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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:
72  case AV_CODEC_ID_PCM_U8:
74  case AV_CODEC_ID_AMR_NB:
75  case AV_CODEC_ID_AMR_WB:
76  case AV_CODEC_ID_VORBIS:
77  case AV_CODEC_ID_THEORA:
78  case AV_CODEC_ID_VP8:
79  case AV_CODEC_ID_VP9:
83  case AV_CODEC_ID_ILBC:
84  case AV_CODEC_ID_MJPEG:
85  case AV_CODEC_ID_SPEEX:
86  case AV_CODEC_ID_OPUS:
87  return 1;
88  default:
89  return 0;
90  }
91 }
92 
94 {
95  RTPMuxContext *s = s1->priv_data;
96  int n, ret = AVERROR(EINVAL);
97  AVStream *st;
98 
99  if (s1->nb_streams != 1) {
100  av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
101  return AVERROR(EINVAL);
102  }
103  st = s1->streams[0];
104  if (!is_supported(st->codecpar->codec_id)) {
105  av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
106 
107  return -1;
108  }
109 
110  if (s->payload_type < 0) {
111  /* Re-validate non-dynamic payload types */
112  if (st->id < RTP_PT_PRIVATE)
113  st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
114 
115  s->payload_type = st->id;
116  } else {
117  /* private option takes priority */
118  st->id = s->payload_type;
119  }
120 
122  s->timestamp = s->base_timestamp;
123  s->cur_timestamp = 0;
124  if (!s->ssrc)
125  s->ssrc = av_get_random_seed();
126  s->first_packet = 1;
129  /* Round the NTP time to whole milliseconds. */
130  s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
132  // Pick a random sequence start number, but in the lower end of the
133  // available range, so that any wraparound doesn't happen immediately.
134  // (Immediate wraparound would be an issue for SRTP.)
135  if (s->seq < 0) {
136  if (s1->flags & AVFMT_FLAG_BITEXACT) {
137  s->seq = 0;
138  } else
139  s->seq = av_get_random_seed() & 0x0fff;
140  } else
141  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
142 
143  if (s1->packet_size) {
144  if (s1->pb->max_packet_size)
145  s1->packet_size = FFMIN(s1->packet_size,
146  s1->pb->max_packet_size);
147  } else
148  s1->packet_size = s1->pb->max_packet_size;
149  if (s1->packet_size <= 12) {
150  av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
151  return AVERROR(EIO);
152  }
153  s->buf = av_malloc(s1->packet_size);
154  if (!s->buf) {
155  return AVERROR(ENOMEM);
156  }
157  s->max_payload_size = s1->packet_size - 12;
158 
159  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
160  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
161  } else {
162  avpriv_set_pts_info(st, 32, 1, 90000);
163  }
164  s->buf_ptr = s->buf;
165  switch(st->codecpar->codec_id) {
166  case AV_CODEC_ID_MP2:
167  case AV_CODEC_ID_MP3:
168  s->buf_ptr = s->buf + 4;
169  avpriv_set_pts_info(st, 32, 1, 90000);
170  break;
173  break;
174  case AV_CODEC_ID_MPEG2TS:
176  if (n < 1)
177  n = 1;
179  break;
180  case AV_CODEC_ID_DIRAC:
182  av_log(s, AV_LOG_ERROR,
183  "Packetizing VC-2 is experimental and does not use all values "
184  "of the specification "
185  "(even though most receivers may handle it just fine). "
186  "Please set -strict experimental in order to enable it.\n");
187  ret = AVERROR_EXPERIMENTAL;
188  goto fail;
189  }
190  break;
191  case AV_CODEC_ID_H261:
193  av_log(s, AV_LOG_ERROR,
194  "Packetizing H.261 is experimental and produces incorrect "
195  "packetization for cases where GOBs don't fit into packets "
196  "(even though most receivers may handle it just fine). "
197  "Please set -f_strict experimental in order to enable it.\n");
198  ret = AVERROR_EXPERIMENTAL;
199  goto fail;
200  }
201  break;
202  case AV_CODEC_ID_H264:
203  /* check for H.264 MP4 syntax */
204  if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
205  s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
206  }
207  break;
208  case AV_CODEC_ID_HEVC:
209  /* Only check for the standardized hvcC version of extradata, keeping
210  * things simple and similar to the avcC/H.264 case above, instead
211  * of trying to handle the pre-standardization versions (as in
212  * libavcodec/hevc.c). */
213  if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
214  s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
215  }
216  break;
217  case AV_CODEC_ID_VP9:
219  av_log(s, AV_LOG_ERROR,
220  "Packetizing VP9 is experimental and its specification is "
221  "still in draft state. "
222  "Please set -strict experimental in order to enable it.\n");
223  ret = AVERROR_EXPERIMENTAL;
224  goto fail;
225  }
226  break;
227  case AV_CODEC_ID_VORBIS:
228  case AV_CODEC_ID_THEORA:
229  s->max_frames_per_packet = 15;
230  break;
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->codecpar->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->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
248  av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
249  goto fail;
250  }
252  break;
253  case AV_CODEC_ID_AMR_NB:
254  case AV_CODEC_ID_AMR_WB:
255  s->max_frames_per_packet = 50;
257  n = 31;
258  else
259  n = 61;
260  /* max_header_toc_size + the largest AMR payload must fit */
261  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
262  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
263  goto fail;
264  }
265  if (st->codecpar->channels != 1) {
266  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
267  goto fail;
268  }
269  break;
270  case AV_CODEC_ID_AAC:
271  s->max_frames_per_packet = 50;
272  break;
273  default:
274  break;
275  }
276 
277  return 0;
278 
279 fail:
280  av_freep(&s->buf);
281  return ret;
282 }
283 
284 /* send an rtcp sender report packet */
285 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
286 {
287  RTPMuxContext *s = s1->priv_data;
288  uint32_t rtp_ts;
289 
290  av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
291 
292  s->last_rtcp_ntp_time = ntp_time;
293  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
294  s1->streams[0]->time_base) + s->base_timestamp;
295  avio_w8(s1->pb, RTP_VERSION << 6);
296  avio_w8(s1->pb, RTCP_SR);
297  avio_wb16(s1->pb, 6); /* length in words - 1 */
298  avio_wb32(s1->pb, s->ssrc);
299  avio_wb32(s1->pb, ntp_time / 1000000);
300  avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
301  avio_wb32(s1->pb, rtp_ts);
302  avio_wb32(s1->pb, s->packet_count);
303  avio_wb32(s1->pb, s->octet_count);
304 
305  if (s->cname) {
306  int len = FFMIN(strlen(s->cname), 255);
307  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
308  avio_w8(s1->pb, RTCP_SDES);
309  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
310 
311  avio_wb32(s1->pb, s->ssrc);
312  avio_w8(s1->pb, 0x01); /* CNAME */
313  avio_w8(s1->pb, len);
314  avio_write(s1->pb, s->cname, len);
315  avio_w8(s1->pb, 0); /* END */
316  for (len = (7 + len) % 4; len % 4; len++)
317  avio_w8(s1->pb, 0);
318  }
319 
320  if (bye) {
321  avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
322  avio_w8(s1->pb, RTCP_BYE);
323  avio_wb16(s1->pb, 1); /* length in words - 1 */
324  avio_wb32(s1->pb, s->ssrc);
325  }
326 
327  avio_flush(s1->pb);
328 }
329 
330 /* send an rtp packet. sequence number is incremented, but the caller
331  must update the timestamp itself */
332 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
333 {
334  RTPMuxContext *s = s1->priv_data;
335 
336  av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
337 
338  /* build the RTP header */
339  avio_w8(s1->pb, RTP_VERSION << 6);
340  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
341  avio_wb16(s1->pb, s->seq);
342  avio_wb32(s1->pb, s->timestamp);
343  avio_wb32(s1->pb, s->ssrc);
344 
345  avio_write(s1->pb, buf1, len);
346  avio_flush(s1->pb);
347 
348  s->seq = (s->seq + 1) & 0xffff;
349  s->octet_count += len;
350  s->packet_count++;
351 }
352 
353 /* send an integer number of samples and compute time stamp and fill
354  the rtp send buffer before sending. */
356  const uint8_t *buf1, int size, int sample_size_bits)
357 {
358  RTPMuxContext *s = s1->priv_data;
359  int len, max_packet_size, n;
360  /* Calculate the number of bytes to get samples aligned on a byte border */
361  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
362 
363  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
364  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
365  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
366  return AVERROR(EINVAL);
367  n = 0;
368  while (size > 0) {
369  s->buf_ptr = s->buf;
370  len = FFMIN(max_packet_size, size);
371 
372  /* copy data */
373  memcpy(s->buf_ptr, buf1, len);
374  s->buf_ptr += len;
375  buf1 += len;
376  size -= len;
377  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
378  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
379  n += (s->buf_ptr - s->buf);
380  }
381  return 0;
382 }
383 
385  const uint8_t *buf1, int size)
386 {
387  RTPMuxContext *s = s1->priv_data;
388  int len, count, max_packet_size;
389 
390  max_packet_size = s->max_payload_size;
391 
392  /* test if we must flush because not enough space */
393  len = (s->buf_ptr - s->buf);
394  if ((len + size) > max_packet_size) {
395  if (len > 4) {
396  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
397  s->buf_ptr = s->buf + 4;
398  }
399  }
400  if (s->buf_ptr == s->buf + 4) {
401  s->timestamp = s->cur_timestamp;
402  }
403 
404  /* add the packet */
405  if (size > max_packet_size) {
406  /* big packet: fragment */
407  count = 0;
408  while (size > 0) {
409  len = max_packet_size - 4;
410  if (len > size)
411  len = size;
412  /* build fragmented packet */
413  s->buf[0] = 0;
414  s->buf[1] = 0;
415  s->buf[2] = count >> 8;
416  s->buf[3] = count;
417  memcpy(s->buf + 4, buf1, len);
418  ff_rtp_send_data(s1, s->buf, len + 4, 0);
419  size -= len;
420  buf1 += len;
421  count += len;
422  }
423  } else {
424  if (s->buf_ptr == s->buf + 4) {
425  /* no fragmentation possible */
426  s->buf[0] = 0;
427  s->buf[1] = 0;
428  s->buf[2] = 0;
429  s->buf[3] = 0;
430  }
431  memcpy(s->buf_ptr, buf1, size);
432  s->buf_ptr += size;
433  }
434 }
435 
437  const uint8_t *buf1, int size)
438 {
439  RTPMuxContext *s = s1->priv_data;
440  int len, max_packet_size;
441 
442  max_packet_size = s->max_payload_size;
443 
444  while (size > 0) {
445  len = max_packet_size;
446  if (len > size)
447  len = size;
448 
449  s->timestamp = s->cur_timestamp;
450  ff_rtp_send_data(s1, buf1, len, (len == size));
451 
452  buf1 += len;
453  size -= len;
454  }
455 }
456 
457 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
459  const uint8_t *buf1, int size)
460 {
461  RTPMuxContext *s = s1->priv_data;
462  int len, out_len;
463 
464  s->timestamp = s->cur_timestamp;
465  while (size >= TS_PACKET_SIZE) {
466  len = s->max_payload_size - (s->buf_ptr - s->buf);
467  if (len > size)
468  len = size;
469  memcpy(s->buf_ptr, buf1, len);
470  buf1 += len;
471  size -= len;
472  s->buf_ptr += len;
473 
474  out_len = s->buf_ptr - s->buf;
475  if (out_len >= s->max_payload_size) {
476  ff_rtp_send_data(s1, s->buf, out_len, 0);
477  s->buf_ptr = s->buf;
478  }
479  }
480 }
481 
482 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
483 {
484  RTPMuxContext *s = s1->priv_data;
485  AVStream *st = s1->streams[0];
486  int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
487  int frame_size = st->codecpar->block_align;
488  int frames = size / frame_size;
489 
490  while (frames > 0) {
491  if (s->num_frames > 0 &&
493  s1->max_delay, AV_TIME_BASE_Q) >= 0) {
494  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
495  s->num_frames = 0;
496  }
497 
498  if (!s->num_frames) {
499  s->buf_ptr = s->buf;
500  s->timestamp = s->cur_timestamp;
501  }
502  memcpy(s->buf_ptr, buf, frame_size);
503  frames--;
504  s->num_frames++;
505  s->buf_ptr += frame_size;
506  buf += frame_size;
507  s->cur_timestamp += frame_duration;
508 
509  if (s->num_frames == s->max_frames_per_packet) {
510  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
511  s->num_frames = 0;
512  }
513  }
514  return 0;
515 }
516 
518 {
519  RTPMuxContext *s = s1->priv_data;
520  AVStream *st = s1->streams[0];
521  int rtcp_bytes;
522  int size= pkt->size;
523 
524  av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
525 
526  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
528  if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
529  (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
530  !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
531  rtcp_send_sr(s1, ff_ntp_time(), 0);
533  s->first_packet = 0;
534  }
535  s->cur_timestamp = s->base_timestamp + pkt->pts;
536 
537  switch(st->codecpar->codec_id) {
540  case AV_CODEC_ID_PCM_U8:
541  case AV_CODEC_ID_PCM_S8:
542  return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
547  return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
549  return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->channels);
551  /* The actual sample size is half a byte per sample, but since the
552  * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
553  * the correct parameter for send_samples_bits is 8 bits per stream
554  * clock. */
555  return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
558  return rtp_send_samples(s1, pkt->data, size,
560  case AV_CODEC_ID_MP2:
561  case AV_CODEC_ID_MP3:
562  rtp_send_mpegaudio(s1, pkt->data, size);
563  break;
566  ff_rtp_send_mpegvideo(s1, pkt->data, size);
567  break;
568  case AV_CODEC_ID_AAC:
569  if (s->flags & FF_RTP_FLAG_MP4A_LATM)
570  ff_rtp_send_latm(s1, pkt->data, size);
571  else
572  ff_rtp_send_aac(s1, pkt->data, size);
573  break;
574  case AV_CODEC_ID_AMR_NB:
575  case AV_CODEC_ID_AMR_WB:
576  ff_rtp_send_amr(s1, pkt->data, size);
577  break;
578  case AV_CODEC_ID_MPEG2TS:
579  rtp_send_mpegts_raw(s1, pkt->data, size);
580  break;
581  case AV_CODEC_ID_DIRAC:
582  ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
583  break;
584  case AV_CODEC_ID_H264:
585  ff_rtp_send_h264_hevc(s1, pkt->data, size);
586  break;
587  case AV_CODEC_ID_H261:
588  ff_rtp_send_h261(s1, pkt->data, size);
589  break;
590  case AV_CODEC_ID_H263:
591  if (s->flags & FF_RTP_FLAG_RFC2190) {
592  int mb_info_size = 0;
593  const uint8_t *mb_info =
595  &mb_info_size);
596  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
597  break;
598  }
599  /* Fallthrough */
600  case AV_CODEC_ID_H263P:
601  ff_rtp_send_h263(s1, pkt->data, size);
602  break;
603  case AV_CODEC_ID_HEVC:
604  ff_rtp_send_h264_hevc(s1, pkt->data, size);
605  break;
606  case AV_CODEC_ID_VORBIS:
607  case AV_CODEC_ID_THEORA:
608  ff_rtp_send_xiph(s1, pkt->data, size);
609  break;
610  case AV_CODEC_ID_VP8:
611  ff_rtp_send_vp8(s1, pkt->data, size);
612  break;
613  case AV_CODEC_ID_VP9:
614  ff_rtp_send_vp9(s1, pkt->data, size);
615  break;
616  case AV_CODEC_ID_ILBC:
617  rtp_send_ilbc(s1, pkt->data, size);
618  break;
619  case AV_CODEC_ID_MJPEG:
620  ff_rtp_send_jpeg(s1, pkt->data, size);
621  break;
622  case AV_CODEC_ID_OPUS:
623  if (size > s->max_payload_size) {
624  av_log(s1, AV_LOG_ERROR,
625  "Packet size %d too large for max RTP payload size %d\n",
626  size, s->max_payload_size);
627  return AVERROR(EINVAL);
628  }
629  /* Intentional fallthrough */
630  default:
631  /* better than nothing : send the codec raw data */
632  rtp_send_raw(s1, pkt->data, size);
633  break;
634  }
635  return 0;
636 }
637 
639 {
640  RTPMuxContext *s = s1->priv_data;
641 
642  /* If the caller closes and recreates ->pb, this might actually
643  * be NULL here even if it was successfully allocated at the start. */
644  if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
645  rtcp_send_sr(s1, ff_ntp_time(), 1);
646  av_freep(&s->buf);
647 
648  return 0;
649 }
650 
652  .name = "rtp",
653  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
654  .priv_data_size = sizeof(RTPMuxContext),
655  .audio_codec = AV_CODEC_ID_PCM_MULAW,
656  .video_codec = AV_CODEC_ID_MPEG4,
660  .priv_class = &rtp_muxer_class,
662 };
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2565
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
Definition: ffmpeg.c:681
unsigned int packet_size
Definition: avformat.h:1429
#define NULL
Definition: coverity.c:32
enum AVFieldOrder field_order
Video only.
Definition: avcodec.h:3919
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:1556
AVOption.
Definition: opt.h:246
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
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:4748
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:355
#define NTP_OFFSET_US
Definition: internal.h:237
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: rtpenc.c:517
#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:3838
int size
Definition: avcodec.h:1415
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:193
#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:1193
int strict_std_compliance
Allow non-standard and experimental extension.
Definition: avformat.h:1611
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:479
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:1325
#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:879
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:1393
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:93
int flags
Flags modifying the (de)muxer behaviour.
Definition: avformat.h:1436
uint8_t * data
Definition: avcodec.h:1414
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:1453
#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:4563
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:241
uint32_t ssrc
Definition: rtpenc.h:32
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:215
#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
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:186
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:552
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:3834
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:1852
#define FF_RTP_FLAG_SKIP_RTCP
Definition: rtpenc.h:70
#define fail()
Definition: checkasm.h:113
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:458
int extradata_size
Size of the extradata content in bytes.
Definition: avcodec.h:3856
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1381
int block_align
Audio only.
Definition: avcodec.h:3955
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:332
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:507
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
Definition: rtpenc.c:482
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:384
preferred ID for MPEG-1/2 video decoding
Definition: avcodec.h:220
#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:1270
Stream structure.
Definition: avformat.h:872
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:651
int frame_size
Definition: mxfenc.c:1947
#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:1367
#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:473
int sample_rate
Audio only.
Definition: avcodec.h:3948
Main libavformat public API header.
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:436
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
Definition: rtpenc.c:285
#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:683
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:1353
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:337
static int rtp_write_trailer(AVFormatContext *s1)
Definition: rtpenc.c:638
int bits_per_coded_sample
The number of bits per sample in the codedwords.
Definition: avcodec.h:3880
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
Definition: avcodec.h:3852
int channels
Audio only.
Definition: avcodec.h:3944
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:375
#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
Codec parameters associated with this stream.
Definition: avformat.h:1014
int stream_index
Definition: avcodec.h:1416
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:901
This structure stores compressed data.
Definition: avcodec.h:1391
#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:1407
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248