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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/time.h"
25 #include "libavcodec/get_bits.h"
26 #include "avformat.h"
27 #include "network.h"
28 #include "srtp.h"
29 #include "url.h"
30 #include "rtpdec.h"
31 #include "rtpdec_formats.h"
32 
33 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
34 
36  .enc_name = "X-MP3-draft-00",
37  .codec_type = AVMEDIA_TYPE_AUDIO,
38  .codec_id = AV_CODEC_ID_MP3ADU,
39 };
40 
42  .enc_name = "speex",
43  .codec_type = AVMEDIA_TYPE_AUDIO,
44  .codec_id = AV_CODEC_ID_SPEEX,
45 };
46 
48  .enc_name = "opus",
49  .codec_type = AVMEDIA_TYPE_AUDIO,
50  .codec_id = AV_CODEC_ID_OPUS,
51 };
52 
54 
56 {
58  rtp_first_dynamic_payload_handler = handler;
59 }
60 
62 {
93  ff_register_dynamic_payload_handler(&opus_dynamic_handler);
94  ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
95  ff_register_dynamic_payload_handler(&speex_dynamic_handler);
96 }
97 
100 {
101  RTPDynamicProtocolHandler *handler;
102  for (handler = rtp_first_dynamic_payload_handler;
103  handler; handler = handler->next)
104  if (!av_strcasecmp(name, handler->enc_name) &&
105  codec_type == handler->codec_type)
106  return handler;
107  return NULL;
108 }
109 
111  enum AVMediaType codec_type)
112 {
113  RTPDynamicProtocolHandler *handler;
114  for (handler = rtp_first_dynamic_payload_handler;
115  handler; handler = handler->next)
116  if (handler->static_payload_id && handler->static_payload_id == id &&
117  codec_type == handler->codec_type)
118  return handler;
119  return NULL;
120 }
121 
122 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
123  int len)
124 {
125  int payload_len;
126  while (len >= 4) {
127  payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
128 
129  switch (buf[1]) {
130  case RTCP_SR:
131  if (payload_len < 20) {
133  "Invalid length for RTCP SR packet\n");
134  return AVERROR_INVALIDDATA;
135  }
136 
138  s->last_rtcp_ntp_time = AV_RB64(buf + 8);
139  s->last_rtcp_timestamp = AV_RB32(buf + 16);
142  if (!s->base_timestamp)
145  }
146 
147  break;
148  case RTCP_BYE:
149  return -RTCP_BYE;
150  }
151 
152  buf += payload_len;
153  len -= payload_len;
154  }
155  return -1;
156 }
157 
158 #define RTP_SEQ_MOD (1 << 16)
159 
160 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
161 {
162  memset(s, 0, sizeof(RTPStatistics));
163  s->max_seq = base_sequence;
164  s->probation = 1;
165 }
166 
167 /*
168  * Called whenever there is a large jump in sequence numbers,
169  * or when they get out of probation...
170  */
171 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
172 {
173  s->max_seq = seq;
174  s->cycles = 0;
175  s->base_seq = seq - 1;
176  s->bad_seq = RTP_SEQ_MOD + 1;
177  s->received = 0;
178  s->expected_prior = 0;
179  s->received_prior = 0;
180  s->jitter = 0;
181  s->transit = 0;
182 }
183 
184 /* Returns 1 if we should handle this packet. */
185 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
186 {
187  uint16_t udelta = seq - s->max_seq;
188  const int MAX_DROPOUT = 3000;
189  const int MAX_MISORDER = 100;
190  const int MIN_SEQUENTIAL = 2;
191 
192  /* source not valid until MIN_SEQUENTIAL packets with sequence
193  * seq. numbers have been received */
194  if (s->probation) {
195  if (seq == s->max_seq + 1) {
196  s->probation--;
197  s->max_seq = seq;
198  if (s->probation == 0) {
199  rtp_init_sequence(s, seq);
200  s->received++;
201  return 1;
202  }
203  } else {
204  s->probation = MIN_SEQUENTIAL - 1;
205  s->max_seq = seq;
206  }
207  } else if (udelta < MAX_DROPOUT) {
208  // in order, with permissible gap
209  if (seq < s->max_seq) {
210  // sequence number wrapped; count another 64k cycles
211  s->cycles += RTP_SEQ_MOD;
212  }
213  s->max_seq = seq;
214  } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
215  // sequence made a large jump...
216  if (seq == s->bad_seq) {
217  /* two sequential packets -- assume that the other side
218  * restarted without telling us; just resync. */
219  rtp_init_sequence(s, seq);
220  } else {
221  s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
222  return 0;
223  }
224  } else {
225  // duplicate or reordered packet...
226  }
227  s->received++;
228  return 1;
229 }
230 
231 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
232  uint32_t arrival_timestamp)
233 {
234  // Most of this is pretty straight from RFC 3550 appendix A.8
235  uint32_t transit = arrival_timestamp - sent_timestamp;
236  uint32_t prev_transit = s->transit;
237  int32_t d = transit - prev_transit;
238  // Doing the FFABS() call directly on the "transit - prev_transit"
239  // expression doesn't work, since it's an unsigned expression. Doing the
240  // transit calculation in unsigned is desired though, since it most
241  // probably will need to wrap around.
242  d = FFABS(d);
243  s->transit = transit;
244  if (!prev_transit)
245  return;
246  s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
247 }
248 
250  AVIOContext *avio, int count)
251 {
252  AVIOContext *pb;
253  uint8_t *buf;
254  int len;
255  int rtcp_bytes;
257  uint32_t lost;
258  uint32_t extended_max;
259  uint32_t expected_interval;
260  uint32_t received_interval;
261  int32_t lost_interval;
262  uint32_t expected;
263  uint32_t fraction;
264 
265  if ((!fd && !avio) || (count < 1))
266  return -1;
267 
268  /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
269  /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
270  s->octet_count += count;
271  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
273  rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
274  if (rtcp_bytes < 28)
275  return -1;
277 
278  if (!fd)
279  pb = avio;
280  else if (avio_open_dyn_buf(&pb) < 0)
281  return -1;
282 
283  // Receiver Report
284  avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
285  avio_w8(pb, RTCP_RR);
286  avio_wb16(pb, 7); /* length in words - 1 */
287  // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
288  avio_wb32(pb, s->ssrc + 1);
289  avio_wb32(pb, s->ssrc); // server SSRC
290  // some placeholders we should really fill...
291  // RFC 1889/p64
292  extended_max = stats->cycles + stats->max_seq;
293  expected = extended_max - stats->base_seq;
294  lost = expected - stats->received;
295  lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
296  expected_interval = expected - stats->expected_prior;
297  stats->expected_prior = expected;
298  received_interval = stats->received - stats->received_prior;
299  stats->received_prior = stats->received;
300  lost_interval = expected_interval - received_interval;
301  if (expected_interval == 0 || lost_interval <= 0)
302  fraction = 0;
303  else
304  fraction = (lost_interval << 8) / expected_interval;
305 
306  fraction = (fraction << 24) | lost;
307 
308  avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
309  avio_wb32(pb, extended_max); /* max sequence received */
310  avio_wb32(pb, stats->jitter >> 4); /* jitter */
311 
313  avio_wb32(pb, 0); /* last SR timestamp */
314  avio_wb32(pb, 0); /* delay since last SR */
315  } else {
316  uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
317  uint32_t delay_since_last = av_rescale(av_gettime() - s->last_rtcp_reception_time,
318  65536, AV_TIME_BASE);
319 
320  avio_wb32(pb, middle_32_bits); /* last SR timestamp */
321  avio_wb32(pb, delay_since_last); /* delay since last SR */
322  }
323 
324  // CNAME
325  avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
326  avio_w8(pb, RTCP_SDES);
327  len = strlen(s->hostname);
328  avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
329  avio_wb32(pb, s->ssrc + 1);
330  avio_w8(pb, 0x01);
331  avio_w8(pb, len);
332  avio_write(pb, s->hostname, len);
333  avio_w8(pb, 0); /* END */
334  // padding
335  for (len = (7 + len) % 4; len % 4; len++)
336  avio_w8(pb, 0);
337 
338  avio_flush(pb);
339  if (!fd)
340  return 0;
341  len = avio_close_dyn_buf(pb, &buf);
342  if ((len > 0) && buf) {
343  int av_unused result;
344  av_dlog(s->ic, "sending %d bytes of RR\n", len);
345  result = ffurl_write(fd, buf, len);
346  av_dlog(s->ic, "result from ffurl_write: %d\n", result);
347  av_free(buf);
348  }
349  return 0;
350 }
351 
353 {
354  AVIOContext *pb;
355  uint8_t *buf;
356  int len;
357 
358  /* Send a small RTP packet */
359  if (avio_open_dyn_buf(&pb) < 0)
360  return;
361 
362  avio_w8(pb, (RTP_VERSION << 6));
363  avio_w8(pb, 0); /* Payload type */
364  avio_wb16(pb, 0); /* Seq */
365  avio_wb32(pb, 0); /* Timestamp */
366  avio_wb32(pb, 0); /* SSRC */
367 
368  avio_flush(pb);
369  len = avio_close_dyn_buf(pb, &buf);
370  if ((len > 0) && buf)
371  ffurl_write(rtp_handle, buf, len);
372  av_free(buf);
373 
374  /* Send a minimal RTCP RR */
375  if (avio_open_dyn_buf(&pb) < 0)
376  return;
377 
378  avio_w8(pb, (RTP_VERSION << 6));
379  avio_w8(pb, RTCP_RR); /* receiver report */
380  avio_wb16(pb, 1); /* length in words - 1 */
381  avio_wb32(pb, 0); /* our own SSRC */
382 
383  avio_flush(pb);
384  len = avio_close_dyn_buf(pb, &buf);
385  if ((len > 0) && buf)
386  ffurl_write(rtp_handle, buf, len);
387  av_free(buf);
388 }
389 
390 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
391  uint16_t *missing_mask)
392 {
393  int i;
394  uint16_t next_seq = s->seq + 1;
395  RTPPacket *pkt = s->queue;
396 
397  if (!pkt || pkt->seq == next_seq)
398  return 0;
399 
400  *missing_mask = 0;
401  for (i = 1; i <= 16; i++) {
402  uint16_t missing_seq = next_seq + i;
403  while (pkt) {
404  int16_t diff = pkt->seq - missing_seq;
405  if (diff >= 0)
406  break;
407  pkt = pkt->next;
408  }
409  if (!pkt)
410  break;
411  if (pkt->seq == missing_seq)
412  continue;
413  *missing_mask |= 1 << (i - 1);
414  }
415 
416  *first_missing = next_seq;
417  return 1;
418 }
419 
421  AVIOContext *avio)
422 {
423  int len, need_keyframe, missing_packets;
424  AVIOContext *pb;
425  uint8_t *buf;
426  int64_t now;
427  uint16_t first_missing = 0, missing_mask = 0;
428 
429  if (!fd && !avio)
430  return -1;
431 
432  need_keyframe = s->handler && s->handler->need_keyframe &&
434  missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
435 
436  if (!need_keyframe && !missing_packets)
437  return 0;
438 
439  /* Send new feedback if enough time has elapsed since the last
440  * feedback packet. */
441 
442  now = av_gettime();
443  if (s->last_feedback_time &&
445  return 0;
446  s->last_feedback_time = now;
447 
448  if (!fd)
449  pb = avio;
450  else if (avio_open_dyn_buf(&pb) < 0)
451  return -1;
452 
453  if (need_keyframe) {
454  avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
455  avio_w8(pb, RTCP_PSFB);
456  avio_wb16(pb, 2); /* length in words - 1 */
457  // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
458  avio_wb32(pb, s->ssrc + 1);
459  avio_wb32(pb, s->ssrc); // server SSRC
460  }
461 
462  if (missing_packets) {
463  avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
464  avio_w8(pb, RTCP_RTPFB);
465  avio_wb16(pb, 3); /* length in words - 1 */
466  avio_wb32(pb, s->ssrc + 1);
467  avio_wb32(pb, s->ssrc); // server SSRC
468 
469  avio_wb16(pb, first_missing);
470  avio_wb16(pb, missing_mask);
471  }
472 
473  avio_flush(pb);
474  if (!fd)
475  return 0;
476  len = avio_close_dyn_buf(pb, &buf);
477  if (len > 0 && buf) {
478  ffurl_write(fd, buf, len);
479  av_free(buf);
480  }
481  return 0;
482 }
483 
484 /**
485  * open a new RTP parse context for stream 'st'. 'st' can be NULL for
486  * MPEG2-TS streams.
487  */
489  int payload_type, int queue_size)
490 {
491  RTPDemuxContext *s;
492 
493  s = av_mallocz(sizeof(RTPDemuxContext));
494  if (!s)
495  return NULL;
496  s->payload_type = payload_type;
499  s->ic = s1;
500  s->st = st;
501  s->queue_size = queue_size;
503  if (st) {
504  switch (st->codec->codec_id) {
506  /* According to RFC 3551, the stream clock rate is 8000
507  * even if the sample rate is 16000. */
508  if (st->codec->sample_rate == 8000)
509  st->codec->sample_rate = 16000;
510  break;
511  default:
512  break;
513  }
514  }
515  // needed to send back RTCP RR in RTSP sessions
516  gethostname(s->hostname, sizeof(s->hostname));
517  return s;
518 }
519 
521  RTPDynamicProtocolHandler *handler)
522 {
523  s->dynamic_protocol_context = ctx;
524  s->handler = handler;
525 }
526 
527 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
528  const char *params)
529 {
530  if (!ff_srtp_set_crypto(&s->srtp, suite, params))
531  s->srtp_enabled = 1;
532 }
533 
534 /**
535  * This was the second switch in rtp_parse packet.
536  * Normalizes time, if required, sets stream_index, etc.
537  */
538 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
539 {
540  if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
541  return; /* Timestamp already set by depacketizer */
542  if (timestamp == RTP_NOTS_VALUE)
543  return;
544 
545  if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
546  int64_t addend;
547  int delta_timestamp;
548 
549  /* compute pts from timestamp with received ntp_time */
550  delta_timestamp = timestamp - s->last_rtcp_timestamp;
551  /* convert to the PTS timebase */
553  s->st->time_base.den,
554  (uint64_t) s->st->time_base.num << 32);
555  pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
556  delta_timestamp;
557  return;
558  }
559 
560  if (!s->base_timestamp)
561  s->base_timestamp = timestamp;
562  /* assume that the difference is INT32_MIN < x < INT32_MAX,
563  * but allow the first timestamp to exceed INT32_MAX */
564  if (!s->timestamp)
565  s->unwrapped_timestamp += timestamp;
566  else
567  s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
568  s->timestamp = timestamp;
570  s->base_timestamp;
571 }
572 
574  const uint8_t *buf, int len)
575 {
576  unsigned int ssrc;
577  int payload_type, seq, flags = 0;
578  int ext, csrc;
579  AVStream *st;
580  uint32_t timestamp;
581  int rv = 0;
582 
583  csrc = buf[0] & 0x0f;
584  ext = buf[0] & 0x10;
585  payload_type = buf[1] & 0x7f;
586  if (buf[1] & 0x80)
587  flags |= RTP_FLAG_MARKER;
588  seq = AV_RB16(buf + 2);
589  timestamp = AV_RB32(buf + 4);
590  ssrc = AV_RB32(buf + 8);
591  /* store the ssrc in the RTPDemuxContext */
592  s->ssrc = ssrc;
593 
594  /* NOTE: we can handle only one payload type */
595  if (s->payload_type != payload_type)
596  return -1;
597 
598  st = s->st;
599  // only do something with this if all the rtp checks pass...
600  if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
601  av_log(st ? st->codec : NULL, AV_LOG_ERROR,
602  "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
603  payload_type, seq, ((s->seq + 1) & 0xffff));
604  return -1;
605  }
606 
607  if (buf[0] & 0x20) {
608  int padding = buf[len - 1];
609  if (len >= 12 + padding)
610  len -= padding;
611  }
612 
613  s->seq = seq;
614  len -= 12;
615  buf += 12;
616 
617  len -= 4 * csrc;
618  buf += 4 * csrc;
619  if (len < 0)
620  return AVERROR_INVALIDDATA;
621 
622  /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
623  if (ext) {
624  if (len < 4)
625  return -1;
626  /* calculate the header extension length (stored as number
627  * of 32-bit words) */
628  ext = (AV_RB16(buf + 2) + 1) << 2;
629 
630  if (len < ext)
631  return -1;
632  // skip past RTP header extension
633  len -= ext;
634  buf += ext;
635  }
636 
637  if (s->handler && s->handler->parse_packet) {
639  s->st, pkt, &timestamp, buf, len, seq,
640  flags);
641  } else if (st) {
642  if ((rv = av_new_packet(pkt, len)) < 0)
643  return rv;
644  memcpy(pkt->data, buf, len);
645  pkt->stream_index = st->index;
646  } else {
647  return AVERROR(EINVAL);
648  }
649 
650  // now perform timestamp things....
651  finalize_packet(s, pkt, timestamp);
652 
653  return rv;
654 }
655 
657 {
658  while (s->queue) {
659  RTPPacket *next = s->queue->next;
660  av_free(s->queue->buf);
661  av_free(s->queue);
662  s->queue = next;
663  }
664  s->seq = 0;
665  s->queue_len = 0;
666  s->prev_ret = 0;
667 }
668 
669 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
670 {
671  uint16_t seq = AV_RB16(buf + 2);
672  RTPPacket **cur = &s->queue, *packet;
673 
674  /* Find the correct place in the queue to insert the packet */
675  while (*cur) {
676  int16_t diff = seq - (*cur)->seq;
677  if (diff < 0)
678  break;
679  cur = &(*cur)->next;
680  }
681 
682  packet = av_mallocz(sizeof(*packet));
683  if (!packet)
684  return;
685  packet->recvtime = av_gettime();
686  packet->seq = seq;
687  packet->len = len;
688  packet->buf = buf;
689  packet->next = *cur;
690  *cur = packet;
691  s->queue_len++;
692 }
693 
695 {
696  return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
697 }
698 
700 {
701  return s->queue ? s->queue->recvtime : 0;
702 }
703 
705 {
706  int rv;
707  RTPPacket *next;
708 
709  if (s->queue_len <= 0)
710  return -1;
711 
712  if (!has_next_packet(s))
713  av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
714  "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
715 
716  /* Parse the first packet in the queue, and dequeue it */
717  rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
718  next = s->queue->next;
719  av_free(s->queue->buf);
720  av_free(s->queue);
721  s->queue = next;
722  s->queue_len--;
723  return rv;
724 }
725 
727  uint8_t **bufptr, int len)
728 {
729  uint8_t *buf = bufptr ? *bufptr : NULL;
730  int flags = 0;
731  uint32_t timestamp;
732  int rv = 0;
733 
734  if (!buf) {
735  /* If parsing of the previous packet actually returned 0 or an error,
736  * there's nothing more to be parsed from that packet, but we may have
737  * indicated that we can return the next enqueued packet. */
738  if (s->prev_ret <= 0)
739  return rtp_parse_queued_packet(s, pkt);
740  /* return the next packets, if any */
741  if (s->handler && s->handler->parse_packet) {
742  /* timestamp should be overwritten by parse_packet, if not,
743  * the packet is left with pts == AV_NOPTS_VALUE */
744  timestamp = RTP_NOTS_VALUE;
746  s->st, pkt, &timestamp, NULL, 0, 0,
747  flags);
748  finalize_packet(s, pkt, timestamp);
749  return rv;
750  }
751  }
752 
753  if (len < 12)
754  return -1;
755 
756  if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
757  return -1;
758  if (RTP_PT_IS_RTCP(buf[1])) {
759  return rtcp_parse_packet(s, buf, len);
760  }
761 
762  if (s->st) {
763  int64_t received = av_gettime();
764  uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
765  s->st->time_base);
766  timestamp = AV_RB32(buf + 4);
767  // Calculate the jitter immediately, before queueing the packet
768  // into the reordering queue.
769  rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
770  }
771 
772  if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
773  /* First packet, or no reordering */
774  return rtp_parse_packet_internal(s, pkt, buf, len);
775  } else {
776  uint16_t seq = AV_RB16(buf + 2);
777  int16_t diff = seq - s->seq;
778  if (diff < 0) {
779  /* Packet older than the previously emitted one, drop */
780  av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
781  "RTP: dropping old packet received too late\n");
782  return -1;
783  } else if (diff <= 1) {
784  /* Correct packet */
785  rv = rtp_parse_packet_internal(s, pkt, buf, len);
786  return rv;
787  } else {
788  /* Still missing some packet, enqueue this one. */
789  enqueue_packet(s, buf, len);
790  *bufptr = NULL;
791  /* Return the first enqueued packet if the queue is full,
792  * even if we're missing something */
793  if (s->queue_len >= s->queue_size)
794  return rtp_parse_queued_packet(s, pkt);
795  return -1;
796  }
797  }
798 }
799 
800 /**
801  * Parse an RTP or RTCP packet directly sent as a buffer.
802  * @param s RTP parse context.
803  * @param pkt returned packet
804  * @param bufptr pointer to the input buffer or NULL to read the next packets
805  * @param len buffer len
806  * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
807  * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
808  */
810  uint8_t **bufptr, int len)
811 {
812  int rv;
813  if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
814  return -1;
815  rv = rtp_parse_one_packet(s, pkt, bufptr, len);
816  s->prev_ret = rv;
817  while (rv == AVERROR(EAGAIN) && has_next_packet(s))
818  rv = rtp_parse_queued_packet(s, pkt);
819  return rv ? rv : has_next_packet(s);
820 }
821 
823 {
825  ff_srtp_free(&s->srtp);
826  av_free(s);
827 }
828 
829 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
830  int (*parse_fmtp)(AVStream *stream,
831  PayloadContext *data,
832  char *attr, char *value))
833 {
834  char attr[256];
835  char *value;
836  int res;
837  int value_size = strlen(p) + 1;
838 
839  if (!(value = av_malloc(value_size))) {
840  av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
841  return AVERROR(ENOMEM);
842  }
843 
844  // remove protocol identifier
845  while (*p && *p == ' ')
846  p++; // strip spaces
847  while (*p && *p != ' ')
848  p++; // eat protocol identifier
849  while (*p && *p == ' ')
850  p++; // strip trailing spaces
851 
852  while (ff_rtsp_next_attr_and_value(&p,
853  attr, sizeof(attr),
854  value, value_size)) {
855  res = parse_fmtp(stream, data, attr, value);
856  if (res < 0 && res != AVERROR_PATCHWELCOME) {
857  av_free(value);
858  return res;
859  }
860  }
861  av_free(value);
862  return 0;
863 }
864 
865 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
866 {
867  av_init_packet(pkt);
868 
869  pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
870  pkt->stream_index = stream_idx;
872  *dyn_buf = NULL;
873  return pkt->size;
874 }