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rtpdec.c
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1 /*
2  * RTP input format
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/mathematics.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/time.h"
26 
27 #include "avformat.h"
28 #include "network.h"
29 #include "srtp.h"
30 #include "url.h"
31 #include "rtpdec.h"
32 #include "rtpdec_formats.h"
33 
34 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
35 
37  .enc_name = "L24",
38  .codec_type = AVMEDIA_TYPE_AUDIO,
39  .codec_id = AV_CODEC_ID_PCM_S24BE,
40 };
41 
43  .enc_name = "GSM",
44  .codec_type = AVMEDIA_TYPE_AUDIO,
45  .codec_id = AV_CODEC_ID_GSM,
46 };
47 
49  .enc_name = "X-MP3-draft-00",
50  .codec_type = AVMEDIA_TYPE_AUDIO,
51  .codec_id = AV_CODEC_ID_MP3ADU,
52 };
53 
55  .enc_name = "speex",
56  .codec_type = AVMEDIA_TYPE_AUDIO,
57  .codec_id = AV_CODEC_ID_SPEEX,
58 };
59 
61  .enc_name = "opus",
62  .codec_type = AVMEDIA_TYPE_AUDIO,
63  .codec_id = AV_CODEC_ID_OPUS,
64 };
65 
67  .enc_name = "t140",
68  .codec_type = AVMEDIA_TYPE_SUBTITLE,
69  .codec_id = AV_CODEC_ID_TEXT,
70 };
71 
73 
75 {
77  rtp_first_dynamic_payload_handler = handler;
78 }
79 
81 {
123  ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
124  ff_register_dynamic_payload_handler(&l24_dynamic_handler);
125  ff_register_dynamic_payload_handler(&opus_dynamic_handler);
126  ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
127  ff_register_dynamic_payload_handler(&speex_dynamic_handler);
128  ff_register_dynamic_payload_handler(&t140_dynamic_handler);
129 }
130 
132  enum AVMediaType codec_type)
133 {
135  for (handler = rtp_first_dynamic_payload_handler;
136  handler; handler = handler->next)
137  if (handler->enc_name &&
138  !av_strcasecmp(name, handler->enc_name) &&
139  codec_type == handler->codec_type)
140  return handler;
141  return NULL;
142 }
143 
145  enum AVMediaType codec_type)
146 {
148  for (handler = rtp_first_dynamic_payload_handler;
149  handler; handler = handler->next)
150  if (handler->static_payload_id && handler->static_payload_id == id &&
151  codec_type == handler->codec_type)
152  return handler;
153  return NULL;
154 }
155 
156 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
157  int len)
158 {
159  int payload_len;
160  while (len >= 4) {
161  payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
162 
163  switch (buf[1]) {
164  case RTCP_SR:
165  if (payload_len < 20) {
166  av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
167  return AVERROR_INVALIDDATA;
168  }
169 
171  s->last_rtcp_ntp_time = AV_RB64(buf + 8);
172  s->last_rtcp_timestamp = AV_RB32(buf + 16);
175  if (!s->base_timestamp)
178  }
179 
180  break;
181  case RTCP_BYE:
182  return -RTCP_BYE;
183  }
184 
185  buf += payload_len;
186  len -= payload_len;
187  }
188  return -1;
189 }
190 
191 #define RTP_SEQ_MOD (1 << 16)
192 
193 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
194 {
195  memset(s, 0, sizeof(RTPStatistics));
196  s->max_seq = base_sequence;
197  s->probation = 1;
198 }
199 
200 /*
201  * Called whenever there is a large jump in sequence numbers,
202  * or when they get out of probation...
203  */
204 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
205 {
206  s->max_seq = seq;
207  s->cycles = 0;
208  s->base_seq = seq - 1;
209  s->bad_seq = RTP_SEQ_MOD + 1;
210  s->received = 0;
211  s->expected_prior = 0;
212  s->received_prior = 0;
213  s->jitter = 0;
214  s->transit = 0;
215 }
216 
217 /* Returns 1 if we should handle this packet. */
218 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
219 {
220  uint16_t udelta = seq - s->max_seq;
221  const int MAX_DROPOUT = 3000;
222  const int MAX_MISORDER = 100;
223  const int MIN_SEQUENTIAL = 2;
224 
225  /* source not valid until MIN_SEQUENTIAL packets with sequence
226  * seq. numbers have been received */
227  if (s->probation) {
228  if (seq == s->max_seq + 1) {
229  s->probation--;
230  s->max_seq = seq;
231  if (s->probation == 0) {
232  rtp_init_sequence(s, seq);
233  s->received++;
234  return 1;
235  }
236  } else {
237  s->probation = MIN_SEQUENTIAL - 1;
238  s->max_seq = seq;
239  }
240  } else if (udelta < MAX_DROPOUT) {
241  // in order, with permissible gap
242  if (seq < s->max_seq) {
243  // sequence number wrapped; count another 64k cycles
244  s->cycles += RTP_SEQ_MOD;
245  }
246  s->max_seq = seq;
247  } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
248  // sequence made a large jump...
249  if (seq == s->bad_seq) {
250  /* two sequential packets -- assume that the other side
251  * restarted without telling us; just resync. */
252  rtp_init_sequence(s, seq);
253  } else {
254  s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
255  return 0;
256  }
257  } else {
258  // duplicate or reordered packet...
259  }
260  s->received++;
261  return 1;
262 }
263 
264 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
265  uint32_t arrival_timestamp)
266 {
267  // Most of this is pretty straight from RFC 3550 appendix A.8
268  uint32_t transit = arrival_timestamp - sent_timestamp;
269  uint32_t prev_transit = s->transit;
270  int32_t d = transit - prev_transit;
271  // Doing the FFABS() call directly on the "transit - prev_transit"
272  // expression doesn't work, since it's an unsigned expression. Doing the
273  // transit calculation in unsigned is desired though, since it most
274  // probably will need to wrap around.
275  d = FFABS(d);
276  s->transit = transit;
277  if (!prev_transit)
278  return;
279  s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
280 }
281 
283  AVIOContext *avio, int count)
284 {
285  AVIOContext *pb;
286  uint8_t *buf;
287  int len;
288  int rtcp_bytes;
290  uint32_t lost;
291  uint32_t extended_max;
292  uint32_t expected_interval;
293  uint32_t received_interval;
294  int32_t lost_interval;
295  uint32_t expected;
296  uint32_t fraction;
297 
298  if ((!fd && !avio) || (count < 1))
299  return -1;
300 
301  /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
302  /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
303  s->octet_count += count;
304  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
306  rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
307  if (rtcp_bytes < 28)
308  return -1;
310 
311  if (!fd)
312  pb = avio;
313  else if (avio_open_dyn_buf(&pb) < 0)
314  return -1;
315 
316  // Receiver Report
317  avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
318  avio_w8(pb, RTCP_RR);
319  avio_wb16(pb, 7); /* length in words - 1 */
320  // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
321  avio_wb32(pb, s->ssrc + 1);
322  avio_wb32(pb, s->ssrc); // server SSRC
323  // some placeholders we should really fill...
324  // RFC 1889/p64
325  extended_max = stats->cycles + stats->max_seq;
326  expected = extended_max - stats->base_seq;
327  lost = expected - stats->received;
328  lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
329  expected_interval = expected - stats->expected_prior;
330  stats->expected_prior = expected;
331  received_interval = stats->received - stats->received_prior;
332  stats->received_prior = stats->received;
333  lost_interval = expected_interval - received_interval;
334  if (expected_interval == 0 || lost_interval <= 0)
335  fraction = 0;
336  else
337  fraction = (lost_interval << 8) / expected_interval;
338 
339  fraction = (fraction << 24) | lost;
340 
341  avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
342  avio_wb32(pb, extended_max); /* max sequence received */
343  avio_wb32(pb, stats->jitter >> 4); /* jitter */
344 
346  avio_wb32(pb, 0); /* last SR timestamp */
347  avio_wb32(pb, 0); /* delay since last SR */
348  } else {
349  uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
350  uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
351  65536, AV_TIME_BASE);
352 
353  avio_wb32(pb, middle_32_bits); /* last SR timestamp */
354  avio_wb32(pb, delay_since_last); /* delay since last SR */
355  }
356 
357  // CNAME
358  avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
359  avio_w8(pb, RTCP_SDES);
360  len = strlen(s->hostname);
361  avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
362  avio_wb32(pb, s->ssrc + 1);
363  avio_w8(pb, 0x01);
364  avio_w8(pb, len);
365  avio_write(pb, s->hostname, len);
366  avio_w8(pb, 0); /* END */
367  // padding
368  for (len = (7 + len) % 4; len % 4; len++)
369  avio_w8(pb, 0);
370 
371  avio_flush(pb);
372  if (!fd)
373  return 0;
374  len = avio_close_dyn_buf(pb, &buf);
375  if ((len > 0) && buf) {
376  int av_unused result;
377  av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
378  result = ffurl_write(fd, buf, len);
379  av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
380  av_free(buf);
381  }
382  return 0;
383 }
384 
386 {
387  AVIOContext *pb;
388  uint8_t *buf;
389  int len;
390 
391  /* Send a small RTP packet */
392  if (avio_open_dyn_buf(&pb) < 0)
393  return;
394 
395  avio_w8(pb, (RTP_VERSION << 6));
396  avio_w8(pb, 0); /* Payload type */
397  avio_wb16(pb, 0); /* Seq */
398  avio_wb32(pb, 0); /* Timestamp */
399  avio_wb32(pb, 0); /* SSRC */
400 
401  avio_flush(pb);
402  len = avio_close_dyn_buf(pb, &buf);
403  if ((len > 0) && buf)
404  ffurl_write(rtp_handle, buf, len);
405  av_free(buf);
406 
407  /* Send a minimal RTCP RR */
408  if (avio_open_dyn_buf(&pb) < 0)
409  return;
410 
411  avio_w8(pb, (RTP_VERSION << 6));
412  avio_w8(pb, RTCP_RR); /* receiver report */
413  avio_wb16(pb, 1); /* length in words - 1 */
414  avio_wb32(pb, 0); /* our own SSRC */
415 
416  avio_flush(pb);
417  len = avio_close_dyn_buf(pb, &buf);
418  if ((len > 0) && buf)
419  ffurl_write(rtp_handle, buf, len);
420  av_free(buf);
421 }
422 
423 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
424  uint16_t *missing_mask)
425 {
426  int i;
427  uint16_t next_seq = s->seq + 1;
428  RTPPacket *pkt = s->queue;
429 
430  if (!pkt || pkt->seq == next_seq)
431  return 0;
432 
433  *missing_mask = 0;
434  for (i = 1; i <= 16; i++) {
435  uint16_t missing_seq = next_seq + i;
436  while (pkt) {
437  int16_t diff = pkt->seq - missing_seq;
438  if (diff >= 0)
439  break;
440  pkt = pkt->next;
441  }
442  if (!pkt)
443  break;
444  if (pkt->seq == missing_seq)
445  continue;
446  *missing_mask |= 1 << (i - 1);
447  }
448 
449  *first_missing = next_seq;
450  return 1;
451 }
452 
454  AVIOContext *avio)
455 {
456  int len, need_keyframe, missing_packets;
457  AVIOContext *pb;
458  uint8_t *buf;
459  int64_t now;
460  uint16_t first_missing = 0, missing_mask = 0;
461 
462  if (!fd && !avio)
463  return -1;
464 
465  need_keyframe = s->handler && s->handler->need_keyframe &&
467  missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
468 
469  if (!need_keyframe && !missing_packets)
470  return 0;
471 
472  /* Send new feedback if enough time has elapsed since the last
473  * feedback packet. */
474 
475  now = av_gettime_relative();
476  if (s->last_feedback_time &&
478  return 0;
479  s->last_feedback_time = now;
480 
481  if (!fd)
482  pb = avio;
483  else if (avio_open_dyn_buf(&pb) < 0)
484  return -1;
485 
486  if (need_keyframe) {
487  avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
488  avio_w8(pb, RTCP_PSFB);
489  avio_wb16(pb, 2); /* length in words - 1 */
490  // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
491  avio_wb32(pb, s->ssrc + 1);
492  avio_wb32(pb, s->ssrc); // server SSRC
493  }
494 
495  if (missing_packets) {
496  avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
497  avio_w8(pb, RTCP_RTPFB);
498  avio_wb16(pb, 3); /* length in words - 1 */
499  avio_wb32(pb, s->ssrc + 1);
500  avio_wb32(pb, s->ssrc); // server SSRC
501 
502  avio_wb16(pb, first_missing);
503  avio_wb16(pb, missing_mask);
504  }
505 
506  avio_flush(pb);
507  if (!fd)
508  return 0;
509  len = avio_close_dyn_buf(pb, &buf);
510  if (len > 0 && buf) {
511  ffurl_write(fd, buf, len);
512  av_free(buf);
513  }
514  return 0;
515 }
516 
517 /**
518  * open a new RTP parse context for stream 'st'. 'st' can be NULL for
519  * MPEG-2 TS streams.
520  */
522  int payload_type, int queue_size)
523 {
525 
526  s = av_mallocz(sizeof(RTPDemuxContext));
527  if (!s)
528  return NULL;
529  s->payload_type = payload_type;
532  s->ic = s1;
533  s->st = st;
534  s->queue_size = queue_size;
535 
536  av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
537  s->queue_size);
538 
540  if (st) {
541  switch (st->codecpar->codec_id) {
543  /* According to RFC 3551, the stream clock rate is 8000
544  * even if the sample rate is 16000. */
545  if (st->codecpar->sample_rate == 8000)
546  st->codecpar->sample_rate = 16000;
547  break;
548  default:
549  break;
550  }
551  }
552  // needed to send back RTCP RR in RTSP sessions
553  gethostname(s->hostname, sizeof(s->hostname));
554  return s;
555 }
556 
559 {
561  s->handler = handler;
562 }
563 
564 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
565  const char *params)
566 {
567  if (!ff_srtp_set_crypto(&s->srtp, suite, params))
568  s->srtp_enabled = 1;
569 }
570 
571 /**
572  * This was the second switch in rtp_parse packet.
573  * Normalizes time, if required, sets stream_index, etc.
574  */
575 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
576 {
577  if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
578  return; /* Timestamp already set by depacketizer */
579  if (timestamp == RTP_NOTS_VALUE)
580  return;
581 
582  if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
583  int64_t addend;
584  int delta_timestamp;
585 
586  /* compute pts from timestamp with received ntp_time */
587  delta_timestamp = timestamp - s->last_rtcp_timestamp;
588  /* convert to the PTS timebase */
590  s->st->time_base.den,
591  (uint64_t) s->st->time_base.num << 32);
592  pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
593  delta_timestamp;
594  return;
595  }
596 
597  if (!s->base_timestamp)
598  s->base_timestamp = timestamp;
599  /* assume that the difference is INT32_MIN < x < INT32_MAX,
600  * but allow the first timestamp to exceed INT32_MAX */
601  if (!s->timestamp)
602  s->unwrapped_timestamp += timestamp;
603  else
604  s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
605  s->timestamp = timestamp;
607  s->base_timestamp;
608 }
609 
611  const uint8_t *buf, int len)
612 {
613  unsigned int ssrc;
614  int payload_type, seq, flags = 0;
615  int ext, csrc;
616  AVStream *st;
617  uint32_t timestamp;
618  int rv = 0;
619 
620  csrc = buf[0] & 0x0f;
621  ext = buf[0] & 0x10;
622  payload_type = buf[1] & 0x7f;
623  if (buf[1] & 0x80)
624  flags |= RTP_FLAG_MARKER;
625  seq = AV_RB16(buf + 2);
626  timestamp = AV_RB32(buf + 4);
627  ssrc = AV_RB32(buf + 8);
628  /* store the ssrc in the RTPDemuxContext */
629  s->ssrc = ssrc;
630 
631  /* NOTE: we can handle only one payload type */
632  if (s->payload_type != payload_type)
633  return -1;
634 
635  st = s->st;
636  // only do something with this if all the rtp checks pass...
637  if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
638  av_log(s->ic, AV_LOG_ERROR,
639  "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
640  payload_type, seq, ((s->seq + 1) & 0xffff));
641  return -1;
642  }
643 
644  if (buf[0] & 0x20) {
645  int padding = buf[len - 1];
646  if (len >= 12 + padding)
647  len -= padding;
648  }
649 
650  s->seq = seq;
651  len -= 12;
652  buf += 12;
653 
654  len -= 4 * csrc;
655  buf += 4 * csrc;
656  if (len < 0)
657  return AVERROR_INVALIDDATA;
658 
659  /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
660  if (ext) {
661  if (len < 4)
662  return -1;
663  /* calculate the header extension length (stored as number
664  * of 32-bit words) */
665  ext = (AV_RB16(buf + 2) + 1) << 2;
666 
667  if (len < ext)
668  return -1;
669  // skip past RTP header extension
670  len -= ext;
671  buf += ext;
672  }
673 
674  if (s->handler && s->handler->parse_packet) {
676  s->st, pkt, &timestamp, buf, len, seq,
677  flags);
678  } else if (st) {
679  if ((rv = av_new_packet(pkt, len)) < 0)
680  return rv;
681  memcpy(pkt->data, buf, len);
682  pkt->stream_index = st->index;
683  } else {
684  return AVERROR(EINVAL);
685  }
686 
687  // now perform timestamp things....
688  finalize_packet(s, pkt, timestamp);
689 
690  return rv;
691 }
692 
694 {
695  while (s->queue) {
696  RTPPacket *next = s->queue->next;
697  av_freep(&s->queue->buf);
698  av_freep(&s->queue);
699  s->queue = next;
700  }
701  s->seq = 0;
702  s->queue_len = 0;
703  s->prev_ret = 0;
704 }
705 
707 {
708  uint16_t seq = AV_RB16(buf + 2);
709  RTPPacket **cur = &s->queue, *packet;
710 
711  /* Find the correct place in the queue to insert the packet */
712  while (*cur) {
713  int16_t diff = seq - (*cur)->seq;
714  if (diff < 0)
715  break;
716  cur = &(*cur)->next;
717  }
718 
719  packet = av_mallocz(sizeof(*packet));
720  if (!packet)
721  return AVERROR(ENOMEM);
722  packet->recvtime = av_gettime_relative();
723  packet->seq = seq;
724  packet->len = len;
725  packet->buf = buf;
726  packet->next = *cur;
727  *cur = packet;
728  s->queue_len++;
729 
730  return 0;
731 }
732 
734 {
735  return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
736 }
737 
739 {
740  return s->queue ? s->queue->recvtime : 0;
741 }
742 
744 {
745  int rv;
746  RTPPacket *next;
747 
748  if (s->queue_len <= 0)
749  return -1;
750 
751  if (!has_next_packet(s))
753  "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
754 
755  /* Parse the first packet in the queue, and dequeue it */
756  rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
757  next = s->queue->next;
758  av_freep(&s->queue->buf);
759  av_freep(&s->queue);
760  s->queue = next;
761  s->queue_len--;
762  return rv;
763 }
764 
766  uint8_t **bufptr, int len)
767 {
768  uint8_t *buf = bufptr ? *bufptr : NULL;
769  int flags = 0;
770  uint32_t timestamp;
771  int rv = 0;
772 
773  if (!buf) {
774  /* If parsing of the previous packet actually returned 0 or an error,
775  * there's nothing more to be parsed from that packet, but we may have
776  * indicated that we can return the next enqueued packet. */
777  if (s->prev_ret <= 0)
778  return rtp_parse_queued_packet(s, pkt);
779  /* return the next packets, if any */
780  if (s->handler && s->handler->parse_packet) {
781  /* timestamp should be overwritten by parse_packet, if not,
782  * the packet is left with pts == AV_NOPTS_VALUE */
783  timestamp = RTP_NOTS_VALUE;
785  s->st, pkt, &timestamp, NULL, 0, 0,
786  flags);
787  finalize_packet(s, pkt, timestamp);
788  return rv;
789  }
790  }
791 
792  if (len < 12)
793  return -1;
794 
795  if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
796  return -1;
797  if (RTP_PT_IS_RTCP(buf[1])) {
798  return rtcp_parse_packet(s, buf, len);
799  }
800 
801  if (s->st) {
802  int64_t received = av_gettime_relative();
803  uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
804  s->st->time_base);
805  timestamp = AV_RB32(buf + 4);
806  // Calculate the jitter immediately, before queueing the packet
807  // into the reordering queue.
808  rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
809  }
810 
811  if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
812  /* First packet, or no reordering */
813  return rtp_parse_packet_internal(s, pkt, buf, len);
814  } else {
815  uint16_t seq = AV_RB16(buf + 2);
816  int16_t diff = seq - s->seq;
817  if (diff < 0) {
818  /* Packet older than the previously emitted one, drop */
820  "RTP: dropping old packet received too late\n");
821  return -1;
822  } else if (diff <= 1) {
823  /* Correct packet */
824  rv = rtp_parse_packet_internal(s, pkt, buf, len);
825  return rv;
826  } else {
827  /* Still missing some packet, enqueue this one. */
828  rv = enqueue_packet(s, buf, len);
829  if (rv < 0)
830  return rv;
831  *bufptr = NULL;
832  /* Return the first enqueued packet if the queue is full,
833  * even if we're missing something */
834  if (s->queue_len >= s->queue_size) {
835  av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
836  return rtp_parse_queued_packet(s, pkt);
837  }
838  return -1;
839  }
840  }
841 }
842 
843 /**
844  * Parse an RTP or RTCP packet directly sent as a buffer.
845  * @param s RTP parse context.
846  * @param pkt returned packet
847  * @param bufptr pointer to the input buffer or NULL to read the next packets
848  * @param len buffer len
849  * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
850  * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
851  */
853  uint8_t **bufptr, int len)
854 {
855  int rv;
856  if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
857  return -1;
858  rv = rtp_parse_one_packet(s, pkt, bufptr, len);
859  s->prev_ret = rv;
860  while (rv < 0 && has_next_packet(s))
861  rv = rtp_parse_queued_packet(s, pkt);
862  return rv ? rv : has_next_packet(s);
863 }
864 
866 {
868  ff_srtp_free(&s->srtp);
869  av_free(s);
870 }
871 
873  AVStream *stream, PayloadContext *data, const char *p,
874  int (*parse_fmtp)(AVFormatContext *s,
875  AVStream *stream,
876  PayloadContext *data,
877  const char *attr, const char *value))
878 {
879  char attr[256];
880  char *value;
881  int res;
882  int value_size = strlen(p) + 1;
883 
884  if (!(value = av_malloc(value_size))) {
885  av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
886  return AVERROR(ENOMEM);
887  }
888 
889  // remove protocol identifier
890  while (*p && *p == ' ')
891  p++; // strip spaces
892  while (*p && *p != ' ')
893  p++; // eat protocol identifier
894  while (*p && *p == ' ')
895  p++; // strip trailing spaces
896 
897  while (ff_rtsp_next_attr_and_value(&p,
898  attr, sizeof(attr),
899  value, value_size)) {
900  res = parse_fmtp(s, stream, data, attr, value);
901  if (res < 0 && res != AVERROR_PATCHWELCOME) {
902  av_free(value);
903  return res;
904  }
905  }
906  av_free(value);
907  return 0;
908 }
909 
910 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
911 {
912  int ret;
913  av_init_packet(pkt);
914 
915  pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
916  pkt->stream_index = stream_idx;
917  *dyn_buf = NULL;
918  if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
919  av_freep(&pkt->data);
920  return ret;
921  }
922  return pkt->size;
923 }
int queue_size
The size of queue, or 0 if reordering is disabled.
Definition: rtpdec.h:174
RTPDynamicProtocolHandler ff_g726le_24_dynamic_handler
#define NULL
Definition: coverity.c:32
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
Definition: rtpdec.c:564
const char * s
Definition: avisynth_c.h:768
Bytestream IO Context.
Definition: avio.h:147
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
AVFormatContext * ic
Definition: rtpdec.h:150
uint16_t seq
Definition: rtpdec.h:154
RTPDynamicProtocolHandler ff_quicktime_rtp_aud_handler
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
Send a dummy packet on both port pairs to set up the connection state in potential NAT routers...
Definition: rtpdec.c:385
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
Definition: aviobuf.c:1280
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
Definition: rtpdec.c:264
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:421
int payload_type
Definition: rtpdec.h:152
int64_t range_start_offset
Definition: rtpdec.h:159
int prev_ret
Fields for packet reordering.
Definition: rtpdec.h:171
RTP/JPEG specific private data.
Definition: rdt.c:83
int64_t last_feedback_time
Definition: rtpdec.h:188
unsigned int last_octet_count
Definition: rtpdec.h:187
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len)
Definition: rtpdec.c:610
RTPPacket * queue
A sorted queue of buffered packets not yet returned.
Definition: rtpdec.h:172
#define RTP_VERSION
Definition: rtp.h:78
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
Definition: rtpdec_latm.c:131
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:3981
RTPDynamicProtocolHandler ff_mp4a_latm_dynamic_handler
Definition: rtpdec_latm.c:165
int num
Numerator.
Definition: rational.h:59
int index
stream index in AVFormatContext
Definition: avformat.h:890
int size
Definition: avcodec.h:1602
RTPDynamicProtocolHandler ff_ms_rtp_asf_pfa_handler
#define RTCP_TX_RATIO_NUM
Definition: rtp.h:82
const RTPDynamicProtocolHandler * handler
Definition: rtpdec.h:191
enum AVMediaType codec_type
Definition: rtp.c:37
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
Definition: rtpdec.c:156
RTPDynamicProtocolHandler ff_jpeg_dynamic_handler
Definition: rtpdec_jpeg.c:382
RTPDynamicProtocolHandler ff_h263_1998_dynamic_handler
Definition: rtpdec_h263.c:92
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:252
static RTPDynamicProtocolHandler l24_dynamic_handler
Definition: rtpdec.c:36
static AVPacket pkt
RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler
Definition: rtpdec_amr.c:195
uint64_t last_rtcp_ntp_time
Definition: rtpdec.h:178
uint32_t cycles
shifted count of sequence number cycles
Definition: rtpdec.h:81
#define RTCP_TX_RATIO_DEN
Definition: rtp.h:83
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
static RTPDynamicProtocolHandler speex_dynamic_handler
Definition: rtpdec.c:54
RTPDynamicProtocolHandler ff_h263_2000_dynamic_handler
Definition: rtpdec_h263.c:100
int avio_open_dyn_buf(AVIOContext **s)
Open a write only memory stream.
Definition: aviobuf.c:1268
static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing, uint16_t *missing_mask)
Definition: rtpdec.c:423
enum AVMediaType codec_type
Definition: rtpdec.h:117
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
Definition: rtpdec.c:521
PayloadContext * dynamic_protocol_context
Definition: rtpdec.h:192
Format I/O context.
Definition: avformat.h:1338
uint64_t first_rtcp_ntp_time
Definition: rtpdec.h:180
static RTPDynamicProtocolHandler gsm_dynamic_handler
Definition: rtpdec.c:42
uint32_t base_seq
base sequence number
Definition: rtpdec.h:82
void ff_srtp_free(struct SRTPContext *s)
Definition: srtp.c:31
uint8_t
#define av_malloc(s)
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
Definition: rtpdec.c:693
RTPDynamicProtocolHandler ff_ilbc_dynamic_handler
Definition: rtpdec_ilbc.c:69
int len
Definition: rtpdec.h:144
int av_packet_from_data(AVPacket *pkt, uint8_t *data, int size)
Initialize a reference-counted packet from av_malloc()ed data.
Definition: avpacket.c:152
static void handler(vbi_event *ev, void *user_data)
RTPDynamicProtocolHandler ff_dv_dynamic_handler
Definition: rtpdec_dv.c:134
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
RTPDynamicProtocolHandler ff_theora_dynamic_handler
Definition: rtpdec_xiph.c:370
uint8_t * data
Definition: avcodec.h:1601
static int flags
Definition: log.c:57
void ff_register_rtp_dynamic_payload_handlers(void)
Definition: rtpdec.c:80
Definition: rtp.h:99
static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
Definition: rtpdec.c:743
RTPDynamicProtocolHandler ff_mpeg_audio_robust_dynamic_handler
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
char hostname[256]
Definition: rtpdec.h:162
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
Definition: aviobuf.c:204
uint32_t expected_prior
packets expected in last interval
Definition: rtpdec.h:86
#define av_log(a,...)
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
int srtp_enabled
Definition: rtpdec.h:164
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:86
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
uint16_t seq
Definition: rtpdec.h:142
#define RTP_FLAG_MARKER
RTP marker bit was set for this packet.
Definition: rtpdec.h:93
#define AVERROR(e)
Definition: error.h:43
Definition: rtp.h:103
int probation
sequence packets till source is valid
Definition: rtpdec.h:84
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
Definition: rtpdec.c:204
GLenum GLint * params
Definition: opengl_enc.c:114
#define RTP_SEQ_MOD
Definition: rtpdec.c:191
RTPDynamicProtocolHandler ff_vorbis_dynamic_handler
Definition: rtpdec_xiph.c:380
RTPDynamicProtocolHandler ff_ms_rtp_asf_pfv_handler
RTPDynamicProtocolHandler ff_ac3_dynamic_handler
Definition: rtpdec_ac3.c:125
RTPDynamicProtocolHandler ff_svq3_dynamic_handler
Definition: rtpdec_svq3.c:113
GLsizei count
Definition: opengl_enc.c:109
DynamicPayloadPacketHandlerProc parse_packet
Parse handler for this dynamic packet.
Definition: rtpdec.h:135
int64_t rtcp_ts_offset
Definition: rtpdec.h:182
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Definition: rtpdec.c:144
uint32_t timestamp
Definition: rtpdec.h:155
uint32_t transit
relative transit time for previous packet
Definition: rtpdec.h:88
uint32_t jitter
estimated jitter.
Definition: rtpdec.h:89
int queue_len
The number of packets in queue.
Definition: rtpdec.h:173
RTPDynamicProtocolHandler ff_qt_rtp_vid_handler
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1394
int void avio_flush(AVIOContext *s)
Force flushing of buffered data.
Definition: aviobuf.c:224
int ff_srtp_decrypt(struct SRTPContext *s, uint8_t *buf, int *lenptr)
Definition: srtp.c:126
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:248
#define FFMIN(a, b)
Definition: common.h:96
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
Definition: avstring.c:213
static RTPDynamicProtocolHandler * rtp_first_dynamic_payload_handler
Definition: rtpdec.c:72
RTPDynamicProtocolHandler ff_g726le_40_dynamic_handler
Definition: rtp.h:98
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
RTPDynamicProtocolHandler ff_g726le_16_dynamic_handler
RTPDynamicProtocolHandler ff_h263_rfc2190_dynamic_handler
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:557
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
Definition: rtpdec.c:453
Stream structure.
Definition: avformat.h:889
uint32_t received
packets received
Definition: rtpdec.h:85
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
RTPDynamicProtocolHandler ff_g726_16_dynamic_handler
int64_t last_rtcp_reception_time
Definition: rtpdec.h:179
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:254
Definition: rtp.h:100
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Definition: rtpdec.c:765
RTPDynamicProtocolHandler ff_vc2hq_dynamic_handler
Definition: rtpdec_vc2hq.c:219
int64_t unwrapped_timestamp
Definition: rtpdec.h:158
uint32_t last_rtcp_timestamp
Definition: rtpdec.h:181
static int has_next_packet(RTPDemuxContext *s)
Definition: rtpdec.c:733
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
Definition: rtpdec.c:74
void avio_w8(AVIOContext *s, int b)
Definition: aviobuf.c:182
RTPDynamicProtocolHandler ff_mp4v_es_dynamic_handler
Definition: rtpdec_mpeg4.c:317
unsigned int octet_count
Definition: rtpdec.h:186
RTPDynamicProtocolHandler ff_h261_dynamic_handler
Definition: rtpdec_h261.c:165
void * buf
Definition: avisynth_c.h:690
Definition: url.h:38
RTPStatistics statistics
Statistics for this stream (used by RTCP receiver reports)
Definition: rtpdec.h:168
uint32_t received_prior
packets received in last interval
Definition: rtpdec.h:87
RTPDynamicProtocolHandler ff_qdm2_dynamic_handler
Definition: rtpdec_qdm2.c:301
uint32_t bad_seq
last bad sequence number + 1
Definition: rtpdec.h:83
AVMediaType
Definition: avutil.h:193
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
Definition: rtpdec.c:738
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count)
some rtp servers assume client is dead if they don't hear from them...
Definition: rtpdec.c:282
#define s1
Definition: regdef.h:38
int ff_parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *p, int(*parse_fmtp)(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value))
Definition: rtpdec.c:872
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
This was the second switch in rtp_parse packet.
Definition: rtpdec.c:575
uint16_t max_seq
highest sequence number seen
Definition: rtpdec.h:80
RTPDynamicProtocolHandler ff_mpegts_dynamic_handler
Definition: rtpdec_mpegts.c:92
RTPDynamicProtocolHandler ff_g726le_32_dynamic_handler
RTPDynamicProtocolHandler ff_g726_32_dynamic_handler
RTPDynamicProtocolHandler ff_qt_rtp_aud_handler
const char * enc_name
Definition: rtpdec.h:116
uint8_t * buf
Definition: rtpdec.h:143
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
void avio_wb16(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:452
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
Definition: time.c:56
RTPDynamicProtocolHandler ff_mpeg_audio_dynamic_handler
Definition: rtpdec_mpeg12.c:51
int sample_rate
Audio only.
Definition: avcodec.h:4091
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_RB64
Definition: bytestream.h:87
RTPDynamicProtocolHandler ff_mpeg_video_dynamic_handler
Definition: rtpdec_mpeg12.c:59
Main libavformat public API header.
struct RTPPacket * next
Definition: rtpdec.h:146
uint32_t ssrc
Definition: rtpdec.h:153
static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler
Definition: rtpdec.c:48
int64_t recvtime
Definition: rtpdec.h:145
RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Definition: rtpdec.c:131
RTPDynamicProtocolHandler ff_g726_40_dynamic_handler
raw UTF-8 text
Definition: avcodec.h:606
RTPDynamicProtocolHandler ff_qcelp_dynamic_handler
Definition: rtpdec_qcelp.c:212
void av_init_packet(AVPacket *pkt)
Initialize optional fields of a packet with default values.
Definition: avpacket.c:33
int den
Denominator.
Definition: rational.h:60
RTPDynamicProtocolHandler ff_hevc_dynamic_handler
Definition: rtpdec_hevc.c:350
Definition: rtp.h:97
int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
Close the dynamic buffer and make a packet from it.
Definition: rtpdec.c:910
uint32_t base_timestamp
Definition: rtpdec.h:156
RTPDynamicProtocolHandler ff_g726_24_dynamic_handler
RTPDynamicProtocolHandler ff_h264_dynamic_handler
Definition: rtpdec_h264.c:411
static av_always_inline int diff(const uint32_t a, const uint32_t b)
#define av_free(p)
RTPDynamicProtocolHandler ff_vp9_dynamic_handler
Definition: rtpdec_vp9.c:333
as in Berlin toast format
Definition: avcodec.h:534
int len
int ff_srtp_set_crypto(struct SRTPContext *s, const char *suite, const char *params)
Definition: srtp.c:65
void ff_rtp_parse_close(RTPDemuxContext *s)
Definition: rtpdec.c:865
static RTPDynamicProtocolHandler t140_dynamic_handler
Definition: rtpdec.c:66
static void stats(const struct CachedBuf *in, int n_in, unsigned *_max, unsigned *_sum)
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
Definition: avcodec.h:1600
void avio_wb32(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:354
#define RTP_NOTS_VALUE
Definition: rtpdec.h:40
static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
Definition: rtpdec.c:706
#define av_freep(p)
unbuffered private I/O API
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
Definition: rtpdec.c:193
AVCodecParameters * codecpar
Definition: avformat.h:1241
int(* need_keyframe)(PayloadContext *context)
Definition: rtpdec.h:136
static RTPDynamicProtocolHandler opus_dynamic_handler
Definition: rtpdec.c:60
int stream_index
Definition: avcodec.h:1603
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:926
RTPDynamicProtocolHandler ff_quicktime_rtp_vid_handler
RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler
Definition: rtpdec_mpeg4.c:326
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
Definition: rtpdec.c:852
RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler
Definition: rtpdec_amr.c:185
AVStream * st
Definition: rtpdec.h:151
This structure stores compressed data.
Definition: avcodec.h:1578
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
Definition: rtpdec.c:218
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1594
RTPDynamicProtocolHandler ff_vp8_dynamic_handler
Definition: rtpdec_vp8.c:279
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:242
#define MIN_FEEDBACK_INTERVAL
Definition: rtpdec.c:34
#define av_unused
Definition: attributes.h:126
struct SRTPContext srtp
Definition: rtpdec.h:165
const char * name
Definition: opengl_enc.c:103
struct RTPDynamicProtocolHandler * next
Definition: rtpdec.h:138