Size Times",
	"Amount of ms to queue (0 = unlimited)", 0, G_MAXUINT,
	DEFAULT_MAX_SIZE_TIME, G_PARAM_READWRITE \| G_PARAM_STATIC_STRINGS));

	g_object_class_install_property (gobject_class, PROP_MAX_SIZE_PACKETS,
	g_param_spec_uint ("max-size-packets", "Max Size Packets",
	"Amount of packets to queue (0 = unlimited)", 0, G_MAXUINT,
	DEFAULT_MAX_SIZE_PACKETS,
	G_PARAM_READWRITE \| G_PARAM_STATIC_STRINGS));

	g_object_class_install_property (gobject_class, PROP_REQUESTS,
	g_param_spec_uint ("requests", "Requests",
	"Total number of retransmission requests", 0, G_MAXUINT,
	0, G_PARAM_READABLE \| G_PARAM_STATIC_STRINGS));

	g_object_class_install_property (gobject_class, PROP_FULFILLED_REQUESTS,
	g_param_spec_uint ("fulfilled-requests", "Fulfilled Requests",
	"Number of fulfilled retransmission requests", 0, G_MAXUINT,
	0, G_PARAM_READABLE \| G_PARAM_STATIC_STRINGS));

	gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
	gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);

	gst_element_class_set_static_metadata (gstelement_class,
	"RTP Retransmission Queue", "Codec",
	"Keep RTP packets in a queue for retransmission",
	"Wim Taymans <wim.taymans@gmail.com>");

	gstelement_class->change_state =
	GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_change_state);
	}

	static void
	gst_rtp_rtx_queue_reset (GstRTPRtxQueue * rtx, gboolean full)
	{
	g_mutex_lock (&rtx->lock);
	g_queue_foreach (rtx->queue, (GFunc) gst_buffer_unref, NULL);
	g_queue_clear (rtx->queue);
	g_list_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
	g_list_free (rtx->pending);
	rtx->pending = NULL;
	rtx->n_requests = 0;
	rtx->n_fulfilled_requests = 0;
	g_mutex_unlock (&rtx->lock);
	}

	static void
	gst_rtp_rtx_queue_finalize (GObject * object)
	{
	GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);

	gst_rtp_rtx_queue_reset (rtx, TRUE);
	g_queue_free (rtx->queue);
	g_mutex_clear (&rtx->lock);

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

	static void
	gst_rtp_rtx_queue_init (GstRTPRtxQueue * rtx)
	{
	GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx);

	rtx->srcpad =
	gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
	"src"), "src");
	GST_PAD_SET_PROXY_CAPS (rtx->srcpad);
	GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad);
	gst_pad_set_event_function (rtx->srcpad,
	GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_src_event));
	gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad);

	rtx->sinkpad =
	gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
	"sink"), "sink");
	GST_PAD_SET_PROXY_CAPS (rtx->sinkpad);
	GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
	gst_pad_set_event_function (rtx->sinkpad,
	GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_sink_event));
	gst_pad_set_chain_function (rtx->sinkpad,
	GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain));
	gst_pad_set_chain_list_function (rtx->sinkpad,
	GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain_list));
	gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);

	rtx->queue = g_queue_new ();
	g_mutex_init (&rtx->lock);

	rtx->max_size_time = DEFAULT_MAX_SIZE_TIME;
	rtx->max_size_packets = DEFAULT_MAX_SIZE_PACKETS;
	}

	typedef struct
	{
	GstRTPRtxQueue *rtx;
	guint seqnum;
	gboolean found;
	} RTXData;

	static void
	push_seqnum (GstBuffer * buffer, RTXData * data)
	{
	GstRTPRtxQueue *rtx = data->rtx;
	GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
	guint16 seqnum;

	if (data->found)
	return;

	if (!GST_IS_BUFFER (buffer) \|\|
	!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer))
	return;

	seqnum = gst_rtp_buffer_get_seq (&rtpbuffer);
	gst_rtp_buffer_unmap (&rtpbuffer);

	if (seqnum == data->seqnum) {
	data->found = TRUE;
	GST_DEBUG_OBJECT (rtx, "found %d", seqnum);
	rtx->pending = g_list_prepend (rtx->pending, gst_buffer_ref (buffer));
	}
	}

	static gboolean
	gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
	{
	GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent);
	gboolean res;

	switch (GST_EVENT_TYPE (event)) {
	case GST_EVENT_CUSTOM_UPSTREAM:
	{
	const GstStructure *s;

	s = gst_event_get_structure (event);
	if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
	guint seqnum;
	RTXData data;

	if (!gst_structure_get_uint (s, "seqnum", &seqnum))
	seqnum = -1;

	GST_DEBUG_OBJECT (rtx, "request %d", seqnum);

	g_mutex_lock (&rtx->lock);
	data.rtx = rtx;
	data.seqnum = seqnum;
	data.found = FALSE;
	rtx->n_requests += 1;
	g_queue_foreach (rtx->queue, (GFunc) push_seqnum, &data);
	g_mutex_unlock (&rtx->lock);

	gst_event_unref (event);
	res = TRUE;
	} else {
	res = gst_pad_event_default (pad, parent, event);
	}
	break;
	}
	default:
	res = gst_pad_event_default (pad, parent, event);
	break;
	}
	return res;
	}

	static gboolean
	gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent,
	GstEvent * event)
	{
	GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent);
	gboolean res;

	switch (GST_EVENT_TYPE (event)) {
	case GST_EVENT_SEGMENT:
	{
	g_mutex_lock (&rtx->lock);
	gst_event_copy_segment (event, &rtx->head_segment);
	g_queue_push_head (rtx->queue, gst_event_ref (event));
	g_mutex_unlock (&rtx->lock);
	/* fall through */
	}
	default:
	res = gst_pad_event_default (pad, parent, event);
	break;
	}
	return res;
	}

	static void
	do_push (GstBuffer * buffer, GstRTPRtxQueue * rtx)
	{
	rtx->n_fulfilled_requests += 1;
	gst_pad_push (rtx->srcpad, buffer);
	}

	static guint32
	get_ts_diff (GstRTPRtxQueue * rtx)
	{
	GstClockTime high_ts, low_ts;
	GstClockTimeDiff result;
	GstBuffer high_buf, low_buf;

	high_buf = g_queue_peek_head (rtx->queue);

	while (GST_IS_EVENT ((low_buf = g_queue_peek_tail (rtx->queue)))) {
	GstEvent *event = g_queue_pop_tail (rtx->queue);
	gst_event_copy_segment (event, &rtx->tail_segment);
	gst_event_unref (event);
	}

	if (!high_buf \|\| !low_buf \|\| high_buf == low_buf)
	return 0;

	high_ts = GST_BUFFER_TIMESTAMP (high_buf);
	low_ts = GST_BUFFER_TIMESTAMP (low_buf);

	high_ts = gst_segment_to_running_time (&rtx->head_segment, GST_FORMAT_TIME,
	high_ts);
	low_ts = gst_segment_to_running_time (&rtx->tail_segment, GST_FORMAT_TIME,
	low_ts);

	result = high_ts - low_ts;

	/* return value in ms instead of ns */
	return (guint32) gst_util_uint64_scale_int (result, 1, GST_MSECOND);
	}

	/* Must be called with rtx->lock */
	static void
	shrink_queue (GstRTPRtxQueue * rtx)
	{
	if (rtx->max_size_packets) {
	while (g_queue_get_length (rtx->queue) > rtx->max_size_packets)
	gst_buffer_unref (g_queue_pop_tail (rtx->queue));
	}
	if (rtx->max_size_time) {
	while (get_ts_diff (rtx) > rtx->max_size_time)
	gst_buffer_unref (g_queue_pop_tail (rtx->queue));
	}
	}

	static GstFlowReturn
	gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
	{
	GstRTPRtxQueue *rtx;
	GstFlowReturn ret;
	GList *pending;

	rtx = GST_RTP_RTX_QUEUE (parent);

	g_mutex_lock (&rtx->lock);
	g_queue_push_head (rtx->queue, gst_buffer_ref (buffer));
	shrink_queue (rtx);

	pending = rtx->pending;
	rtx->pending = NULL;
	g_mutex_unlock (&rtx->lock);

	pending = g_list_reverse (pending);
	g_list_foreach (pending, (GFunc) do_push, rtx);
	g_list_free (pending);

	ret = gst_pad_push (rtx->srcpad, buffer);

	return ret;
	}

	static gboolean
	push_to_queue (GstBuffer ** buffer, guint idx, gpointer user_data)
	{
	GQueue *queue = user_data;

	g_queue_push_head (queue, gst_buffer_ref (*buffer));

	return TRUE;
	}

	static GstFlowReturn
	gst_rtp_rtx_queue_chain_list (GstPad * pad, GstObject * parent,
	GstBufferList * list)
	{
	GstRTPRtxQueue *rtx;
	GstFlowReturn ret;
	GList *pending;

	rtx = GST_RTP_RTX_QUEUE (parent);

	g_mutex_lock (&rtx->lock);
	gst_buffer_list_foreach (list, push_to_queue, rtx->queue);
	shrink_queue (rtx);

	pending = rtx->pending;
	rtx->pending = NULL;
	g_mutex_unlock (&rtx->lock);

	pending = g_list_reverse (pending);
	g_list_foreach (pending, (GFunc) do_push, rtx);
	g_list_free (pending);

	ret = gst_pad_push_list (rtx->srcpad, list);

	return ret;
	}

	static void
	gst_rtp_rtx_queue_get_property (GObject * object,
	guint prop_id, GValue * value, GParamSpec * pspec)
	{
	GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);

	switch (prop_id) {
	case PROP_MAX_SIZE_TIME:
	g_value_set_uint (value, rtx->max_size_time);
	break;
	case PROP_MAX_SIZE_PACKETS:
	g_value_set_uint (value, rtx->max_size_packets);
	break;
	case PROP_REQUESTS:
	g_value_set_uint (value, rtx->n_requests);
	break;
	case PROP_FULFILLED_REQUESTS:
	g_value_set_uint (value, rtx->n_fulfilled_requests);
	break;
	default:
	G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
	break;
	}
	}

	static void
	gst_rtp_rtx_queue_set_property (GObject * object,
	guint prop_id, const GValue * value, GParamSpec * pspec)
	{
	GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);

	switch (prop_id) {
	case PROP_MAX_SIZE_TIME:
	rtx->max_size_time = g_value_get_uint (value);
	break;
	case PROP_MAX_SIZE_PACKETS:
	rtx->max_size_packets = g_value_get_uint (value);
	break;
	default:
	G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
	break;
	}
	}

	static GstStateChangeReturn
	gst_rtp_rtx_queue_change_state (GstElement * element, GstStateChange transition)
	{
	GstStateChangeReturn ret;
	GstRTPRtxQueue *rtx;

	rtx = GST_RTP_RTX_QUEUE (element);

	switch (transition) {
	default:
	break;
	}

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

	switch (transition) {
	case GST_STATE_CHANGE_PAUSED_TO_READY:
	gst_rtp_rtx_queue_reset (rtx, TRUE);
	break;
	default:
	break;
	}

	return ret;
	}

	gboolean
	gst_rtp_rtx_queue_plugin_init (GstPlugin * plugin)
	{
	GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_queue_debug, "rtprtxqueue", 0,
	"rtp retransmission queue");

	return gst_element_register (plugin, "rtprtxqueue", GST_RANK_NONE,
	GST_TYPE_RTP_RTX_QUEUE);
	}