1 /* RTP Retransmission sender element for GStreamer
5 * Copyright (C) 2013 Collabora Ltd.
6 * @author Julien Isorce <julien.isorce@collabora.co.uk>
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Library General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Library General Public License for more details.
18 * You should have received a copy of the GNU Library General Public
19 * License along with this library; if not, write to the
20 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
21 * Boston, MA 02110-1301, USA.
25 * SECTION:element-rtprtxsend
27 * See #GstRtpRtxReceive for examples
29 * The purpose of the sender RTX object is to keep a history of RTP packets up
30 * to a configurable limit (max-size-time or max-size-packets). It will listen
31 * for upstream custom retransmission events (GstRTPRetransmissionRequest) that
32 * comes from downstream (#GstRtpSession). When receiving a request it will
33 * look up the requested seqnum in its list of stored packets. If the packet
34 * is available, it will create a RTX packet according to RFC 4588 and send
35 * this as an auxiliary stream. RTX is SSRC-multiplexed
43 #include <gst/rtp/gstrtpbuffer.h>
46 #include "gstrtprtxsend.h"
48 GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_send_debug);
49 #define GST_CAT_DEFAULT gst_rtp_rtx_send_debug
51 #define DEFAULT_RTX_PAYLOAD_TYPE 0
52 #define DEFAULT_MAX_SIZE_TIME 0
53 #define DEFAULT_MAX_SIZE_PACKETS 100
58 PROP_RTX_PAYLOAD_TYPE,
60 PROP_MAX_SIZE_PACKETS,
61 PROP_NUM_RTX_REQUESTS,
66 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
69 GST_STATIC_CAPS ("application/x-rtp")
72 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
75 GST_STATIC_CAPS ("application/x-rtp")
78 static gboolean gst_rtp_rtx_send_src_event (GstPad * pad, GstObject * parent,
80 static gboolean gst_rtp_rtx_send_sink_event (GstPad * pad, GstObject * parent,
82 static GstFlowReturn gst_rtp_rtx_send_chain (GstPad * pad, GstObject * parent,
85 static GstStateChangeReturn gst_rtp_rtx_send_change_state (GstElement *
86 element, GstStateChange transition);
88 static void gst_rtp_rtx_send_set_property (GObject * object, guint prop_id,
89 const GValue * value, GParamSpec * pspec);
90 static void gst_rtp_rtx_send_get_property (GObject * object, guint prop_id,
91 GValue * value, GParamSpec * pspec);
92 static void gst_rtp_rtx_send_finalize (GObject * object);
94 G_DEFINE_TYPE (GstRtpRtxSend, gst_rtp_rtx_send, GST_TYPE_ELEMENT);
104 buffer_queue_item_free (BufferQueueItem * item)
106 gst_buffer_unref (item->buffer);
111 gst_rtp_rtx_send_class_init (GstRtpRtxSendClass * klass)
113 GObjectClass *gobject_class;
114 GstElementClass *gstelement_class;
116 gobject_class = (GObjectClass *) klass;
117 gstelement_class = (GstElementClass *) klass;
119 gobject_class->get_property = gst_rtp_rtx_send_get_property;
120 gobject_class->set_property = gst_rtp_rtx_send_set_property;
121 gobject_class->finalize = gst_rtp_rtx_send_finalize;
123 g_object_class_install_property (gobject_class, PROP_RTX_PAYLOAD_TYPE,
124 g_param_spec_uint ("rtx-payload-type", "RTX Payload Type",
125 "Payload type of the retransmission stream (fmtp in SDP)", 0,
126 G_MAXUINT, DEFAULT_RTX_PAYLOAD_TYPE,
127 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
129 g_object_class_install_property (gobject_class, PROP_MAX_SIZE_TIME,
130 g_param_spec_uint ("max-size-time", "Max Size Time",
131 "Amount of ms to queue (0 = unlimited)", 0, G_MAXUINT,
132 DEFAULT_MAX_SIZE_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
134 g_object_class_install_property (gobject_class, PROP_MAX_SIZE_PACKETS,
135 g_param_spec_uint ("max-size-packets", "Max Size Packets",
136 "Amount of packets to queue (0 = unlimited)", 0, G_MAXUINT,
137 DEFAULT_MAX_SIZE_PACKETS,
138 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
140 g_object_class_install_property (gobject_class, PROP_NUM_RTX_REQUESTS,
141 g_param_spec_uint ("num-rtx-requests", "Num RTX Requests",
142 "Number of retransmission events received", 0, G_MAXUINT,
143 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
145 g_object_class_install_property (gobject_class, PROP_NUM_RTX_PACKETS,
146 g_param_spec_uint ("num-rtx-packets", "Num RTX Packets",
147 " Number of retransmission packets sent", 0, G_MAXUINT,
148 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
150 gst_element_class_add_pad_template (gstelement_class,
151 gst_static_pad_template_get (&src_factory));
152 gst_element_class_add_pad_template (gstelement_class,
153 gst_static_pad_template_get (&sink_factory));
155 gst_element_class_set_static_metadata (gstelement_class,
156 "RTP Retransmission Sender", "Codec",
157 "Retransmit RTP packets when needed, according to RFC4588",
158 "Julien Isorce <julien.isorce@collabora.co.uk>");
160 gstelement_class->change_state =
161 GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_change_state);
165 gst_rtp_rtx_send_reset (GstRtpRtxSend * rtx, gboolean full)
167 g_mutex_lock (&rtx->lock);
168 g_queue_foreach (rtx->queue, (GFunc) buffer_queue_item_free, NULL);
169 g_queue_clear (rtx->queue);
170 g_list_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
171 g_list_free (rtx->pending);
173 rtx->master_ssrc = 0;
174 rtx->next_seqnum = g_random_int_range (0, G_MAXUINT16);
175 rtx->rtx_ssrc = g_random_int ();
176 rtx->num_rtx_requests = 0;
177 rtx->num_rtx_packets = 0;
178 g_mutex_unlock (&rtx->lock);
182 gst_rtp_rtx_send_finalize (GObject * object)
184 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (object);
186 gst_rtp_rtx_send_reset (rtx, TRUE);
187 g_queue_free (rtx->queue);
188 g_mutex_clear (&rtx->lock);
190 G_OBJECT_CLASS (gst_rtp_rtx_send_parent_class)->finalize (object);
194 gst_rtp_rtx_send_init (GstRtpRtxSend * rtx)
196 GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx);
199 gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
201 GST_PAD_SET_PROXY_CAPS (rtx->srcpad);
202 GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad);
203 gst_pad_set_event_function (rtx->srcpad,
204 GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_src_event));
205 gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad);
208 gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
210 GST_PAD_SET_PROXY_CAPS (rtx->sinkpad);
211 GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
212 gst_pad_set_event_function (rtx->sinkpad,
213 GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_sink_event));
214 gst_pad_set_chain_function (rtx->sinkpad,
215 GST_DEBUG_FUNCPTR (gst_rtp_rtx_send_chain));
216 gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);
218 rtx->queue = g_queue_new ();
220 g_mutex_init (&rtx->lock);
222 rtx->next_seqnum = g_random_int_range (0, G_MAXUINT16);
223 rtx->rtx_ssrc = g_random_int ();
225 rtx->max_size_time = DEFAULT_MAX_SIZE_TIME;
226 rtx->max_size_packets = DEFAULT_MAX_SIZE_PACKETS;
230 choose_ssrc (GstRtpRtxSend * rtx)
235 ssrc = g_random_int ();
237 /* make sure to be different than master */
238 if (ssrc != rtx->master_ssrc)
251 /* traverse queue history and try to find the buffer that the
252 * requested seqnum */
254 push_seqnum (BufferQueueItem * item, RTXData * data)
256 GstRtpRtxSend *rtx = data->rtx;
261 /* data->seqnum comes from the request */
262 if (item->seqnum == data->seqnum) {
264 GST_DEBUG_OBJECT (rtx, "found %" G_GUINT16_FORMAT, item->seqnum);
265 rtx->pending = g_list_prepend (rtx->pending, gst_buffer_ref (item->buffer));
270 gst_rtp_rtx_send_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
272 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
275 switch (GST_EVENT_TYPE (event)) {
276 case GST_EVENT_CUSTOM_UPSTREAM:
278 const GstStructure *s = gst_event_get_structure (event);
280 /* This event usually comes from the downstream gstrtpsession */
281 if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
286 /* retrieve seqnum of the packet that need to be restransmisted */
287 if (!gst_structure_get_uint (s, "seqnum", &seqnum))
290 /* retrieve ssrc of the packet that need to be restransmisted */
291 if (!gst_structure_get_uint (s, "ssrc", &ssrc))
294 GST_DEBUG_OBJECT (rtx,
295 "request seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT,
298 g_mutex_lock (&rtx->lock);
299 /* check if request is for us */
300 if (rtx->master_ssrc == ssrc) {
301 ++rtx->num_rtx_requests;
303 data.seqnum = seqnum;
305 /* TODO do a binary search because rtx->queue is sorted by seq num */
306 g_queue_foreach (rtx->queue, (GFunc) push_seqnum, &data);
308 g_mutex_unlock (&rtx->lock);
310 gst_event_unref (event);
313 /* This event usually comes from the downstream gstrtpsession */
314 } else if (gst_structure_has_name (s, "GstRTPCollision")) {
317 if (!gst_structure_get_uint (s, "ssrc", &ssrc))
320 GST_DEBUG_OBJECT (rtx, "collision ssrc: %" G_GUINT32_FORMAT, ssrc);
322 g_mutex_lock (&rtx->lock);
324 /* choose another ssrc for our retransmited stream */
325 if (ssrc == rtx->rtx_ssrc) {
326 rtx->rtx_ssrc = choose_ssrc (rtx);
328 /* clear buffers we already saved */
329 g_queue_foreach (rtx->queue, (GFunc) gst_buffer_unref, NULL);
330 g_queue_clear (rtx->queue);
332 /* clear buffers that are about to be retransmited */
333 g_list_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
334 g_list_free (rtx->pending);
337 g_mutex_unlock (&rtx->lock);
339 /* no need to forward to payloader because we make sure to have
342 gst_event_unref (event);
345 g_mutex_unlock (&rtx->lock);
347 /* forward event to payloader in case collided ssrc is
349 res = gst_pad_event_default (pad, parent, event);
352 res = gst_pad_event_default (pad, parent, event);
357 res = gst_pad_event_default (pad, parent, event);
364 gst_rtp_rtx_send_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
366 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
368 switch (GST_EVENT_TYPE (event)) {
374 gst_event_parse_caps (event, &caps);
375 g_assert (gst_caps_is_fixed (caps));
377 s = gst_caps_get_structure (caps, 0);
378 gst_structure_get_int (s, "clock-rate", &rtx->clock_rate);
380 GST_DEBUG_OBJECT (rtx, "got clock-rate from caps: %d", rtx->clock_rate);
387 return gst_pad_event_default (pad, parent, event);
390 /* like rtp_jitter_buffer_get_ts_diff() */
392 gst_rtp_rtx_send_get_ts_diff (GstRtpRtxSend * self)
394 guint64 high_ts, low_ts;
395 BufferQueueItem *high_buf, *low_buf;
398 high_buf = g_queue_peek_head (self->queue);
399 low_buf = g_queue_peek_tail (self->queue);
401 if (!high_buf || !low_buf || high_buf == low_buf)
404 high_ts = high_buf->timestamp;
405 low_ts = low_buf->timestamp;
407 /* it needs to work if ts wraps */
408 if (high_ts >= low_ts) {
409 result = (guint32) (high_ts - low_ts);
411 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
414 /* return value in ms instead of clock ticks */
415 return (guint32) gst_util_uint64_scale_int (result, 1000, self->clock_rate);
418 /* Copy fixed header and extension. Add OSN before to copy payload
419 * Copy memory to avoid to manually copy each rtp buffer field.
422 _gst_rtp_rtx_buffer_new (GstBuffer * buffer, guint32 ssrc, guint16 seqnum,
425 GstMemory *mem = NULL;
426 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
427 GstRTPBuffer new_rtp = GST_RTP_BUFFER_INIT;
428 GstBuffer *new_buffer = gst_buffer_new ();
430 guint payload_len = 0;
432 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
434 /* gst_rtp_buffer_map does not map the payload so do it now */
435 gst_rtp_buffer_get_payload (&rtp);
437 /* If payload type is not set through SDP/property then
438 * just bump the value */
440 fmtp = gst_rtp_buffer_get_payload_type (&rtp) + 1;
442 /* copy fixed header */
443 mem = gst_memory_copy (rtp.map[0].memory, 0, rtp.size[0]);
444 gst_buffer_append_memory (new_buffer, mem);
446 /* copy extension if any */
448 mem = gst_memory_copy (rtp.map[1].memory, 0, rtp.size[1]);
449 gst_buffer_append_memory (new_buffer, mem);
452 /* copy payload and add OSN just before */
453 payload_len = 2 + rtp.size[2];
454 mem = gst_allocator_alloc (NULL, payload_len, NULL);
456 gst_memory_map (mem, &map, GST_MAP_WRITE);
457 GST_WRITE_UINT16_BE (map.data, gst_rtp_buffer_get_seq (&rtp));
459 memcpy (map.data + 2, rtp.data[2], rtp.size[2]);
460 gst_memory_unmap (mem, &map);
461 gst_buffer_append_memory (new_buffer, mem);
463 /* everything needed is copied */
464 gst_rtp_buffer_unmap (&rtp);
466 /* set ssrc, seqnum and fmtp */
467 gst_rtp_buffer_map (new_buffer, GST_MAP_WRITE, &new_rtp);
468 gst_rtp_buffer_set_ssrc (&new_rtp, ssrc);
469 gst_rtp_buffer_set_seq (&new_rtp, seqnum);
470 gst_rtp_buffer_set_payload_type (&new_rtp, fmtp);
471 /* RFC 4588: let other elements do the padding, as normal */
472 gst_rtp_buffer_set_padding (&new_rtp, FALSE);
473 gst_rtp_buffer_unmap (&new_rtp);
478 /* psuh pending retransmission packet.
479 * it constructs rtx packet from original paclets */
481 do_push (GstBuffer * buffer, GstRtpRtxSend * rtx)
483 /* RFC4588 two streams multiplexed by sending them in the same session using
484 * different SSRC values, i.e., SSRC-multiplexing. */
485 GST_DEBUG_OBJECT (rtx,
486 "retransmit seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT,
487 rtx->next_seqnum, rtx->rtx_ssrc);
488 gst_pad_push (rtx->srcpad, _gst_rtp_rtx_buffer_new (buffer, rtx->rtx_ssrc,
489 rtx->next_seqnum++, rtx->rtx_payload_type));
493 gst_rtp_rtx_send_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
495 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
496 GstFlowReturn ret = GST_FLOW_ERROR;
497 GList *pending = NULL;
498 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
499 BufferQueueItem *item;
501 guint32 ssrc, rtptime;
503 rtx = GST_RTP_RTX_SEND (parent);
505 /* read the information we want from the buffer */
506 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
507 seqnum = gst_rtp_buffer_get_seq (&rtp);
508 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
509 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
510 gst_rtp_buffer_unmap (&rtp);
512 g_mutex_lock (&rtx->lock);
514 /* retrieve master stream ssrc */
515 rtx->master_ssrc = ssrc;
516 /* check if our initial aux ssrc is equal to master */
517 if (rtx->rtx_ssrc == rtx->master_ssrc)
520 /* add current rtp buffer to queue history */
521 item = g_new0 (BufferQueueItem, 1);
522 item->seqnum = seqnum;
523 item->timestamp = rtptime;
524 item->buffer = gst_buffer_ref (buffer);
525 g_queue_push_head (rtx->queue, item);
527 /* remove oldest packets from history if they are too many */
528 if (rtx->max_size_packets) {
529 while (g_queue_get_length (rtx->queue) > rtx->max_size_packets)
530 buffer_queue_item_free (g_queue_pop_tail (rtx->queue));
532 if (rtx->max_size_time) {
533 while (gst_rtp_rtx_send_get_ts_diff (rtx) > rtx->max_size_time)
534 buffer_queue_item_free (g_queue_pop_tail (rtx->queue));
537 /* within lock, get packets that have to be retransmited */
538 pending = rtx->pending;
541 /* update statistics - assume we will succeed to retransmit those packets */
542 rtx->num_rtx_packets += g_list_length (pending);
544 /* transfer payload type while holding the lock */
545 rtx->rtx_payload_type = rtx->rtx_payload_type_pending;
547 /* no need to hold the lock to push rtx packets */
548 g_mutex_unlock (&rtx->lock);
550 /* retransmit requested packets */
551 g_list_foreach (pending, (GFunc) do_push, rtx);
552 g_list_foreach (pending, (GFunc) gst_buffer_unref, NULL);
553 g_list_free (pending);
556 "push seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT, seqnum,
559 /* push current rtp packet */
560 ret = gst_pad_push (rtx->srcpad, buffer);
566 gst_rtp_rtx_send_get_property (GObject * object,
567 guint prop_id, GValue * value, GParamSpec * pspec)
569 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (object);
572 case PROP_RTX_PAYLOAD_TYPE:
573 g_mutex_lock (&rtx->lock);
574 g_value_set_uint (value, rtx->rtx_payload_type_pending);
575 g_mutex_unlock (&rtx->lock);
577 case PROP_MAX_SIZE_TIME:
578 g_mutex_lock (&rtx->lock);
579 g_value_set_uint (value, rtx->max_size_time);
580 g_mutex_unlock (&rtx->lock);
582 case PROP_MAX_SIZE_PACKETS:
583 g_mutex_lock (&rtx->lock);
584 g_value_set_uint (value, rtx->max_size_packets);
585 g_mutex_unlock (&rtx->lock);
587 case PROP_NUM_RTX_REQUESTS:
588 g_mutex_lock (&rtx->lock);
589 g_value_set_uint (value, rtx->num_rtx_requests);
590 g_mutex_unlock (&rtx->lock);
592 case PROP_NUM_RTX_PACKETS:
593 g_mutex_lock (&rtx->lock);
594 g_value_set_uint (value, rtx->num_rtx_packets);
595 g_mutex_unlock (&rtx->lock);
598 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
604 gst_rtp_rtx_send_set_property (GObject * object,
605 guint prop_id, const GValue * value, GParamSpec * pspec)
607 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (object);
610 case PROP_RTX_PAYLOAD_TYPE:
611 g_mutex_lock (&rtx->lock);
612 rtx->rtx_payload_type_pending = g_value_get_uint (value);
613 g_mutex_unlock (&rtx->lock);
615 case PROP_MAX_SIZE_TIME:
616 g_mutex_lock (&rtx->lock);
617 rtx->max_size_time = g_value_get_uint (value);
618 g_mutex_unlock (&rtx->lock);
620 case PROP_MAX_SIZE_PACKETS:
621 g_mutex_lock (&rtx->lock);
622 rtx->max_size_packets = g_value_get_uint (value);
623 g_mutex_unlock (&rtx->lock);
626 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
631 static GstStateChangeReturn
632 gst_rtp_rtx_send_change_state (GstElement * element, GstStateChange transition)
634 GstStateChangeReturn ret;
637 rtx = GST_RTP_RTX_SEND (element);
639 switch (transition) {
645 GST_ELEMENT_CLASS (gst_rtp_rtx_send_parent_class)->change_state (element,
648 switch (transition) {
649 case GST_STATE_CHANGE_PAUSED_TO_READY:
650 gst_rtp_rtx_send_reset (rtx, TRUE);
660 gst_rtp_rtx_send_plugin_init (GstPlugin * plugin)
662 GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_send_debug, "rtprtxsend", 0,
663 "rtp retransmission sender");
665 return gst_element_register (plugin, "rtprtxsend", GST_RANK_NONE,
666 GST_TYPE_RTP_RTX_SEND);