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rtpenc.c
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1 /*
2  * RTP output 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 "avformat.h"
23 #include "mpegts.h"
24 #include "internal.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/random_seed.h"
27 #include "libavutil/opt.h"
28 
29 #include "rtpenc.h"
30 
31 static const AVOption options[] = {
33  { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
34  { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
35  { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
36  { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
37  { NULL },
38 };
39 
40 static const AVClass rtp_muxer_class = {
41  .class_name = "RTP muxer",
42  .item_name = av_default_item_name,
43  .option = options,
44  .version = LIBAVUTIL_VERSION_INT,
45 };
46 
47 #define RTCP_SR_SIZE 28
48 
49 static int is_supported(enum AVCodecID id)
50 {
51  switch(id) {
52  case AV_CODEC_ID_H261:
53  case AV_CODEC_ID_H263:
54  case AV_CODEC_ID_H263P:
55  case AV_CODEC_ID_H264:
56  case AV_CODEC_ID_HEVC:
59  case AV_CODEC_ID_MPEG4:
60  case AV_CODEC_ID_AAC:
61  case AV_CODEC_ID_MP2:
62  case AV_CODEC_ID_MP3:
65  case AV_CODEC_ID_PCM_S8:
70  case AV_CODEC_ID_PCM_U8:
72  case AV_CODEC_ID_AMR_NB:
73  case AV_CODEC_ID_AMR_WB:
74  case AV_CODEC_ID_VORBIS:
75  case AV_CODEC_ID_THEORA:
76  case AV_CODEC_ID_VP8:
79  case AV_CODEC_ID_ILBC:
80  case AV_CODEC_ID_MJPEG:
81  case AV_CODEC_ID_SPEEX:
82  case AV_CODEC_ID_OPUS:
83  return 1;
84  default:
85  return 0;
86  }
87 }
88 
90 {
91  RTPMuxContext *s = s1->priv_data;
92  int n, ret = AVERROR(EINVAL);
93  AVStream *st;
94 
95  if (s1->nb_streams != 1) {
96  av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
97  return AVERROR(EINVAL);
98  }
99  st = s1->streams[0];
100  if (!is_supported(st->codec->codec_id)) {
101  av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
102 
103  return -1;
104  }
105 
106  if (s->payload_type < 0) {
107  /* Re-validate non-dynamic payload types */
108  if (st->id < RTP_PT_PRIVATE)
109  st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
110 
111  s->payload_type = st->id;
112  } else {
113  /* private option takes priority */
114  st->id = s->payload_type;
115  }
116 
118  s->timestamp = s->base_timestamp;
119  s->cur_timestamp = 0;
120  if (!s->ssrc)
121  s->ssrc = av_get_random_seed();
122  s->first_packet = 1;
125  /* Round the NTP time to whole milliseconds. */
126  s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
128  // Pick a random sequence start number, but in the lower end of the
129  // available range, so that any wraparound doesn't happen immediately.
130  // (Immediate wraparound would be an issue for SRTP.)
131  if (s->seq < 0) {
132  if (s1->flags & AVFMT_FLAG_BITEXACT) {
133  s->seq = 0;
134  } else
135  s->seq = av_get_random_seed() & 0x0fff;
136  } else
137  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
138 
139  if (s1->packet_size) {
140  if (s1->pb->max_packet_size)
141  s1->packet_size = FFMIN(s1->packet_size,
142  s1->pb->max_packet_size);
143  } else
144  s1->packet_size = s1->pb->max_packet_size;
145  if (s1->packet_size <= 12) {
146  av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
147  return AVERROR(EIO);
148  }
149  s->buf = av_malloc(s1->packet_size);
150  if (!s->buf) {
151  return AVERROR(ENOMEM);
152  }
153  s->max_payload_size = s1->packet_size - 12;
154 
155  s->max_frames_per_packet = 0;
156  if (s1->max_delay > 0) {
157  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
159  if (!frame_size)
160  frame_size = st->codec->frame_size;
161  if (frame_size == 0) {
162  av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
163  } else {
167  (AVRational){ frame_size, st->codec->sample_rate },
168  AV_ROUND_DOWN);
169  }
170  }
171  if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
172  /* FIXME: We should round down here... */
173  if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
175  (AVRational){1, 1000000},
176  av_inv_q(st->avg_frame_rate));
177  } else
178  s->max_frames_per_packet = 1;
179  }
180  }
181 
182  avpriv_set_pts_info(st, 32, 1, 90000);
183  switch(st->codec->codec_id) {
184  case AV_CODEC_ID_MP2:
185  case AV_CODEC_ID_MP3:
186  s->buf_ptr = s->buf + 4;
187  break;
190  break;
191  case AV_CODEC_ID_MPEG2TS:
192  n = s->max_payload_size / TS_PACKET_SIZE;
193  if (n < 1)
194  n = 1;
195  s->max_payload_size = n * TS_PACKET_SIZE;
196  s->buf_ptr = s->buf;
197  break;
198  case AV_CODEC_ID_H261:
199  if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
201  "Packetizing H261 is experimental and produces incorrect "
202  "packetization for cases where GOBs don't fit into packets "
203  "(even though most receivers may handle it just fine). "
204  "Please set -f_strict experimental in order to enable it.\n");
206  goto fail;
207  }
208  break;
209  case AV_CODEC_ID_H264:
210  /* check for H.264 MP4 syntax */
211  if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
212  s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
213  }
214  break;
215  case AV_CODEC_ID_HEVC:
216  /* Only check for the standardized hvcC version of extradata, keeping
217  * things simple and similar to the avcC/H264 case above, instead
218  * of trying to handle the pre-standardization versions (as in
219  * libavcodec/hevc.c). */
220  if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
221  s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
222  }
223  break;
224  case AV_CODEC_ID_VORBIS:
225  case AV_CODEC_ID_THEORA:
226  if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
227  s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
228  s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
229  s->num_frames = 0;
230  goto defaultcase;
232  /* Due to a historical error, the clock rate for G722 in RTP is
233  * 8000, even if the sample rate is 16000. See RFC 3551. */
234  avpriv_set_pts_info(st, 32, 1, 8000);
235  break;
236  case AV_CODEC_ID_OPUS:
237  if (st->codec->channels > 2) {
238  av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
239  goto fail;
240  }
241  /* The opus RTP RFC says that all opus streams should use 48000 Hz
242  * as clock rate, since all opus sample rates can be expressed in
243  * this clock rate, and sample rate changes on the fly are supported. */
244  avpriv_set_pts_info(st, 32, 1, 48000);
245  break;
246  case AV_CODEC_ID_ILBC:
247  if (st->codec->block_align != 38 && st->codec->block_align != 50) {
248  av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
249  goto fail;
250  }
251  if (!s->max_frames_per_packet)
252  s->max_frames_per_packet = 1;
253  s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
254  s->max_payload_size / st->codec->block_align);
255  goto defaultcase;
256  case AV_CODEC_ID_AMR_NB:
257  case AV_CODEC_ID_AMR_WB:
258  if (!s->max_frames_per_packet)
259  s->max_frames_per_packet = 12;
260  if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
261  n = 31;
262  else
263  n = 61;
264  /* max_header_toc_size + the largest AMR payload must fit */
265  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
266  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
267  goto fail;
268  }
269  if (st->codec->channels != 1) {
270  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
271  goto fail;
272  }
273  s->num_frames = 0;
274  goto defaultcase;
275  case AV_CODEC_ID_AAC:
276  s->num_frames = 0;
277  goto defaultcase;
278  default:
279 defaultcase:
280  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
281  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
282  }
283  s->buf_ptr = s->buf;
284  break;
285  }
286 
287  return 0;
288 
289 fail:
290  av_freep(&s->buf);
291  return ret;
292 }
293 
294 /* send an rtcp sender report packet */
295 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
296 {
297  RTPMuxContext *s = s1->priv_data;
298  uint32_t rtp_ts;
299 
300  av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
301 
302  s->last_rtcp_ntp_time = ntp_time;
303  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
304  s1->streams[0]->time_base) + s->base_timestamp;
305  avio_w8(s1->pb, RTP_VERSION << 6);
306  avio_w8(s1->pb, RTCP_SR);
307  avio_wb16(s1->pb, 6); /* length in words - 1 */
308  avio_wb32(s1->pb, s->ssrc);
309  avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
310  avio_wb32(s1->pb, rtp_ts);
311  avio_wb32(s1->pb, s->packet_count);
312  avio_wb32(s1->pb, s->octet_count);
313 
314  if (s->cname) {
315  int len = FFMIN(strlen(s->cname), 255);
316  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
317  avio_w8(s1->pb, RTCP_SDES);
318  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
319 
320  avio_wb32(s1->pb, s->ssrc);
321  avio_w8(s1->pb, 0x01); /* CNAME */
322  avio_w8(s1->pb, len);
323  avio_write(s1->pb, s->cname, len);
324  avio_w8(s1->pb, 0); /* END */
325  for (len = (7 + len) % 4; len % 4; len++)
326  avio_w8(s1->pb, 0);
327  }
328 
329  if (bye) {
330  avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
331  avio_w8(s1->pb, RTCP_BYE);
332  avio_wb16(s1->pb, 1); /* length in words - 1 */
333  avio_wb32(s1->pb, s->ssrc);
334  }
335 
336  avio_flush(s1->pb);
337 }
338 
339 /* send an rtp packet. sequence number is incremented, but the caller
340  must update the timestamp itself */
341 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
342 {
343  RTPMuxContext *s = s1->priv_data;
344 
345  av_dlog(s1, "rtp_send_data size=%d\n", len);
346 
347  /* build the RTP header */
348  avio_w8(s1->pb, RTP_VERSION << 6);
349  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
350  avio_wb16(s1->pb, s->seq);
351  avio_wb32(s1->pb, s->timestamp);
352  avio_wb32(s1->pb, s->ssrc);
353 
354  avio_write(s1->pb, buf1, len);
355  avio_flush(s1->pb);
356 
357  s->seq = (s->seq + 1) & 0xffff;
358  s->octet_count += len;
359  s->packet_count++;
360 }
361 
362 /* send an integer number of samples and compute time stamp and fill
363  the rtp send buffer before sending. */
365  const uint8_t *buf1, int size, int sample_size_bits)
366 {
367  RTPMuxContext *s = s1->priv_data;
368  int len, max_packet_size, n;
369  /* Calculate the number of bytes to get samples aligned on a byte border */
370  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
371 
372  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
373  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
374  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
375  return AVERROR(EINVAL);
376  n = 0;
377  while (size > 0) {
378  s->buf_ptr = s->buf;
379  len = FFMIN(max_packet_size, size);
380 
381  /* copy data */
382  memcpy(s->buf_ptr, buf1, len);
383  s->buf_ptr += len;
384  buf1 += len;
385  size -= len;
386  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
387  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
388  n += (s->buf_ptr - s->buf);
389  }
390  return 0;
391 }
392 
394  const uint8_t *buf1, int size)
395 {
396  RTPMuxContext *s = s1->priv_data;
397  int len, count, max_packet_size;
398 
399  max_packet_size = s->max_payload_size;
400 
401  /* test if we must flush because not enough space */
402  len = (s->buf_ptr - s->buf);
403  if ((len + size) > max_packet_size) {
404  if (len > 4) {
405  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
406  s->buf_ptr = s->buf + 4;
407  }
408  }
409  if (s->buf_ptr == s->buf + 4) {
410  s->timestamp = s->cur_timestamp;
411  }
412 
413  /* add the packet */
414  if (size > max_packet_size) {
415  /* big packet: fragment */
416  count = 0;
417  while (size > 0) {
418  len = max_packet_size - 4;
419  if (len > size)
420  len = size;
421  /* build fragmented packet */
422  s->buf[0] = 0;
423  s->buf[1] = 0;
424  s->buf[2] = count >> 8;
425  s->buf[3] = count;
426  memcpy(s->buf + 4, buf1, len);
427  ff_rtp_send_data(s1, s->buf, len + 4, 0);
428  size -= len;
429  buf1 += len;
430  count += len;
431  }
432  } else {
433  if (s->buf_ptr == s->buf + 4) {
434  /* no fragmentation possible */
435  s->buf[0] = 0;
436  s->buf[1] = 0;
437  s->buf[2] = 0;
438  s->buf[3] = 0;
439  }
440  memcpy(s->buf_ptr, buf1, size);
441  s->buf_ptr += size;
442  }
443 }
444 
446  const uint8_t *buf1, int size)
447 {
448  RTPMuxContext *s = s1->priv_data;
449  int len, max_packet_size;
450 
451  max_packet_size = s->max_payload_size;
452 
453  while (size > 0) {
454  len = max_packet_size;
455  if (len > size)
456  len = size;
457 
458  s->timestamp = s->cur_timestamp;
459  ff_rtp_send_data(s1, buf1, len, (len == size));
460 
461  buf1 += len;
462  size -= len;
463  }
464 }
465 
466 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
468  const uint8_t *buf1, int size)
469 {
470  RTPMuxContext *s = s1->priv_data;
471  int len, out_len;
472 
473  s->timestamp = s->cur_timestamp;
474  while (size >= TS_PACKET_SIZE) {
475  len = s->max_payload_size - (s->buf_ptr - s->buf);
476  if (len > size)
477  len = size;
478  memcpy(s->buf_ptr, buf1, len);
479  buf1 += len;
480  size -= len;
481  s->buf_ptr += len;
482 
483  out_len = s->buf_ptr - s->buf;
484  if (out_len >= s->max_payload_size) {
485  ff_rtp_send_data(s1, s->buf, out_len, 0);
486  s->buf_ptr = s->buf;
487  }
488  }
489 }
490 
491 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
492 {
493  RTPMuxContext *s = s1->priv_data;
494  AVStream *st = s1->streams[0];
495  int frame_duration = av_get_audio_frame_duration(st->codec, 0);
496  int frame_size = st->codec->block_align;
497  int frames = size / frame_size;
498 
499  while (frames > 0) {
500  int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
501 
502  if (!s->num_frames) {
503  s->buf_ptr = s->buf;
504  s->timestamp = s->cur_timestamp;
505  }
506  memcpy(s->buf_ptr, buf, n * frame_size);
507  frames -= n;
508  s->num_frames += n;
509  s->buf_ptr += n * frame_size;
510  buf += n * frame_size;
511  s->cur_timestamp += n * frame_duration;
512 
513  if (s->num_frames == s->max_frames_per_packet) {
514  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
515  s->num_frames = 0;
516  }
517  }
518  return 0;
519 }
520 
522 {
523  RTPMuxContext *s = s1->priv_data;
524  AVStream *st = s1->streams[0];
525  int rtcp_bytes;
526  int size= pkt->size;
527 
528  av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
529 
530  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
532  if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
533  (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
534  !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
535  rtcp_send_sr(s1, ff_ntp_time(), 0);
537  s->first_packet = 0;
538  }
539  s->cur_timestamp = s->base_timestamp + pkt->pts;
540 
541  switch(st->codec->codec_id) {
544  case AV_CODEC_ID_PCM_U8:
545  case AV_CODEC_ID_PCM_S8:
546  return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
551  return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
553  /* The actual sample size is half a byte per sample, but since the
554  * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
555  * the correct parameter for send_samples_bits is 8 bits per stream
556  * clock. */
557  return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
559  return rtp_send_samples(s1, pkt->data, size,
561  case AV_CODEC_ID_MP2:
562  case AV_CODEC_ID_MP3:
563  rtp_send_mpegaudio(s1, pkt->data, size);
564  break;
567  ff_rtp_send_mpegvideo(s1, pkt->data, size);
568  break;
569  case AV_CODEC_ID_AAC:
570  if (s->flags & FF_RTP_FLAG_MP4A_LATM)
571  ff_rtp_send_latm(s1, pkt->data, size);
572  else
573  ff_rtp_send_aac(s1, pkt->data, size);
574  break;
575  case AV_CODEC_ID_AMR_NB:
576  case AV_CODEC_ID_AMR_WB:
577  ff_rtp_send_amr(s1, pkt->data, size);
578  break;
579  case AV_CODEC_ID_MPEG2TS:
580  rtp_send_mpegts_raw(s1, pkt->data, size);
581  break;
582  case AV_CODEC_ID_H264:
583  ff_rtp_send_h264(s1, pkt->data, size);
584  break;
585  case AV_CODEC_ID_H261:
586  ff_rtp_send_h261(s1, pkt->data, size);
587  break;
588  case AV_CODEC_ID_H263:
589  if (s->flags & FF_RTP_FLAG_RFC2190) {
590  int mb_info_size = 0;
591  const uint8_t *mb_info =
593  &mb_info_size);
594  if (!mb_info) {
595  av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n");
596  return AVERROR(ENOMEM);
597  }
598  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
599  break;
600  }
601  /* Fallthrough */
602  case AV_CODEC_ID_H263P:
603  ff_rtp_send_h263(s1, pkt->data, size);
604  break;
605  case AV_CODEC_ID_HEVC:
606  ff_rtp_send_hevc(s1, pkt->data, size);
607  break;
608  case AV_CODEC_ID_VORBIS:
609  case AV_CODEC_ID_THEORA:
610  ff_rtp_send_xiph(s1, pkt->data, size);
611  break;
612  case AV_CODEC_ID_VP8:
613  ff_rtp_send_vp8(s1, pkt->data, size);
614  break;
615  case AV_CODEC_ID_ILBC:
616  rtp_send_ilbc(s1, pkt->data, size);
617  break;
618  case AV_CODEC_ID_MJPEG:
619  ff_rtp_send_jpeg(s1, pkt->data, size);
620  break;
621  case AV_CODEC_ID_OPUS:
622  if (size > s->max_payload_size) {
623  av_log(s1, AV_LOG_ERROR,
624  "Packet size %d too large for max RTP payload size %d\n",
625  size, s->max_payload_size);
626  return AVERROR(EINVAL);
627  }
628  /* Intentional fallthrough */
629  default:
630  /* better than nothing : send the codec raw data */
631  rtp_send_raw(s1, pkt->data, size);
632  break;
633  }
634  return 0;
635 }
636 
638 {
639  RTPMuxContext *s = s1->priv_data;
640 
641  /* If the caller closes and recreates ->pb, this might actually
642  * be NULL here even if it was successfully allocated at the start. */
643  if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
644  rtcp_send_sr(s1, ff_ntp_time(), 1);
645  av_freep(&s->buf);
646 
647  return 0;
648 }
649 
651  .name = "rtp",
652  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
653  .priv_data_size = sizeof(RTPMuxContext),
654  .audio_codec = AV_CODEC_ID_PCM_MULAW,
655  .video_codec = AV_CODEC_ID_MPEG4,
659  .priv_class = &rtp_muxer_class,
661 };