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_sequence_remove_range (g_sequence_get_begin_iter (rtx->queue),
169 g_sequence_get_end_iter (rtx->queue));
170 g_queue_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
171 g_queue_clear (rtx->pending);
172 rtx->master_ssrc = 0;
173 rtx->next_seqnum = g_random_int_range (0, G_MAXUINT16);
174 rtx->rtx_ssrc = g_random_int ();
175 rtx->num_rtx_requests = 0;
176 rtx->num_rtx_packets = 0;
177 g_mutex_unlock (&rtx->lock);
181 gst_rtp_rtx_send_finalize (GObject * object)
183 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (object);
185 gst_rtp_rtx_send_reset (rtx, TRUE);
186 g_sequence_free (rtx->queue);
187 g_queue_free (rtx->pending);
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_sequence_new ((GDestroyNotify) buffer_queue_item_free);
219 rtx->pending = 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)
245 buffer_queue_items_cmp (BufferQueueItem * a, BufferQueueItem * b,
248 /* gst_rtp_buffer_compare_seqnum returns the opposite of what we want,
249 * it returns negative when seqnum1 > seqnum2 and we want negative
250 * when b > a, i.e. a is smaller, so it comes first in the sequence */
251 return gst_rtp_buffer_compare_seqnum (b->seqnum, a->seqnum);
255 gst_rtp_rtx_send_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
257 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
260 switch (GST_EVENT_TYPE (event)) {
261 case GST_EVENT_CUSTOM_UPSTREAM:
263 const GstStructure *s = gst_event_get_structure (event);
265 /* This event usually comes from the downstream gstrtpsession */
266 if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
270 /* retrieve seqnum of the packet that need to be restransmisted */
271 if (!gst_structure_get_uint (s, "seqnum", &seqnum))
274 /* retrieve ssrc of the packet that need to be restransmisted */
275 if (!gst_structure_get_uint (s, "ssrc", &ssrc))
278 GST_DEBUG_OBJECT (rtx,
279 "request seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT,
282 g_mutex_lock (&rtx->lock);
283 /* check if request is for us */
284 if (rtx->master_ssrc == ssrc) {
286 BufferQueueItem search_item;
288 /* update statistics */
289 ++rtx->num_rtx_requests;
291 search_item.seqnum = seqnum;
292 iter = g_sequence_lookup (rtx->queue, &search_item,
293 (GCompareDataFunc) buffer_queue_items_cmp, NULL);
295 BufferQueueItem *item = g_sequence_get (iter);
296 GST_DEBUG_OBJECT (rtx, "found %" G_GUINT16_FORMAT, item->seqnum);
297 g_queue_push_tail (rtx->pending, gst_buffer_ref (item->buffer));
300 g_mutex_unlock (&rtx->lock);
302 gst_event_unref (event);
305 /* This event usually comes from the downstream gstrtpsession */
306 } else if (gst_structure_has_name (s, "GstRTPCollision")) {
309 if (!gst_structure_get_uint (s, "ssrc", &ssrc))
312 GST_DEBUG_OBJECT (rtx, "collision ssrc: %" G_GUINT32_FORMAT, ssrc);
314 g_mutex_lock (&rtx->lock);
316 /* choose another ssrc for our retransmited stream */
317 if (ssrc == rtx->rtx_ssrc) {
318 rtx->rtx_ssrc = choose_ssrc (rtx);
320 /* clear buffers we already saved */
321 g_sequence_remove_range (g_sequence_get_begin_iter (rtx->queue),
322 g_sequence_get_end_iter (rtx->queue));
324 /* clear buffers that are about to be retransmited */
325 g_queue_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
326 g_queue_clear (rtx->pending);
328 g_mutex_unlock (&rtx->lock);
330 /* no need to forward to payloader because we make sure to have
333 gst_event_unref (event);
336 g_mutex_unlock (&rtx->lock);
338 /* forward event to payloader in case collided ssrc is
340 res = gst_pad_event_default (pad, parent, event);
343 res = gst_pad_event_default (pad, parent, event);
348 res = gst_pad_event_default (pad, parent, event);
355 gst_rtp_rtx_send_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
357 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
359 switch (GST_EVENT_TYPE (event)) {
365 gst_event_parse_caps (event, &caps);
366 g_assert (gst_caps_is_fixed (caps));
368 s = gst_caps_get_structure (caps, 0);
369 gst_structure_get_int (s, "clock-rate", &rtx->clock_rate);
371 GST_DEBUG_OBJECT (rtx, "got clock-rate from caps: %d", rtx->clock_rate);
378 return gst_pad_event_default (pad, parent, event);
381 /* like rtp_jitter_buffer_get_ts_diff() */
383 gst_rtp_rtx_send_get_ts_diff (GstRtpRtxSend * self)
385 guint64 high_ts, low_ts;
386 BufferQueueItem *high_buf, *low_buf;
390 g_sequence_get (g_sequence_iter_prev (g_sequence_get_end_iter
392 low_buf = g_sequence_get (g_sequence_get_begin_iter (self->queue));
394 if (!high_buf || !low_buf || high_buf == low_buf)
397 high_ts = high_buf->timestamp;
398 low_ts = low_buf->timestamp;
400 /* it needs to work if ts wraps */
401 if (high_ts >= low_ts) {
402 result = (guint32) (high_ts - low_ts);
404 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
407 /* return value in ms instead of clock ticks */
408 return (guint32) gst_util_uint64_scale_int (result, 1000, self->clock_rate);
411 /* Copy fixed header and extension. Add OSN before to copy payload
412 * Copy memory to avoid to manually copy each rtp buffer field.
415 _gst_rtp_rtx_buffer_new (GstBuffer * buffer, guint32 ssrc, guint16 seqnum,
418 GstMemory *mem = NULL;
419 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
420 GstRTPBuffer new_rtp = GST_RTP_BUFFER_INIT;
421 GstBuffer *new_buffer = gst_buffer_new ();
423 guint payload_len = 0;
425 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
427 /* gst_rtp_buffer_map does not map the payload so do it now */
428 gst_rtp_buffer_get_payload (&rtp);
430 /* If payload type is not set through SDP/property then
431 * just bump the value */
433 fmtp = gst_rtp_buffer_get_payload_type (&rtp) + 1;
435 /* copy fixed header */
436 mem = gst_memory_copy (rtp.map[0].memory, 0, rtp.size[0]);
437 gst_buffer_append_memory (new_buffer, mem);
439 /* copy extension if any */
441 mem = gst_memory_copy (rtp.map[1].memory, 0, rtp.size[1]);
442 gst_buffer_append_memory (new_buffer, mem);
445 /* copy payload and add OSN just before */
446 payload_len = 2 + rtp.size[2];
447 mem = gst_allocator_alloc (NULL, payload_len, NULL);
449 gst_memory_map (mem, &map, GST_MAP_WRITE);
450 GST_WRITE_UINT16_BE (map.data, gst_rtp_buffer_get_seq (&rtp));
452 memcpy (map.data + 2, rtp.data[2], rtp.size[2]);
453 gst_memory_unmap (mem, &map);
454 gst_buffer_append_memory (new_buffer, mem);
456 /* everything needed is copied */
457 gst_rtp_buffer_unmap (&rtp);
459 /* set ssrc, seqnum and fmtp */
460 gst_rtp_buffer_map (new_buffer, GST_MAP_WRITE, &new_rtp);
461 gst_rtp_buffer_set_ssrc (&new_rtp, ssrc);
462 gst_rtp_buffer_set_seq (&new_rtp, seqnum);
463 gst_rtp_buffer_set_payload_type (&new_rtp, fmtp);
464 /* RFC 4588: let other elements do the padding, as normal */
465 gst_rtp_buffer_set_padding (&new_rtp, FALSE);
466 gst_rtp_buffer_unmap (&new_rtp);
471 /* push pending retransmission packet.
472 * it constructs rtx packet from original paclets */
474 do_push (GstBuffer * buffer, GstRtpRtxSend * rtx)
476 /* RFC4588 two streams multiplexed by sending them in the same session using
477 * different SSRC values, i.e., SSRC-multiplexing. */
478 GST_DEBUG_OBJECT (rtx,
479 "retransmit seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT,
480 rtx->next_seqnum, rtx->rtx_ssrc);
481 gst_pad_push (rtx->srcpad, _gst_rtp_rtx_buffer_new (buffer, rtx->rtx_ssrc,
482 rtx->next_seqnum++, rtx->rtx_payload_type));
486 gst_rtp_rtx_send_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
488 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (parent);
489 GstFlowReturn ret = GST_FLOW_ERROR;
490 GQueue *pending = NULL;
491 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
492 BufferQueueItem *item;
494 guint32 ssrc, rtptime;
496 rtx = GST_RTP_RTX_SEND (parent);
498 /* read the information we want from the buffer */
499 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
500 seqnum = gst_rtp_buffer_get_seq (&rtp);
501 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
502 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
503 gst_rtp_buffer_unmap (&rtp);
505 g_mutex_lock (&rtx->lock);
507 /* retrieve master stream ssrc */
508 rtx->master_ssrc = ssrc;
509 /* check if our initial aux ssrc is equal to master */
510 if (rtx->rtx_ssrc == rtx->master_ssrc)
513 /* add current rtp buffer to queue history */
514 item = g_new0 (BufferQueueItem, 1);
515 item->seqnum = seqnum;
516 item->timestamp = rtptime;
517 item->buffer = gst_buffer_ref (buffer);
518 g_sequence_append (rtx->queue, item);
520 /* remove oldest packets from history if they are too many */
521 if (rtx->max_size_packets) {
522 while (g_sequence_get_length (rtx->queue) > rtx->max_size_packets)
523 g_sequence_remove (g_sequence_get_begin_iter (rtx->queue));
525 if (rtx->max_size_time) {
526 while (gst_rtp_rtx_send_get_ts_diff (rtx) > rtx->max_size_time)
527 g_sequence_remove (g_sequence_get_begin_iter (rtx->queue));
530 /* within lock, get packets that have to be retransmited */
531 if (g_queue_get_length (rtx->pending) > 0) {
532 pending = rtx->pending;
533 rtx->pending = g_queue_new ();
535 /* update statistics - assume we will succeed to retransmit those packets */
536 rtx->num_rtx_packets += g_queue_get_length (pending);
539 /* transfer payload type while holding the lock */
540 rtx->rtx_payload_type = rtx->rtx_payload_type_pending;
542 /* no need to hold the lock to push rtx packets */
543 g_mutex_unlock (&rtx->lock);
545 /* retransmit requested packets */
547 g_queue_foreach (pending, (GFunc) do_push, rtx);
548 g_queue_free_full (pending, (GDestroyNotify) gst_buffer_unref);
552 "push seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT, seqnum,
555 /* push current rtp packet */
556 ret = gst_pad_push (rtx->srcpad, buffer);
562 gst_rtp_rtx_send_get_property (GObject * object,
563 guint prop_id, GValue * value, GParamSpec * pspec)
565 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (object);
568 case PROP_RTX_PAYLOAD_TYPE:
569 g_mutex_lock (&rtx->lock);
570 g_value_set_uint (value, rtx->rtx_payload_type_pending);
571 g_mutex_unlock (&rtx->lock);
573 case PROP_MAX_SIZE_TIME:
574 g_mutex_lock (&rtx->lock);
575 g_value_set_uint (value, rtx->max_size_time);
576 g_mutex_unlock (&rtx->lock);
578 case PROP_MAX_SIZE_PACKETS:
579 g_mutex_lock (&rtx->lock);
580 g_value_set_uint (value, rtx->max_size_packets);
581 g_mutex_unlock (&rtx->lock);
583 case PROP_NUM_RTX_REQUESTS:
584 g_mutex_lock (&rtx->lock);
585 g_value_set_uint (value, rtx->num_rtx_requests);
586 g_mutex_unlock (&rtx->lock);
588 case PROP_NUM_RTX_PACKETS:
589 g_mutex_lock (&rtx->lock);
590 g_value_set_uint (value, rtx->num_rtx_packets);
591 g_mutex_unlock (&rtx->lock);
594 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
600 gst_rtp_rtx_send_set_property (GObject * object,
601 guint prop_id, const GValue * value, GParamSpec * pspec)
603 GstRtpRtxSend *rtx = GST_RTP_RTX_SEND (object);
606 case PROP_RTX_PAYLOAD_TYPE:
607 g_mutex_lock (&rtx->lock);
608 rtx->rtx_payload_type_pending = g_value_get_uint (value);
609 g_mutex_unlock (&rtx->lock);
611 case PROP_MAX_SIZE_TIME:
612 g_mutex_lock (&rtx->lock);
613 rtx->max_size_time = g_value_get_uint (value);
614 g_mutex_unlock (&rtx->lock);
616 case PROP_MAX_SIZE_PACKETS:
617 g_mutex_lock (&rtx->lock);
618 rtx->max_size_packets = g_value_get_uint (value);
619 g_mutex_unlock (&rtx->lock);
622 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
627 static GstStateChangeReturn
628 gst_rtp_rtx_send_change_state (GstElement * element, GstStateChange transition)
630 GstStateChangeReturn ret;
633 rtx = GST_RTP_RTX_SEND (element);
635 switch (transition) {
641 GST_ELEMENT_CLASS (gst_rtp_rtx_send_parent_class)->change_state (element,
644 switch (transition) {
645 case GST_STATE_CHANGE_PAUSED_TO_READY:
646 gst_rtp_rtx_send_reset (rtx, TRUE);
656 gst_rtp_rtx_send_plugin_init (GstPlugin * plugin)
658 GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_send_debug, "rtprtxsend", 0,
659 "rtp retransmission sender");
661 return gst_element_register (plugin, "rtprtxsend", GST_RANK_NONE,
662 GST_TYPE_RTP_RTX_SEND);