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rtpenc.c
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1 /*
2  * RTP output format
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "avformat.h"
23 #include "mpegts.h"
24 #include "internal.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/random_seed.h"
27 #include "libavutil/opt.h"
28 
29 #include "rtpenc.h"
30 
31 static const AVOption options[] = {
33  { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
34  { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
35  { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
36  { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
37  { NULL },
38 };
39 
40 static const AVClass rtp_muxer_class = {
41  .class_name = "RTP muxer",
42  .item_name = av_default_item_name,
43  .option = options,
44  .version = LIBAVUTIL_VERSION_INT,
45 };
46 
47 #define RTCP_SR_SIZE 28
48 
49 static int is_supported(enum AVCodecID id)
50 {
51  switch(id) {
52  case AV_CODEC_ID_H261:
53  case AV_CODEC_ID_H263:
54  case AV_CODEC_ID_H263P:
55  case AV_CODEC_ID_H264:
56  case AV_CODEC_ID_HEVC:
59  case AV_CODEC_ID_MPEG4:
60  case AV_CODEC_ID_AAC:
61  case AV_CODEC_ID_MP2:
62  case AV_CODEC_ID_MP3:
65  case AV_CODEC_ID_PCM_S8:
70  case AV_CODEC_ID_PCM_U8:
72  case AV_CODEC_ID_AMR_NB:
73  case AV_CODEC_ID_AMR_WB:
74  case AV_CODEC_ID_VORBIS:
75  case AV_CODEC_ID_THEORA:
76  case AV_CODEC_ID_VP8:
79  case AV_CODEC_ID_ILBC:
80  case AV_CODEC_ID_MJPEG:
81  case AV_CODEC_ID_SPEEX:
82  case AV_CODEC_ID_OPUS:
83  return 1;
84  default:
85  return 0;
86  }
87 }
88 
90 {
91  RTPMuxContext *s = s1->priv_data;
92  int n, ret = AVERROR(EINVAL);
93  AVStream *st;
94 
95  if (s1->nb_streams != 1) {
96  av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
97  return AVERROR(EINVAL);
98  }
99  st = s1->streams[0];
100  if (!is_supported(st->codec->codec_id)) {
101  av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
102 
103  return -1;
104  }
105 
106  if (s->payload_type < 0) {
107  /* Re-validate non-dynamic payload types */
108  if (st->id < RTP_PT_PRIVATE)
109  st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
110 
111  s->payload_type = st->id;
112  } else {
113  /* private option takes priority */
114  st->id = s->payload_type;
115  }
116 
118  s->timestamp = s->base_timestamp;
119  s->cur_timestamp = 0;
120  if (!s->ssrc)
121  s->ssrc = av_get_random_seed();
122  s->first_packet = 1;
125  /* Round the NTP time to whole milliseconds. */
126  s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
128  // Pick a random sequence start number, but in the lower end of the
129  // available range, so that any wraparound doesn't happen immediately.
130  // (Immediate wraparound would be an issue for SRTP.)
131  if (s->seq < 0) {
132  if (s1->flags & AVFMT_FLAG_BITEXACT) {
133  s->seq = 0;
134  } else
135  s->seq = av_get_random_seed() & 0x0fff;
136  } else
137  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
138 
139  if (s1->packet_size) {
140  if (s1->pb->max_packet_size)
141  s1->packet_size = FFMIN(s1->packet_size,
142  s1->pb->max_packet_size);
143  } else
144  s1->packet_size = s1->pb->max_packet_size;
145  if (s1->packet_size <= 12) {
146  av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
147  return AVERROR(EIO);
148  }
149  s->buf = av_malloc(s1->packet_size);
150  if (!s->buf) {
151  return AVERROR(ENOMEM);
152  }
153  s->max_payload_size = s1->packet_size - 12;
154 
155  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
156  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
157  } else {
158  avpriv_set_pts_info(st, 32, 1, 90000);
159  }
160  s->buf_ptr = s->buf;
161  switch(st->codec->codec_id) {
162  case AV_CODEC_ID_MP2:
163  case AV_CODEC_ID_MP3:
164  s->buf_ptr = s->buf + 4;
165  avpriv_set_pts_info(st, 32, 1, 90000);
166  break;
169  break;
170  case AV_CODEC_ID_MPEG2TS:
172  if (n < 1)
173  n = 1;
175  break;
176  case AV_CODEC_ID_H261:
178  av_log(s, AV_LOG_ERROR,
179  "Packetizing H261 is experimental and produces incorrect "
180  "packetization for cases where GOBs don't fit into packets "
181  "(even though most receivers may handle it just fine). "
182  "Please set -f_strict experimental in order to enable it.\n");
183  ret = AVERROR_EXPERIMENTAL;
184  goto fail;
185  }
186  break;
187  case AV_CODEC_ID_H264:
188  /* check for H.264 MP4 syntax */
189  if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
190  s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
191  }
192  break;
193  case AV_CODEC_ID_HEVC:
194  /* Only check for the standardized hvcC version of extradata, keeping
195  * things simple and similar to the avcC/H264 case above, instead
196  * of trying to handle the pre-standardization versions (as in
197  * libavcodec/hevc.c). */
198  if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
199  s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
200  }
201  break;
202  case AV_CODEC_ID_VORBIS:
203  case AV_CODEC_ID_THEORA:
204  s->max_frames_per_packet = 15;
205  break;
207  /* Due to a historical error, the clock rate for G722 in RTP is
208  * 8000, even if the sample rate is 16000. See RFC 3551. */
209  avpriv_set_pts_info(st, 32, 1, 8000);
210  break;
211  case AV_CODEC_ID_OPUS:
212  if (st->codec->channels > 2) {
213  av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
214  goto fail;
215  }
216  /* The opus RTP RFC says that all opus streams should use 48000 Hz
217  * as clock rate, since all opus sample rates can be expressed in
218  * this clock rate, and sample rate changes on the fly are supported. */
219  avpriv_set_pts_info(st, 32, 1, 48000);
220  break;
221  case AV_CODEC_ID_ILBC:
222  if (st->codec->block_align != 38 && st->codec->block_align != 50) {
223  av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
224  goto fail;
225  }
227  break;
228  case AV_CODEC_ID_AMR_NB:
229  case AV_CODEC_ID_AMR_WB:
230  s->max_frames_per_packet = 50;
231  if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
232  n = 31;
233  else
234  n = 61;
235  /* max_header_toc_size + the largest AMR payload must fit */
236  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
237  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
238  goto fail;
239  }
240  if (st->codec->channels != 1) {
241  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
242  goto fail;
243  }
244  break;
245  case AV_CODEC_ID_AAC:
246  s->max_frames_per_packet = 50;
247  break;
248  default:
249  break;
250  }
251 
252  return 0;
253 
254 fail:
255  av_freep(&s->buf);
256  return ret;
257 }
258 
259 /* send an rtcp sender report packet */
260 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
261 {
262  RTPMuxContext *s = s1->priv_data;
263  uint32_t rtp_ts;
264 
265  av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
266 
267  s->last_rtcp_ntp_time = ntp_time;
268  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
269  s1->streams[0]->time_base) + s->base_timestamp;
270  avio_w8(s1->pb, RTP_VERSION << 6);
271  avio_w8(s1->pb, RTCP_SR);
272  avio_wb16(s1->pb, 6); /* length in words - 1 */
273  avio_wb32(s1->pb, s->ssrc);
274  avio_wb32(s1->pb, ntp_time / 1000000);
275  avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
276  avio_wb32(s1->pb, rtp_ts);
277  avio_wb32(s1->pb, s->packet_count);
278  avio_wb32(s1->pb, s->octet_count);
279 
280  if (s->cname) {
281  int len = FFMIN(strlen(s->cname), 255);
282  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
283  avio_w8(s1->pb, RTCP_SDES);
284  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
285 
286  avio_wb32(s1->pb, s->ssrc);
287  avio_w8(s1->pb, 0x01); /* CNAME */
288  avio_w8(s1->pb, len);
289  avio_write(s1->pb, s->cname, len);
290  avio_w8(s1->pb, 0); /* END */
291  for (len = (7 + len) % 4; len % 4; len++)
292  avio_w8(s1->pb, 0);
293  }
294 
295  if (bye) {
296  avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
297  avio_w8(s1->pb, RTCP_BYE);
298  avio_wb16(s1->pb, 1); /* length in words - 1 */
299  avio_wb32(s1->pb, s->ssrc);
300  }
301 
302  avio_flush(s1->pb);
303 }
304 
305 /* send an rtp packet. sequence number is incremented, but the caller
306  must update the timestamp itself */
307 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
308 {
309  RTPMuxContext *s = s1->priv_data;
310 
311  av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
312 
313  /* build the RTP header */
314  avio_w8(s1->pb, RTP_VERSION << 6);
315  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
316  avio_wb16(s1->pb, s->seq);
317  avio_wb32(s1->pb, s->timestamp);
318  avio_wb32(s1->pb, s->ssrc);
319 
320  avio_write(s1->pb, buf1, len);
321  avio_flush(s1->pb);
322 
323  s->seq = (s->seq + 1) & 0xffff;
324  s->octet_count += len;
325  s->packet_count++;
326 }
327 
328 /* send an integer number of samples and compute time stamp and fill
329  the rtp send buffer before sending. */
331  const uint8_t *buf1, int size, int sample_size_bits)
332 {
333  RTPMuxContext *s = s1->priv_data;
334  int len, max_packet_size, n;
335  /* Calculate the number of bytes to get samples aligned on a byte border */
336  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
337 
338  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
339  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
340  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
341  return AVERROR(EINVAL);
342  n = 0;
343  while (size > 0) {
344  s->buf_ptr = s->buf;
345  len = FFMIN(max_packet_size, size);
346 
347  /* copy data */
348  memcpy(s->buf_ptr, buf1, len);
349  s->buf_ptr += len;
350  buf1 += len;
351  size -= len;
352  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
353  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
354  n += (s->buf_ptr - s->buf);
355  }
356  return 0;
357 }
358 
360  const uint8_t *buf1, int size)
361 {
362  RTPMuxContext *s = s1->priv_data;
363  int len, count, max_packet_size;
364 
365  max_packet_size = s->max_payload_size;
366 
367  /* test if we must flush because not enough space */
368  len = (s->buf_ptr - s->buf);
369  if ((len + size) > max_packet_size) {
370  if (len > 4) {
371  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
372  s->buf_ptr = s->buf + 4;
373  }
374  }
375  if (s->buf_ptr == s->buf + 4) {
376  s->timestamp = s->cur_timestamp;
377  }
378 
379  /* add the packet */
380  if (size > max_packet_size) {
381  /* big packet: fragment */
382  count = 0;
383  while (size > 0) {
384  len = max_packet_size - 4;
385  if (len > size)
386  len = size;
387  /* build fragmented packet */
388  s->buf[0] = 0;
389  s->buf[1] = 0;
390  s->buf[2] = count >> 8;
391  s->buf[3] = count;
392  memcpy(s->buf + 4, buf1, len);
393  ff_rtp_send_data(s1, s->buf, len + 4, 0);
394  size -= len;
395  buf1 += len;
396  count += len;
397  }
398  } else {
399  if (s->buf_ptr == s->buf + 4) {
400  /* no fragmentation possible */
401  s->buf[0] = 0;
402  s->buf[1] = 0;
403  s->buf[2] = 0;
404  s->buf[3] = 0;
405  }
406  memcpy(s->buf_ptr, buf1, size);
407  s->buf_ptr += size;
408  }
409 }
410 
412  const uint8_t *buf1, int size)
413 {
414  RTPMuxContext *s = s1->priv_data;
415  int len, max_packet_size;
416 
417  max_packet_size = s->max_payload_size;
418 
419  while (size > 0) {
420  len = max_packet_size;
421  if (len > size)
422  len = size;
423 
424  s->timestamp = s->cur_timestamp;
425  ff_rtp_send_data(s1, buf1, len, (len == size));
426 
427  buf1 += len;
428  size -= len;
429  }
430 }
431 
432 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
434  const uint8_t *buf1, int size)
435 {
436  RTPMuxContext *s = s1->priv_data;
437  int len, out_len;
438 
439  s->timestamp = s->cur_timestamp;
440  while (size >= TS_PACKET_SIZE) {
441  len = s->max_payload_size - (s->buf_ptr - s->buf);
442  if (len > size)
443  len = size;
444  memcpy(s->buf_ptr, buf1, len);
445  buf1 += len;
446  size -= len;
447  s->buf_ptr += len;
448 
449  out_len = s->buf_ptr - s->buf;
450  if (out_len >= s->max_payload_size) {
451  ff_rtp_send_data(s1, s->buf, out_len, 0);
452  s->buf_ptr = s->buf;
453  }
454  }
455 }
456 
457 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
458 {
459  RTPMuxContext *s = s1->priv_data;
460  AVStream *st = s1->streams[0];
461  int frame_duration = av_get_audio_frame_duration(st->codec, 0);
462  int frame_size = st->codec->block_align;
463  int frames = size / frame_size;
464 
465  while (frames > 0) {
466  if (s->num_frames > 0 &&
468  s1->max_delay, AV_TIME_BASE_Q) >= 0) {
469  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
470  s->num_frames = 0;
471  }
472 
473  if (!s->num_frames) {
474  s->buf_ptr = s->buf;
475  s->timestamp = s->cur_timestamp;
476  }
477  memcpy(s->buf_ptr, buf, frame_size);
478  frames--;
479  s->num_frames++;
480  s->buf_ptr += frame_size;
481  buf += frame_size;
482  s->cur_timestamp += frame_duration;
483 
484  if (s->num_frames == s->max_frames_per_packet) {
485  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
486  s->num_frames = 0;
487  }
488  }
489  return 0;
490 }
491 
493 {
494  RTPMuxContext *s = s1->priv_data;
495  AVStream *st = s1->streams[0];
496  int rtcp_bytes;
497  int size= pkt->size;
498 
499  av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
500 
501  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
503  if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
504  (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
505  !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
506  rtcp_send_sr(s1, ff_ntp_time(), 0);
508  s->first_packet = 0;
509  }
510  s->cur_timestamp = s->base_timestamp + pkt->pts;
511 
512  switch(st->codec->codec_id) {
515  case AV_CODEC_ID_PCM_U8:
516  case AV_CODEC_ID_PCM_S8:
517  return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
522  return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
524  /* The actual sample size is half a byte per sample, but since the
525  * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
526  * the correct parameter for send_samples_bits is 8 bits per stream
527  * clock. */
528  return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
530  return rtp_send_samples(s1, pkt->data, size,
532  case AV_CODEC_ID_MP2:
533  case AV_CODEC_ID_MP3:
534  rtp_send_mpegaudio(s1, pkt->data, size);
535  break;
538  ff_rtp_send_mpegvideo(s1, pkt->data, size);
539  break;
540  case AV_CODEC_ID_AAC:
541  if (s->flags & FF_RTP_FLAG_MP4A_LATM)
542  ff_rtp_send_latm(s1, pkt->data, size);
543  else
544  ff_rtp_send_aac(s1, pkt->data, size);
545  break;
546  case AV_CODEC_ID_AMR_NB:
547  case AV_CODEC_ID_AMR_WB:
548  ff_rtp_send_amr(s1, pkt->data, size);
549  break;
550  case AV_CODEC_ID_MPEG2TS:
551  rtp_send_mpegts_raw(s1, pkt->data, size);
552  break;
553  case AV_CODEC_ID_H264:
554  ff_rtp_send_h264_hevc(s1, pkt->data, size);
555  break;
556  case AV_CODEC_ID_H261:
557  ff_rtp_send_h261(s1, pkt->data, size);
558  break;
559  case AV_CODEC_ID_H263:
560  if (s->flags & FF_RTP_FLAG_RFC2190) {
561  int mb_info_size = 0;
562  const uint8_t *mb_info =
564  &mb_info_size);
565  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
566  break;
567  }
568  /* Fallthrough */
569  case AV_CODEC_ID_H263P:
570  ff_rtp_send_h263(s1, pkt->data, size);
571  break;
572  case AV_CODEC_ID_HEVC:
573  ff_rtp_send_h264_hevc(s1, pkt->data, size);
574  break;
575  case AV_CODEC_ID_VORBIS:
576  case AV_CODEC_ID_THEORA:
577  ff_rtp_send_xiph(s1, pkt->data, size);
578  break;
579  case AV_CODEC_ID_VP8:
580  ff_rtp_send_vp8(s1, pkt->data, size);
581  break;
582  case AV_CODEC_ID_ILBC:
583  rtp_send_ilbc(s1, pkt->data, size);
584  break;
585  case AV_CODEC_ID_MJPEG:
586  ff_rtp_send_jpeg(s1, pkt->data, size);
587  break;
588  case AV_CODEC_ID_OPUS:
589  if (size > s->max_payload_size) {
590  av_log(s1, AV_LOG_ERROR,
591  "Packet size %d too large for max RTP payload size %d\n",
592  size, s->max_payload_size);
593  return AVERROR(EINVAL);
594  }
595  /* Intentional fallthrough */
596  default:
597  /* better than nothing : send the codec raw data */
598  rtp_send_raw(s1, pkt->data, size);
599  break;
600  }
601  return 0;
602 }
603 
605 {
606  RTPMuxContext *s = s1->priv_data;
607 
608  /* If the caller closes and recreates ->pb, this might actually
609  * be NULL here even if it was successfully allocated at the start. */
610  if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
611  rtcp_send_sr(s1, ff_ntp_time(), 1);
612  av_freep(&s->buf);
613 
614  return 0;
615 }
616 
618  .name = "rtp",
619  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
620  .priv_data_size = sizeof(RTPMuxContext),
621  .audio_codec = AV_CODEC_ID_PCM_MULAW,
622  .video_codec = AV_CODEC_ID_MPEG4,
626  .priv_class = &rtp_muxer_class,
628 };
unsigned int packet_size
Definition: avformat.h:1389
#define NULL
Definition: coverity.c:32
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:1510
AVOption.
Definition: opt.h:255
#define LIBAVUTIL_VERSION_INT
Definition: version.h:62
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:4093
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:330
#define NTP_OFFSET_US
Definition: internal.h:160
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: rtpenc.c:492
#define RTP_VERSION
Definition: rtp.h:78
int64_t last_rtcp_ntp_time
Definition: rtpenc.h:42
int size
Definition: avcodec.h:1434
#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:1264
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2309
int strict_std_compliance
Allow non-standard and experimental extension.
Definition: avformat.h:1565
#define AVFMT_TS_NONSTRICT
Format does not require strictly increasing timestamps, but they must still be monotonic.
Definition: avformat.h:495
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:1285
#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:861
int max_frames_per_packet
Definition: rtpenc.h:52
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1627
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
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1353
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:89
int flags
Flags modifying the (de)muxer behaviour.
Definition: avformat.h:1396
uint8_t * data
Definition: avcodec.h:1433
Definition: rtp.h:99
uint8_t * buf
Definition: rtpenc.h:49
ptrdiff_t size
Definition: opengl_enc.c:101
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:3006
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
Definition: aviobuf.c:178
#define AVFMT_FLAG_BITEXACT
When muxing, try to avoid writing any random/volatile data to the output.
Definition: avformat.h:1413
#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:285
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:3913
int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec, int idx)
Return the payload type for a given stream used in the given format context.
Definition: rtp.c:90
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:147
int max_packet_size
Definition: avio.h:141
uint32_t ssrc
Definition: rtpenc.h:32
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:102
#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:178
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:425
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)
Return the greatest common divisor of a and b.
Definition: mathematics.c:55
GLsizei count
Definition: opengl_enc.c:109
#define FF_RTP_FLAG_SKIP_RTCP
Definition: rtpenc.h:70
#define fail()
Definition: checkasm.h:57
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:152
static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:433
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:873
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1341
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2835
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:198
#define FFMIN(a, b)
Definition: common.h:92
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
Definition: rtpenc.c:307
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:525
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
Definition: rtpenc.c:457
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
static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:359
preferred ID for MPEG-1/2 video decoding
Definition: avcodec.h:107
#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:3047
Stream structure.
Definition: avformat.h:854
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:617
int frame_size
Definition: mxfenc.c:1819
enum AVMediaType codec_type
Definition: avcodec.h:1520
enum AVCodecID codec_id
Definition: avcodec.h:1529
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:252
Definition: rtp.h:100
int sample_rate
samples per second
Definition: avcodec.h:2272
AVIOContext * pb
I/O context.
Definition: avformat.h:1327
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:74
void avio_w8(AVIOContext *s, int b)
Definition: aviobuf.c:156
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
int extradata_size
Definition: avcodec.h:1628
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
int av_get_audio_frame_duration(AVCodecContext *avctx, int frame_bytes)
Return audio frame duration.
Definition: utils.c:3422
uint8_t * buf_ptr
Definition: rtpenc.h:50
void avio_wb16(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:427
static int flags
Definition: cpu.c:47
Main libavformat public API header.
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:411
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
Definition: rtpenc.c:260
#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:550
uint32_t timestamp
Definition: rtpenc.h:35
int len
int channels
number of audio channels
Definition: avcodec.h:2273
void * priv_data
Format private data.
Definition: avformat.h:1313
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:493
static int rtp_write_trailer(AVFormatContext *s1)
Definition: rtpenc.c:604
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:329
#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:329
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:116
int stream_index
Definition: avcodec.h:1435
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:896
const char * cname
Definition: rtpenc.h:33
This structure stores compressed data.
Definition: avcodec.h:1410
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:1426
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:240