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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_DIRAC:
53  case AV_CODEC_ID_H261:
54  case AV_CODEC_ID_H263:
55  case AV_CODEC_ID_H263P:
56  case AV_CODEC_ID_H264:
57  case AV_CODEC_ID_HEVC:
60  case AV_CODEC_ID_MPEG4:
61  case AV_CODEC_ID_AAC:
62  case AV_CODEC_ID_MP2:
63  case AV_CODEC_ID_MP3:
66  case AV_CODEC_ID_PCM_S8:
71  case AV_CODEC_ID_PCM_U8:
73  case AV_CODEC_ID_AMR_NB:
74  case AV_CODEC_ID_AMR_WB:
75  case AV_CODEC_ID_VORBIS:
76  case AV_CODEC_ID_THEORA:
77  case AV_CODEC_ID_VP8:
78  case AV_CODEC_ID_VP9:
82  case AV_CODEC_ID_ILBC:
83  case AV_CODEC_ID_MJPEG:
84  case AV_CODEC_ID_SPEEX:
85  case AV_CODEC_ID_OPUS:
86  return 1;
87  default:
88  return 0;
89  }
90 }
91 
93 {
94  RTPMuxContext *s = s1->priv_data;
95  int n, ret = AVERROR(EINVAL);
96  AVStream *st;
97 
98  if (s1->nb_streams != 1) {
99  av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
100  return AVERROR(EINVAL);
101  }
102  st = s1->streams[0];
103  if (!is_supported(st->codecpar->codec_id)) {
104  av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
105 
106  return -1;
107  }
108 
109  if (s->payload_type < 0) {
110  /* Re-validate non-dynamic payload types */
111  if (st->id < RTP_PT_PRIVATE)
112  st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
113 
114  s->payload_type = st->id;
115  } else {
116  /* private option takes priority */
117  st->id = s->payload_type;
118  }
119 
121  s->timestamp = s->base_timestamp;
122  s->cur_timestamp = 0;
123  if (!s->ssrc)
124  s->ssrc = av_get_random_seed();
125  s->first_packet = 1;
128  /* Round the NTP time to whole milliseconds. */
129  s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
131  // Pick a random sequence start number, but in the lower end of the
132  // available range, so that any wraparound doesn't happen immediately.
133  // (Immediate wraparound would be an issue for SRTP.)
134  if (s->seq < 0) {
135  if (s1->flags & AVFMT_FLAG_BITEXACT) {
136  s->seq = 0;
137  } else
138  s->seq = av_get_random_seed() & 0x0fff;
139  } else
140  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
141 
142  if (s1->packet_size) {
143  if (s1->pb->max_packet_size)
144  s1->packet_size = FFMIN(s1->packet_size,
145  s1->pb->max_packet_size);
146  } else
147  s1->packet_size = s1->pb->max_packet_size;
148  if (s1->packet_size <= 12) {
149  av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
150  return AVERROR(EIO);
151  }
152  s->buf = av_malloc(s1->packet_size);
153  if (!s->buf) {
154  return AVERROR(ENOMEM);
155  }
156  s->max_payload_size = s1->packet_size - 12;
157 
158  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
159  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
160  } else {
161  avpriv_set_pts_info(st, 32, 1, 90000);
162  }
163  s->buf_ptr = s->buf;
164  switch(st->codecpar->codec_id) {
165  case AV_CODEC_ID_MP2:
166  case AV_CODEC_ID_MP3:
167  s->buf_ptr = s->buf + 4;
168  avpriv_set_pts_info(st, 32, 1, 90000);
169  break;
172  break;
173  case AV_CODEC_ID_MPEG2TS:
175  if (n < 1)
176  n = 1;
178  break;
179  case AV_CODEC_ID_DIRAC:
181  av_log(s, AV_LOG_ERROR,
182  "Packetizing VC-2 is experimental and does not use all values "
183  "of the specification "
184  "(even though most receivers may handle it just fine). "
185  "Please set -strict experimental in order to enable it.\n");
186  ret = AVERROR_EXPERIMENTAL;
187  goto fail;
188  }
189  break;
190  case AV_CODEC_ID_H261:
192  av_log(s, AV_LOG_ERROR,
193  "Packetizing H.261 is experimental and produces incorrect "
194  "packetization for cases where GOBs don't fit into packets "
195  "(even though most receivers may handle it just fine). "
196  "Please set -f_strict experimental in order to enable it.\n");
197  ret = AVERROR_EXPERIMENTAL;
198  goto fail;
199  }
200  break;
201  case AV_CODEC_ID_H264:
202  /* check for H.264 MP4 syntax */
203  if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
204  s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
205  }
206  break;
207  case AV_CODEC_ID_HEVC:
208  /* Only check for the standardized hvcC version of extradata, keeping
209  * things simple and similar to the avcC/H.264 case above, instead
210  * of trying to handle the pre-standardization versions (as in
211  * libavcodec/hevc.c). */
212  if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
213  s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
214  }
215  break;
216  case AV_CODEC_ID_VP9:
218  av_log(s, AV_LOG_ERROR,
219  "Packetizing VP9 is experimental and its specification is "
220  "still in draft state. "
221  "Please set -strict experimental in order to enable it.\n");
222  ret = AVERROR_EXPERIMENTAL;
223  goto fail;
224  }
225  break;
226  case AV_CODEC_ID_VORBIS:
227  case AV_CODEC_ID_THEORA:
228  s->max_frames_per_packet = 15;
229  break;
231  /* Due to a historical error, the clock rate for G722 in RTP is
232  * 8000, even if the sample rate is 16000. See RFC 3551. */
233  avpriv_set_pts_info(st, 32, 1, 8000);
234  break;
235  case AV_CODEC_ID_OPUS:
236  if (st->codecpar->channels > 2) {
237  av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
238  goto fail;
239  }
240  /* The opus RTP RFC says that all opus streams should use 48000 Hz
241  * as clock rate, since all opus sample rates can be expressed in
242  * this clock rate, and sample rate changes on the fly are supported. */
243  avpriv_set_pts_info(st, 32, 1, 48000);
244  break;
245  case AV_CODEC_ID_ILBC:
246  if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
247  av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
248  goto fail;
249  }
251  break;
252  case AV_CODEC_ID_AMR_NB:
253  case AV_CODEC_ID_AMR_WB:
254  s->max_frames_per_packet = 50;
256  n = 31;
257  else
258  n = 61;
259  /* max_header_toc_size + the largest AMR payload must fit */
260  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
261  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
262  goto fail;
263  }
264  if (st->codecpar->channels != 1) {
265  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
266  goto fail;
267  }
268  break;
269  case AV_CODEC_ID_AAC:
270  s->max_frames_per_packet = 50;
271  break;
272  default:
273  break;
274  }
275 
276  return 0;
277 
278 fail:
279  av_freep(&s->buf);
280  return ret;
281 }
282 
283 /* send an rtcp sender report packet */
284 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
285 {
286  RTPMuxContext *s = s1->priv_data;
287  uint32_t rtp_ts;
288 
289  av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
290 
291  s->last_rtcp_ntp_time = ntp_time;
292  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
293  s1->streams[0]->time_base) + s->base_timestamp;
294  avio_w8(s1->pb, RTP_VERSION << 6);
295  avio_w8(s1->pb, RTCP_SR);
296  avio_wb16(s1->pb, 6); /* length in words - 1 */
297  avio_wb32(s1->pb, s->ssrc);
298  avio_wb32(s1->pb, ntp_time / 1000000);
299  avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
300  avio_wb32(s1->pb, rtp_ts);
301  avio_wb32(s1->pb, s->packet_count);
302  avio_wb32(s1->pb, s->octet_count);
303 
304  if (s->cname) {
305  int len = FFMIN(strlen(s->cname), 255);
306  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
307  avio_w8(s1->pb, RTCP_SDES);
308  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
309 
310  avio_wb32(s1->pb, s->ssrc);
311  avio_w8(s1->pb, 0x01); /* CNAME */
312  avio_w8(s1->pb, len);
313  avio_write(s1->pb, s->cname, len);
314  avio_w8(s1->pb, 0); /* END */
315  for (len = (7 + len) % 4; len % 4; len++)
316  avio_w8(s1->pb, 0);
317  }
318 
319  if (bye) {
320  avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
321  avio_w8(s1->pb, RTCP_BYE);
322  avio_wb16(s1->pb, 1); /* length in words - 1 */
323  avio_wb32(s1->pb, s->ssrc);
324  }
325 
326  avio_flush(s1->pb);
327 }
328 
329 /* send an rtp packet. sequence number is incremented, but the caller
330  must update the timestamp itself */
331 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
332 {
333  RTPMuxContext *s = s1->priv_data;
334 
335  av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
336 
337  /* build the RTP header */
338  avio_w8(s1->pb, RTP_VERSION << 6);
339  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
340  avio_wb16(s1->pb, s->seq);
341  avio_wb32(s1->pb, s->timestamp);
342  avio_wb32(s1->pb, s->ssrc);
343 
344  avio_write(s1->pb, buf1, len);
345  avio_flush(s1->pb);
346 
347  s->seq = (s->seq + 1) & 0xffff;
348  s->octet_count += len;
349  s->packet_count++;
350 }
351 
352 /* send an integer number of samples and compute time stamp and fill
353  the rtp send buffer before sending. */
355  const uint8_t *buf1, int size, int sample_size_bits)
356 {
357  RTPMuxContext *s = s1->priv_data;
358  int len, max_packet_size, n;
359  /* Calculate the number of bytes to get samples aligned on a byte border */
360  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
361 
362  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
363  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
364  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
365  return AVERROR(EINVAL);
366  n = 0;
367  while (size > 0) {
368  s->buf_ptr = s->buf;
369  len = FFMIN(max_packet_size, size);
370 
371  /* copy data */
372  memcpy(s->buf_ptr, buf1, len);
373  s->buf_ptr += len;
374  buf1 += len;
375  size -= len;
376  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
377  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
378  n += (s->buf_ptr - s->buf);
379  }
380  return 0;
381 }
382 
384  const uint8_t *buf1, int size)
385 {
386  RTPMuxContext *s = s1->priv_data;
387  int len, count, max_packet_size;
388 
389  max_packet_size = s->max_payload_size;
390 
391  /* test if we must flush because not enough space */
392  len = (s->buf_ptr - s->buf);
393  if ((len + size) > max_packet_size) {
394  if (len > 4) {
395  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
396  s->buf_ptr = s->buf + 4;
397  }
398  }
399  if (s->buf_ptr == s->buf + 4) {
400  s->timestamp = s->cur_timestamp;
401  }
402 
403  /* add the packet */
404  if (size > max_packet_size) {
405  /* big packet: fragment */
406  count = 0;
407  while (size > 0) {
408  len = max_packet_size - 4;
409  if (len > size)
410  len = size;
411  /* build fragmented packet */
412  s->buf[0] = 0;
413  s->buf[1] = 0;
414  s->buf[2] = count >> 8;
415  s->buf[3] = count;
416  memcpy(s->buf + 4, buf1, len);
417  ff_rtp_send_data(s1, s->buf, len + 4, 0);
418  size -= len;
419  buf1 += len;
420  count += len;
421  }
422  } else {
423  if (s->buf_ptr == s->buf + 4) {
424  /* no fragmentation possible */
425  s->buf[0] = 0;
426  s->buf[1] = 0;
427  s->buf[2] = 0;
428  s->buf[3] = 0;
429  }
430  memcpy(s->buf_ptr, buf1, size);
431  s->buf_ptr += size;
432  }
433 }
434 
436  const uint8_t *buf1, int size)
437 {
438  RTPMuxContext *s = s1->priv_data;
439  int len, max_packet_size;
440 
441  max_packet_size = s->max_payload_size;
442 
443  while (size > 0) {
444  len = max_packet_size;
445  if (len > size)
446  len = size;
447 
448  s->timestamp = s->cur_timestamp;
449  ff_rtp_send_data(s1, buf1, len, (len == size));
450 
451  buf1 += len;
452  size -= len;
453  }
454 }
455 
456 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
458  const uint8_t *buf1, int size)
459 {
460  RTPMuxContext *s = s1->priv_data;
461  int len, out_len;
462 
463  s->timestamp = s->cur_timestamp;
464  while (size >= TS_PACKET_SIZE) {
465  len = s->max_payload_size - (s->buf_ptr - s->buf);
466  if (len > size)
467  len = size;
468  memcpy(s->buf_ptr, buf1, len);
469  buf1 += len;
470  size -= len;
471  s->buf_ptr += len;
472 
473  out_len = s->buf_ptr - s->buf;
474  if (out_len >= s->max_payload_size) {
475  ff_rtp_send_data(s1, s->buf, out_len, 0);
476  s->buf_ptr = s->buf;
477  }
478  }
479 }
480 
481 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
482 {
483  RTPMuxContext *s = s1->priv_data;
484  AVStream *st = s1->streams[0];
485  int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
486  int frame_size = st->codecpar->block_align;
487  int frames = size / frame_size;
488 
489  while (frames > 0) {
490  if (s->num_frames > 0 &&
492  s1->max_delay, AV_TIME_BASE_Q) >= 0) {
493  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
494  s->num_frames = 0;
495  }
496 
497  if (!s->num_frames) {
498  s->buf_ptr = s->buf;
499  s->timestamp = s->cur_timestamp;
500  }
501  memcpy(s->buf_ptr, buf, frame_size);
502  frames--;
503  s->num_frames++;
504  s->buf_ptr += frame_size;
505  buf += frame_size;
506  s->cur_timestamp += frame_duration;
507 
508  if (s->num_frames == s->max_frames_per_packet) {
509  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
510  s->num_frames = 0;
511  }
512  }
513  return 0;
514 }
515 
517 {
518  RTPMuxContext *s = s1->priv_data;
519  AVStream *st = s1->streams[0];
520  int rtcp_bytes;
521  int size= pkt->size;
522 
523  av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
524 
525  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
527  if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
528  (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
529  !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
530  rtcp_send_sr(s1, ff_ntp_time(), 0);
532  s->first_packet = 0;
533  }
534  s->cur_timestamp = s->base_timestamp + pkt->pts;
535 
536  switch(st->codecpar->codec_id) {
539  case AV_CODEC_ID_PCM_U8:
540  case AV_CODEC_ID_PCM_S8:
541  return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
546  return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
548  /* The actual sample size is half a byte per sample, but since the
549  * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
550  * the correct parameter for send_samples_bits is 8 bits per stream
551  * clock. */
552  return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
555  return rtp_send_samples(s1, pkt->data, size,
557  case AV_CODEC_ID_MP2:
558  case AV_CODEC_ID_MP3:
559  rtp_send_mpegaudio(s1, pkt->data, size);
560  break;
563  ff_rtp_send_mpegvideo(s1, pkt->data, size);
564  break;
565  case AV_CODEC_ID_AAC:
566  if (s->flags & FF_RTP_FLAG_MP4A_LATM)
567  ff_rtp_send_latm(s1, pkt->data, size);
568  else
569  ff_rtp_send_aac(s1, pkt->data, size);
570  break;
571  case AV_CODEC_ID_AMR_NB:
572  case AV_CODEC_ID_AMR_WB:
573  ff_rtp_send_amr(s1, pkt->data, size);
574  break;
575  case AV_CODEC_ID_MPEG2TS:
576  rtp_send_mpegts_raw(s1, pkt->data, size);
577  break;
578  case AV_CODEC_ID_DIRAC:
579  ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
580  break;
581  case AV_CODEC_ID_H264:
582  ff_rtp_send_h264_hevc(s1, pkt->data, size);
583  break;
584  case AV_CODEC_ID_H261:
585  ff_rtp_send_h261(s1, pkt->data, size);
586  break;
587  case AV_CODEC_ID_H263:
588  if (s->flags & FF_RTP_FLAG_RFC2190) {
589  int mb_info_size = 0;
590  const uint8_t *mb_info =
592  &mb_info_size);
593  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
594  break;
595  }
596  /* Fallthrough */
597  case AV_CODEC_ID_H263P:
598  ff_rtp_send_h263(s1, pkt->data, size);
599  break;
600  case AV_CODEC_ID_HEVC:
601  ff_rtp_send_h264_hevc(s1, pkt->data, size);
602  break;
603  case AV_CODEC_ID_VORBIS:
604  case AV_CODEC_ID_THEORA:
605  ff_rtp_send_xiph(s1, pkt->data, size);
606  break;
607  case AV_CODEC_ID_VP8:
608  ff_rtp_send_vp8(s1, pkt->data, size);
609  break;
610  case AV_CODEC_ID_VP9:
611  ff_rtp_send_vp9(s1, pkt->data, size);
612  break;
613  case AV_CODEC_ID_ILBC:
614  rtp_send_ilbc(s1, pkt->data, size);
615  break;
616  case AV_CODEC_ID_MJPEG:
617  ff_rtp_send_jpeg(s1, pkt->data, size);
618  break;
619  case AV_CODEC_ID_OPUS:
620  if (size > s->max_payload_size) {
621  av_log(s1, AV_LOG_ERROR,
622  "Packet size %d too large for max RTP payload size %d\n",
623  size, s->max_payload_size);
624  return AVERROR(EINVAL);
625  }
626  /* Intentional fallthrough */
627  default:
628  /* better than nothing : send the codec raw data */
629  rtp_send_raw(s1, pkt->data, size);
630  break;
631  }
632  return 0;
633 }
634 
636 {
637  RTPMuxContext *s = s1->priv_data;
638 
639  /* If the caller closes and recreates ->pb, this might actually
640  * be NULL here even if it was successfully allocated at the start. */
641  if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
642  rtcp_send_sr(s1, ff_ntp_time(), 1);
643  av_freep(&s->buf);
644 
645  return 0;
646 }
647 
649  .name = "rtp",
650  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
651  .priv_data_size = sizeof(RTPMuxContext),
652  .audio_codec = AV_CODEC_ID_PCM_MULAW,
653  .video_codec = AV_CODEC_ID_MPEG4,
657  .priv_class = &rtp_muxer_class,
659 };
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2986
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
Definition: ffmpeg.c:671
unsigned int packet_size
Definition: avformat.h:1453
#define NULL
Definition: coverity.c:32
enum AVFieldOrder field_order
Video only.
Definition: avcodec.h:4233
const char * s
Definition: avisynth_c.h:768
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
Definition: avformat.h:1580
AVOption.
Definition: opt.h:246
#define LIBAVUTIL_VERSION_INT
Definition: version.h:86
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4741
int payload_type
Definition: rtpenc.h:31
static int rtp_send_samples(AVFormatContext *s1, const uint8_t *buf1, int size, int sample_size_bits)
Definition: rtpenc.c:354
#define NTP_OFFSET_US
Definition: internal.h:241
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: rtpenc.c:516
#define RTP_VERSION
Definition: rtp.h:78
int64_t last_rtcp_ntp_time
Definition: rtpenc.h:42
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:4152
int size
Definition: avcodec.h:1680
#define RTCP_TX_RATIO_NUM
Definition: rtp.h:82
unsigned int last_octet_count
Definition: rtpenc.h:46
static const AVOption options[]
Definition: rtpenc.c:31
void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size)
Packetize AMR frames into RTP packets according to RFC 3267, in octet-aligned mode.
Definition: rtpenc_amr.c:30
int max_payload_size
Definition: rtpenc.h:38
static AVPacket pkt
#define RTCP_TX_RATIO_DEN
Definition: rtp.h:83
An AV_PKT_DATA_H263_MB_INFO side data packet contains a number of structures with info about macroblo...
Definition: avcodec.h:1458
int strict_std_compliance
Allow non-standard and experimental extension.
Definition: avformat.h:1635
void ff_rtp_send_vc2hq(AVFormatContext *s1, const uint8_t *buf, int size, int interlaced)
Definition: rtpenc_vc2hq.c:102
#define AVFMT_TS_NONSTRICT
Format does not require strictly increasing timestamps, but they must still be monotonic.
Definition: avformat.h:496
int nal_length_size
Number of bytes used for H.264 NAL length, if the MP4 syntax is used (1, 2 or 4)
Definition: rtpenc.h:58
#define FF_RTP_FLAG_MP4A_LATM
Definition: rtpenc.h:68
Format I/O context.
Definition: avformat.h:1349
#define RTP_PT_PRIVATE
Definition: rtp.h:77
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
uint8_t
#define av_malloc(s)
AVOptions.
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
static const AVClass rtp_muxer_class
Definition: rtpenc.c:40
int id
Format-specific stream ID.
Definition: avformat.h:896
int max_frames_per_packet
Definition: rtpenc.h:52
void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size)
Packetize H.263 frames into RTP packets according to RFC 4629.
Definition: rtpenc_h263.c:43
void ff_rtp_send_vp9(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_vp9.c:26
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1417
unsigned int octet_count
Definition: rtpenc.h:45
#define TS_PACKET_SIZE
Definition: mpegts.h:29
static int rtp_write_header(AVFormatContext *s1)
Definition: rtpenc.c:92
int flags
Flags modifying the (de)muxer behaviour.
Definition: avformat.h:1460
uint8_t * data
Definition: avcodec.h:1679
static int flags
Definition: log.c:57
Definition: rtp.h:99
uint8_t * buf
Definition: rtpenc.h:49
ptrdiff_t size
Definition: opengl_enc.c:101
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
Definition: aviobuf.c:216
#define AVFMT_FLAG_BITEXACT
When muxing, try to avoid writing any random/volatile data to the output.
Definition: avformat.h:1477
#define av_log(a,...)
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
Definition: opt.h:276
#define FF_RTP_FLAG_RFC2190
Definition: rtpenc.h:69
uint64_t ff_ntp_time(void)
Get the current time since NTP epoch in microseconds.
Definition: utils.c:4556
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 max_packet_size
Definition: avio.h:242
uint32_t ssrc
Definition: rtpenc.h:32
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:214
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_aac.c:27
av_default_item_name
void ff_rtp_send_jpeg(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_jpeg.c:28
#define AVERROR(e)
Definition: error.h:43
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:350
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:179
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:557
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:4148
void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_latm.c:25
void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_vp8.c:26
int64_t av_gcd(int64_t a, int64_t b)
Compute the greatest common divisor of two integer operands.
Definition: mathematics.c:37
GLsizei count
Definition: opengl_enc.c:109
int av_get_audio_frame_duration2(AVCodecParameters *par, int frame_bytes)
This function is the same as av_get_audio_frame_duration(), except it works with AVCodecParameters in...
Definition: utils.c:1876
#define FF_RTP_FLAG_SKIP_RTCP
Definition: rtpenc.h:70
#define fail()
Definition: checkasm.h:109
int av_compare_ts(int64_t ts_a, AVRational tb_a, int64_t ts_b, AVRational tb_b)
Compare two timestamps each in its own time base.
Definition: mathematics.c:147
static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:457
int extradata_size
Size of the extradata content in bytes.
Definition: avcodec.h:4170
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1405
int block_align
Audio only.
Definition: avcodec.h:4269
void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
Packetize Xiph frames into RTP according to RFC 5215 (Vorbis) and the Theora RFC draft.
Definition: rtpenc_xiph.c:33
int void avio_flush(AVIOContext *s)
Force flushing of buffered data.
Definition: aviobuf.c:236
#define FFMIN(a, b)
Definition: common.h:96
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
Definition: rtpenc.c:331
void ff_rtp_send_h261(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc_h261.c:39
static int write_trailer(AVFormatContext *s1)
Definition: v4l2enc.c:94
void ff_rtp_send_h264_hevc(AVFormatContext *s1, const uint8_t *buf1, int size)
const char * name
Definition: avformat.h:524
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
Definition: rtpenc.c:481
int n
Definition: avisynth_c.h:684
void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc_mpv.c:29
int frames
Definition: movenc.c:65
static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:383
preferred ID for MPEG-1/2 video decoding
Definition: avcodec.h:219
#define AVERROR_EXPERIMENTAL
Requested feature is flagged experimental. Set strict_std_compliance if you really want to use it...
Definition: error.h:72
const char * avcodec_get_name(enum AVCodecID id)
Get the name of a codec.
Definition: utils.c:1294
Stream structure.
Definition: avformat.h:889
int64_t first_rtcp_ntp_time
Definition: rtpenc.h:43
uint32_t cur_timestamp
Definition: rtpenc.h:37
AVOutputFormat ff_rtp_muxer
Definition: rtpenc.c:648
int frame_size
Definition: mxfenc.c:1896
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:260
Definition: rtp.h:100
AVIOContext * pb
I/O context.
Definition: avformat.h:1391
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
void avio_w8(AVIOContext *s, int b)
Definition: aviobuf.c:194
static int is_supported(enum AVCodecID id)
Definition: rtpenc.c:49
int first_packet
Definition: rtpenc.h:47
void * buf
Definition: avisynth_c.h:690
Describe the class of an AVClass context structure.
Definition: log.h:67
Rational number (pair of numerator and denominator).
Definition: rational.h:58
int flags
Definition: rtpenc.h:61
#define s1
Definition: regdef.h:38
int num_frames
Definition: rtpenc.h:39
uint32_t base_timestamp
Definition: rtpenc.h:36
uint8_t * buf_ptr
Definition: rtpenc.h:50
void avio_wb16(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:480
int sample_rate
Audio only.
Definition: avcodec.h:4262
Main libavformat public API header.
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:435
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
Definition: rtpenc.c:284
#define RTCP_SR_SIZE
Definition: rtpenc.c:47
Definition: rtp.h:97
unsigned int packet_count
Definition: rtpenc.h:44
FAKE codec to indicate a raw MPEG-2 TS stream (only used by libavformat)
Definition: avcodec.h:690
uint32_t timestamp
Definition: rtpenc.h:35
int len
int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecParameters *par, int idx)
Return the payload type for a given stream used in the given format context.
Definition: rtp.c:90
void * priv_data
Format private data.
Definition: avformat.h:1377
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:337
static int rtp_write_trailer(AVFormatContext *s1)
Definition: rtpenc.c:635
int bits_per_coded_sample
The number of bits per sample in the codedwords.
Definition: avcodec.h:4194
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
Definition: avcodec.h:4166
int channels
Audio only.
Definition: avcodec.h:4258
void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size, const uint8_t *mb_info, int mb_info_size)
void avio_wb32(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:382
#define av_freep(p)
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:120
AVCodecParameters * codecpar
Definition: avformat.h:1252
int stream_index
Definition: avcodec.h:1681
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:926
This structure stores compressed data.
Definition: avcodec.h:1656
#define FF_RTP_FLAG_SEND_BYE
Definition: rtpenc.h:72
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1672
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248