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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:
81  case AV_CODEC_ID_ILBC:
82  case AV_CODEC_ID_MJPEG:
83  case AV_CODEC_ID_SPEEX:
84  case AV_CODEC_ID_OPUS:
85  return 1;
86  default:
87  return 0;
88  }
89 }
90 
92 {
93  RTPMuxContext *s = s1->priv_data;
94  int n, ret = AVERROR(EINVAL);
95  AVStream *st;
96 
97  if (s1->nb_streams != 1) {
98  av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
99  return AVERROR(EINVAL);
100  }
101  st = s1->streams[0];
102  if (!is_supported(st->codecpar->codec_id)) {
103  av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
104 
105  return -1;
106  }
107 
108  if (s->payload_type < 0) {
109  /* Re-validate non-dynamic payload types */
110  if (st->id < RTP_PT_PRIVATE)
111  st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
112 
113  s->payload_type = st->id;
114  } else {
115  /* private option takes priority */
116  st->id = s->payload_type;
117  }
118 
120  s->timestamp = s->base_timestamp;
121  s->cur_timestamp = 0;
122  if (!s->ssrc)
123  s->ssrc = av_get_random_seed();
124  s->first_packet = 1;
127  /* Round the NTP time to whole milliseconds. */
128  s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
130  // Pick a random sequence start number, but in the lower end of the
131  // available range, so that any wraparound doesn't happen immediately.
132  // (Immediate wraparound would be an issue for SRTP.)
133  if (s->seq < 0) {
134  if (s1->flags & AVFMT_FLAG_BITEXACT) {
135  s->seq = 0;
136  } else
137  s->seq = av_get_random_seed() & 0x0fff;
138  } else
139  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
140 
141  if (s1->packet_size) {
142  if (s1->pb->max_packet_size)
143  s1->packet_size = FFMIN(s1->packet_size,
144  s1->pb->max_packet_size);
145  } else
146  s1->packet_size = s1->pb->max_packet_size;
147  if (s1->packet_size <= 12) {
148  av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
149  return AVERROR(EIO);
150  }
151  s->buf = av_malloc(s1->packet_size);
152  if (!s->buf) {
153  return AVERROR(ENOMEM);
154  }
155  s->max_payload_size = s1->packet_size - 12;
156 
157  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
158  avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
159  } else {
160  avpriv_set_pts_info(st, 32, 1, 90000);
161  }
162  s->buf_ptr = s->buf;
163  switch(st->codecpar->codec_id) {
164  case AV_CODEC_ID_MP2:
165  case AV_CODEC_ID_MP3:
166  s->buf_ptr = s->buf + 4;
167  avpriv_set_pts_info(st, 32, 1, 90000);
168  break;
171  break;
172  case AV_CODEC_ID_MPEG2TS:
174  if (n < 1)
175  n = 1;
177  break;
178  case AV_CODEC_ID_DIRAC:
180  av_log(s, AV_LOG_ERROR,
181  "Packetizing VC-2 is experimental and does not use all values "
182  "of the specification "
183  "(even though most receivers may handle it just fine). "
184  "Please set -strict experimental in order to enable it.\n");
185  ret = AVERROR_EXPERIMENTAL;
186  goto fail;
187  }
188  break;
189  case AV_CODEC_ID_H261:
191  av_log(s, AV_LOG_ERROR,
192  "Packetizing H.261 is experimental and produces incorrect "
193  "packetization for cases where GOBs don't fit into packets "
194  "(even though most receivers may handle it just fine). "
195  "Please set -f_strict experimental in order to enable it.\n");
196  ret = AVERROR_EXPERIMENTAL;
197  goto fail;
198  }
199  break;
200  case AV_CODEC_ID_H264:
201  /* check for H.264 MP4 syntax */
202  if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
203  s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
204  }
205  break;
206  case AV_CODEC_ID_HEVC:
207  /* Only check for the standardized hvcC version of extradata, keeping
208  * things simple and similar to the avcC/H.264 case above, instead
209  * of trying to handle the pre-standardization versions (as in
210  * libavcodec/hevc.c). */
211  if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
212  s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
213  }
214  break;
215  case AV_CODEC_ID_VP9:
217  av_log(s, AV_LOG_ERROR,
218  "Packetizing VP9 is experimental and its specification is "
219  "still in draft state. "
220  "Please set -strict experimental in order to enable it.\n");
221  ret = AVERROR_EXPERIMENTAL;
222  goto fail;
223  }
224  break;
225  case AV_CODEC_ID_VORBIS:
226  case AV_CODEC_ID_THEORA:
227  s->max_frames_per_packet = 15;
228  break;
230  /* Due to a historical error, the clock rate for G722 in RTP is
231  * 8000, even if the sample rate is 16000. See RFC 3551. */
232  avpriv_set_pts_info(st, 32, 1, 8000);
233  break;
234  case AV_CODEC_ID_OPUS:
235  if (st->codecpar->channels > 2) {
236  av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
237  goto fail;
238  }
239  /* The opus RTP RFC says that all opus streams should use 48000 Hz
240  * as clock rate, since all opus sample rates can be expressed in
241  * this clock rate, and sample rate changes on the fly are supported. */
242  avpriv_set_pts_info(st, 32, 1, 48000);
243  break;
244  case AV_CODEC_ID_ILBC:
245  if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
246  av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
247  goto fail;
248  }
250  break;
251  case AV_CODEC_ID_AMR_NB:
252  case AV_CODEC_ID_AMR_WB:
253  s->max_frames_per_packet = 50;
255  n = 31;
256  else
257  n = 61;
258  /* max_header_toc_size + the largest AMR payload must fit */
259  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
260  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
261  goto fail;
262  }
263  if (st->codecpar->channels != 1) {
264  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
265  goto fail;
266  }
267  break;
268  case AV_CODEC_ID_AAC:
269  s->max_frames_per_packet = 50;
270  break;
271  default:
272  break;
273  }
274 
275  return 0;
276 
277 fail:
278  av_freep(&s->buf);
279  return ret;
280 }
281 
282 /* send an rtcp sender report packet */
283 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
284 {
285  RTPMuxContext *s = s1->priv_data;
286  uint32_t rtp_ts;
287 
288  av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
289 
290  s->last_rtcp_ntp_time = ntp_time;
291  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
292  s1->streams[0]->time_base) + s->base_timestamp;
293  avio_w8(s1->pb, RTP_VERSION << 6);
294  avio_w8(s1->pb, RTCP_SR);
295  avio_wb16(s1->pb, 6); /* length in words - 1 */
296  avio_wb32(s1->pb, s->ssrc);
297  avio_wb32(s1->pb, ntp_time / 1000000);
298  avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
299  avio_wb32(s1->pb, rtp_ts);
300  avio_wb32(s1->pb, s->packet_count);
301  avio_wb32(s1->pb, s->octet_count);
302 
303  if (s->cname) {
304  int len = FFMIN(strlen(s->cname), 255);
305  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
306  avio_w8(s1->pb, RTCP_SDES);
307  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
308 
309  avio_wb32(s1->pb, s->ssrc);
310  avio_w8(s1->pb, 0x01); /* CNAME */
311  avio_w8(s1->pb, len);
312  avio_write(s1->pb, s->cname, len);
313  avio_w8(s1->pb, 0); /* END */
314  for (len = (7 + len) % 4; len % 4; len++)
315  avio_w8(s1->pb, 0);
316  }
317 
318  if (bye) {
319  avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
320  avio_w8(s1->pb, RTCP_BYE);
321  avio_wb16(s1->pb, 1); /* length in words - 1 */
322  avio_wb32(s1->pb, s->ssrc);
323  }
324 
325  avio_flush(s1->pb);
326 }
327 
328 /* send an rtp packet. sequence number is incremented, but the caller
329  must update the timestamp itself */
330 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
331 {
332  RTPMuxContext *s = s1->priv_data;
333 
334  av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
335 
336  /* build the RTP header */
337  avio_w8(s1->pb, RTP_VERSION << 6);
338  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
339  avio_wb16(s1->pb, s->seq);
340  avio_wb32(s1->pb, s->timestamp);
341  avio_wb32(s1->pb, s->ssrc);
342 
343  avio_write(s1->pb, buf1, len);
344  avio_flush(s1->pb);
345 
346  s->seq = (s->seq + 1) & 0xffff;
347  s->octet_count += len;
348  s->packet_count++;
349 }
350 
351 /* send an integer number of samples and compute time stamp and fill
352  the rtp send buffer before sending. */
354  const uint8_t *buf1, int size, int sample_size_bits)
355 {
356  RTPMuxContext *s = s1->priv_data;
357  int len, max_packet_size, n;
358  /* Calculate the number of bytes to get samples aligned on a byte border */
359  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
360 
361  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
362  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
363  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
364  return AVERROR(EINVAL);
365  n = 0;
366  while (size > 0) {
367  s->buf_ptr = s->buf;
368  len = FFMIN(max_packet_size, size);
369 
370  /* copy data */
371  memcpy(s->buf_ptr, buf1, len);
372  s->buf_ptr += len;
373  buf1 += len;
374  size -= len;
375  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
376  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
377  n += (s->buf_ptr - s->buf);
378  }
379  return 0;
380 }
381 
383  const uint8_t *buf1, int size)
384 {
385  RTPMuxContext *s = s1->priv_data;
386  int len, count, max_packet_size;
387 
388  max_packet_size = s->max_payload_size;
389 
390  /* test if we must flush because not enough space */
391  len = (s->buf_ptr - s->buf);
392  if ((len + size) > max_packet_size) {
393  if (len > 4) {
394  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
395  s->buf_ptr = s->buf + 4;
396  }
397  }
398  if (s->buf_ptr == s->buf + 4) {
399  s->timestamp = s->cur_timestamp;
400  }
401 
402  /* add the packet */
403  if (size > max_packet_size) {
404  /* big packet: fragment */
405  count = 0;
406  while (size > 0) {
407  len = max_packet_size - 4;
408  if (len > size)
409  len = size;
410  /* build fragmented packet */
411  s->buf[0] = 0;
412  s->buf[1] = 0;
413  s->buf[2] = count >> 8;
414  s->buf[3] = count;
415  memcpy(s->buf + 4, buf1, len);
416  ff_rtp_send_data(s1, s->buf, len + 4, 0);
417  size -= len;
418  buf1 += len;
419  count += len;
420  }
421  } else {
422  if (s->buf_ptr == s->buf + 4) {
423  /* no fragmentation possible */
424  s->buf[0] = 0;
425  s->buf[1] = 0;
426  s->buf[2] = 0;
427  s->buf[3] = 0;
428  }
429  memcpy(s->buf_ptr, buf1, size);
430  s->buf_ptr += size;
431  }
432 }
433 
435  const uint8_t *buf1, int size)
436 {
437  RTPMuxContext *s = s1->priv_data;
438  int len, max_packet_size;
439 
440  max_packet_size = s->max_payload_size;
441 
442  while (size > 0) {
443  len = max_packet_size;
444  if (len > size)
445  len = size;
446 
447  s->timestamp = s->cur_timestamp;
448  ff_rtp_send_data(s1, buf1, len, (len == size));
449 
450  buf1 += len;
451  size -= len;
452  }
453 }
454 
455 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
457  const uint8_t *buf1, int size)
458 {
459  RTPMuxContext *s = s1->priv_data;
460  int len, out_len;
461 
462  s->timestamp = s->cur_timestamp;
463  while (size >= TS_PACKET_SIZE) {
464  len = s->max_payload_size - (s->buf_ptr - s->buf);
465  if (len > size)
466  len = size;
467  memcpy(s->buf_ptr, buf1, len);
468  buf1 += len;
469  size -= len;
470  s->buf_ptr += len;
471 
472  out_len = s->buf_ptr - s->buf;
473  if (out_len >= s->max_payload_size) {
474  ff_rtp_send_data(s1, s->buf, out_len, 0);
475  s->buf_ptr = s->buf;
476  }
477  }
478 }
479 
480 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
481 {
482  RTPMuxContext *s = s1->priv_data;
483  AVStream *st = s1->streams[0];
484  int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
485  int frame_size = st->codecpar->block_align;
486  int frames = size / frame_size;
487 
488  while (frames > 0) {
489  if (s->num_frames > 0 &&
491  s1->max_delay, AV_TIME_BASE_Q) >= 0) {
492  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
493  s->num_frames = 0;
494  }
495 
496  if (!s->num_frames) {
497  s->buf_ptr = s->buf;
498  s->timestamp = s->cur_timestamp;
499  }
500  memcpy(s->buf_ptr, buf, frame_size);
501  frames--;
502  s->num_frames++;
503  s->buf_ptr += frame_size;
504  buf += frame_size;
505  s->cur_timestamp += frame_duration;
506 
507  if (s->num_frames == s->max_frames_per_packet) {
508  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
509  s->num_frames = 0;
510  }
511  }
512  return 0;
513 }
514 
516 {
517  RTPMuxContext *s = s1->priv_data;
518  AVStream *st = s1->streams[0];
519  int rtcp_bytes;
520  int size= pkt->size;
521 
522  av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
523 
524  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
526  if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
527  (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
528  !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
529  rtcp_send_sr(s1, ff_ntp_time(), 0);
531  s->first_packet = 0;
532  }
533  s->cur_timestamp = s->base_timestamp + pkt->pts;
534 
535  switch(st->codecpar->codec_id) {
538  case AV_CODEC_ID_PCM_U8:
539  case AV_CODEC_ID_PCM_S8:
540  return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
545  return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
547  /* The actual sample size is half a byte per sample, but since the
548  * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
549  * the correct parameter for send_samples_bits is 8 bits per stream
550  * clock. */
551  return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
553  return rtp_send_samples(s1, pkt->data, size,
555  case AV_CODEC_ID_MP2:
556  case AV_CODEC_ID_MP3:
557  rtp_send_mpegaudio(s1, pkt->data, size);
558  break;
561  ff_rtp_send_mpegvideo(s1, pkt->data, size);
562  break;
563  case AV_CODEC_ID_AAC:
564  if (s->flags & FF_RTP_FLAG_MP4A_LATM)
565  ff_rtp_send_latm(s1, pkt->data, size);
566  else
567  ff_rtp_send_aac(s1, pkt->data, size);
568  break;
569  case AV_CODEC_ID_AMR_NB:
570  case AV_CODEC_ID_AMR_WB:
571  ff_rtp_send_amr(s1, pkt->data, size);
572  break;
573  case AV_CODEC_ID_MPEG2TS:
574  rtp_send_mpegts_raw(s1, pkt->data, size);
575  break;
576  case AV_CODEC_ID_DIRAC:
577  ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
578  break;
579  case AV_CODEC_ID_H264:
580  ff_rtp_send_h264_hevc(s1, pkt->data, size);
581  break;
582  case AV_CODEC_ID_H261:
583  ff_rtp_send_h261(s1, pkt->data, size);
584  break;
585  case AV_CODEC_ID_H263:
586  if (s->flags & FF_RTP_FLAG_RFC2190) {
587  int mb_info_size = 0;
588  const uint8_t *mb_info =
590  &mb_info_size);
591  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
592  break;
593  }
594  /* Fallthrough */
595  case AV_CODEC_ID_H263P:
596  ff_rtp_send_h263(s1, pkt->data, size);
597  break;
598  case AV_CODEC_ID_HEVC:
599  ff_rtp_send_h264_hevc(s1, pkt->data, size);
600  break;
601  case AV_CODEC_ID_VORBIS:
602  case AV_CODEC_ID_THEORA:
603  ff_rtp_send_xiph(s1, pkt->data, size);
604  break;
605  case AV_CODEC_ID_VP8:
606  ff_rtp_send_vp8(s1, pkt->data, size);
607  break;
608  case AV_CODEC_ID_VP9:
609  ff_rtp_send_vp9(s1, pkt->data, size);
610  break;
611  case AV_CODEC_ID_ILBC:
612  rtp_send_ilbc(s1, pkt->data, size);
613  break;
614  case AV_CODEC_ID_MJPEG:
615  ff_rtp_send_jpeg(s1, pkt->data, size);
616  break;
617  case AV_CODEC_ID_OPUS:
618  if (size > s->max_payload_size) {
619  av_log(s1, AV_LOG_ERROR,
620  "Packet size %d too large for max RTP payload size %d\n",
621  size, s->max_payload_size);
622  return AVERROR(EINVAL);
623  }
624  /* Intentional fallthrough */
625  default:
626  /* better than nothing : send the codec raw data */
627  rtp_send_raw(s1, pkt->data, size);
628  break;
629  }
630  return 0;
631 }
632 
634 {
635  RTPMuxContext *s = s1->priv_data;
636 
637  /* If the caller closes and recreates ->pb, this might actually
638  * be NULL here even if it was successfully allocated at the start. */
639  if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
640  rtcp_send_sr(s1, ff_ntp_time(), 1);
641  av_freep(&s->buf);
642 
643  return 0;
644 }
645 
647  .name = "rtp",
648  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
649  .priv_data_size = sizeof(RTPMuxContext),
650  .audio_codec = AV_CODEC_ID_PCM_MULAW,
651  .video_codec = AV_CODEC_ID_MPEG4,
655  .priv_class = &rtp_muxer_class,
657 };
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2871
unsigned int packet_size
Definition: avformat.h:1429
#define NULL
Definition: coverity.c:32
enum AVFieldOrder field_order
Video only.
Definition: avcodec.h:4003
const char * s
Definition: avisynth_c.h:631
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:1552
AVOption.
Definition: opt.h:245
#define LIBAVUTIL_VERSION_INT
Definition: version.h:70
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:4436
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:353
#define NTP_OFFSET_US
Definition: internal.h:207
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: rtpenc.c:515
#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:3922
int size
Definition: avcodec.h:1581
#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:1389
int strict_std_compliance
Allow non-standard and experimental extension.
Definition: avformat.h:1607
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:494
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:883
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:91
int flags
Flags modifying the (de)muxer behaviour.
Definition: avformat.h:1436
uint8_t * data
Definition: avcodec.h:1580
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:204
#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:275
unsigned m
Definition: audioconvert.c:187
#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:4256
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:224
uint32_t ssrc
Definition: rtpenc.h:32
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:189
#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
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:515
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:3918
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 a and b.
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:3657
#define FF_RTP_FLAG_SKIP_RTCP
Definition: rtpenc.h:70
#define fail()
Definition: checkasm.h:81
int av_compare_ts(int64_t ts_a, AVRational tb_a, int64_t ts_b, AVRational tb_b)
Compare 2 timestamps each in its own timebases.
Definition: mathematics.c:147
static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:456
int extradata_size
Size of the extradata content in bytes.
Definition: avcodec.h:3940
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1381
int block_align
Audio only.
Definition: avcodec.h:4039
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:224
#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:330
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:522
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
Definition: rtpenc.c:480
int n
Definition: avisynth_c.h:547
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:382
preferred ID for MPEG-1/2 video decoding
Definition: avcodec.h:194
#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:3082
Stream structure.
Definition: avformat.h:876
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:646
int frame_size
Definition: mxfenc.c:1821
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:252
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:182
static int is_supported(enum AVCodecID id)
Definition: rtpenc.c:49
int first_packet
Definition: rtpenc.h:47
void * buf
Definition: avisynth_c.h:553
Describe the class of an AVClass context structure.
Definition: log.h:67
rational number numerator/denominator
Definition: rational.h:43
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:452
static int flags
Definition: cpu.c:47
int sample_rate
Audio only.
Definition: avcodec.h:4032
Main libavformat public API header.
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:434
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
Definition: rtpenc.c:283
#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:644
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:498
static int rtp_write_trailer(AVFormatContext *s1)
Definition: rtpenc.c:633
int bits_per_coded_sample
The number of bits per sample in the codedwords.
Definition: avcodec.h:3964
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
Definition: avcodec.h:3936
int channels
Audio only.
Definition: avcodec.h:4028
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:354
#define av_freep(p)
uint8_t * av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:334
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:114
AVCodecParameters * codecpar
Definition: avformat.h:1228
int stream_index
Definition: avcodec.h:1582
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:913
This structure stores compressed data.
Definition: avcodec.h:1557
static int write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: v4l2enc.c:86
#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:1573
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:240