Libav
rtpenc.c
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1 /*
2  * RTP output format
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of Libav.
6  *
7  * Libav 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  * Libav 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 Libav; 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_H263:
53  case AV_CODEC_ID_H263P:
54  case AV_CODEC_ID_H264:
57  case AV_CODEC_ID_MPEG4:
58  case AV_CODEC_ID_AAC:
59  case AV_CODEC_ID_MP2:
60  case AV_CODEC_ID_MP3:
63  case AV_CODEC_ID_PCM_S8:
68  case AV_CODEC_ID_PCM_U8:
70  case AV_CODEC_ID_AMR_NB:
71  case AV_CODEC_ID_AMR_WB:
72  case AV_CODEC_ID_VORBIS:
73  case AV_CODEC_ID_THEORA:
74  case AV_CODEC_ID_VP8:
77  case AV_CODEC_ID_ILBC:
78  case AV_CODEC_ID_MJPEG:
79  case AV_CODEC_ID_SPEEX:
80  case AV_CODEC_ID_OPUS:
81  return 1;
82  default:
83  return 0;
84  }
85 }
86 
88 {
89  RTPMuxContext *s = s1->priv_data;
90  int n;
91  AVStream *st;
92 
93  if (s1->nb_streams != 1) {
94  av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
95  return AVERROR(EINVAL);
96  }
97  st = s1->streams[0];
98  if (!is_supported(st->codec->codec_id)) {
99  av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
100 
101  return -1;
102  }
103 
104  if (s->payload_type < 0) {
105  /* Re-validate non-dynamic payload types */
106  if (st->id < RTP_PT_PRIVATE)
107  st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
108 
109  s->payload_type = st->id;
110  } else {
111  /* private option takes priority */
112  st->id = s->payload_type;
113  }
114 
116  s->timestamp = s->base_timestamp;
117  s->cur_timestamp = 0;
118  if (!s->ssrc)
119  s->ssrc = av_get_random_seed();
120  s->first_packet = 1;
122  if (s1->start_time_realtime)
123  /* Round the NTP time to whole milliseconds. */
124  s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
126  // Pick a random sequence start number, but in the lower end of the
127  // available range, so that any wraparound doesn't happen immediately.
128  // (Immediate wraparound would be an issue for SRTP.)
129  if (s->seq < 0)
130  s->seq = av_get_random_seed() & 0x0fff;
131  else
132  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
133 
134  if (s1->packet_size) {
135  if (s1->pb->max_packet_size)
136  s1->packet_size = FFMIN(s1->packet_size,
137  s1->pb->max_packet_size);
138  } else
139  s1->packet_size = s1->pb->max_packet_size;
140  if (s1->packet_size <= 12) {
141  av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
142  return AVERROR(EIO);
143  }
144  s->buf = av_malloc(s1->packet_size);
145  if (!s->buf) {
146  return AVERROR(ENOMEM);
147  }
148  s->max_payload_size = s1->packet_size - 12;
149 
150  s->max_frames_per_packet = 0;
151  if (s1->max_delay > 0) {
152  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
154  if (!frame_size)
155  frame_size = st->codec->frame_size;
156  if (frame_size == 0) {
157  av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
158  } else {
162  (AVRational){ frame_size, st->codec->sample_rate },
163  AV_ROUND_DOWN);
164  }
165  }
166  if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
167  /* FIXME: We should round down here... */
168  if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
170  (AVRational){1, 1000000},
171  av_inv_q(st->avg_frame_rate));
172  } else
173  s->max_frames_per_packet = 1;
174  }
175  }
176 
177  avpriv_set_pts_info(st, 32, 1, 90000);
178  switch(st->codec->codec_id) {
179  case AV_CODEC_ID_MP2:
180  case AV_CODEC_ID_MP3:
181  s->buf_ptr = s->buf + 4;
182  break;
185  break;
186  case AV_CODEC_ID_MPEG2TS:
188  if (n < 1)
189  n = 1;
191  s->buf_ptr = s->buf;
192  break;
193  case AV_CODEC_ID_H264:
194  /* check for H.264 MP4 syntax */
195  if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
196  s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
197  }
198  break;
199  case AV_CODEC_ID_VORBIS:
200  case AV_CODEC_ID_THEORA:
202  s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
203  s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
204  s->num_frames = 0;
205  goto defaultcase;
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  }
226  if (!s->max_frames_per_packet)
227  s->max_frames_per_packet = 1;
230  goto defaultcase;
231  case AV_CODEC_ID_AMR_NB:
232  case AV_CODEC_ID_AMR_WB:
233  if (!s->max_frames_per_packet)
234  s->max_frames_per_packet = 12;
235  if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
236  n = 31;
237  else
238  n = 61;
239  /* max_header_toc_size + the largest AMR payload must fit */
240  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
241  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
242  goto fail;
243  }
244  if (st->codec->channels != 1) {
245  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
246  goto fail;
247  }
248  case AV_CODEC_ID_AAC:
249  s->num_frames = 0;
250  default:
251 defaultcase:
252  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
253  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
254  }
255  s->buf_ptr = s->buf;
256  break;
257  }
258 
259  return 0;
260 
261 fail:
262  av_freep(&s->buf);
263  return AVERROR(EINVAL);
264 }
265 
266 /* send an rtcp sender report packet */
267 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
268 {
269  RTPMuxContext *s = s1->priv_data;
270  uint32_t rtp_ts;
271 
272  av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
273 
274  s->last_rtcp_ntp_time = ntp_time;
275  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
276  s1->streams[0]->time_base) + s->base_timestamp;
277  avio_w8(s1->pb, RTP_VERSION << 6);
278  avio_w8(s1->pb, RTCP_SR);
279  avio_wb16(s1->pb, 6); /* length in words - 1 */
280  avio_wb32(s1->pb, s->ssrc);
281  avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
282  avio_wb32(s1->pb, rtp_ts);
283  avio_wb32(s1->pb, s->packet_count);
284  avio_wb32(s1->pb, s->octet_count);
285 
286  if (s->cname) {
287  int len = FFMIN(strlen(s->cname), 255);
288  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
289  avio_w8(s1->pb, RTCP_SDES);
290  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
291 
292  avio_wb32(s1->pb, s->ssrc);
293  avio_w8(s1->pb, 0x01); /* CNAME */
294  avio_w8(s1->pb, len);
295  avio_write(s1->pb, s->cname, len);
296  avio_w8(s1->pb, 0); /* END */
297  for (len = (7 + len) % 4; len % 4; len++)
298  avio_w8(s1->pb, 0);
299  }
300 
301  if (bye) {
302  avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
303  avio_w8(s1->pb, RTCP_BYE);
304  avio_wb16(s1->pb, 1); /* length in words - 1 */
305  avio_wb32(s1->pb, s->ssrc);
306  }
307 
308  avio_flush(s1->pb);
309 }
310 
311 /* send an rtp packet. sequence number is incremented, but the caller
312  must update the timestamp itself */
313 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
314 {
315  RTPMuxContext *s = s1->priv_data;
316 
317  av_dlog(s1, "rtp_send_data size=%d\n", len);
318 
319  /* build the RTP header */
320  avio_w8(s1->pb, RTP_VERSION << 6);
321  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
322  avio_wb16(s1->pb, s->seq);
323  avio_wb32(s1->pb, s->timestamp);
324  avio_wb32(s1->pb, s->ssrc);
325 
326  avio_write(s1->pb, buf1, len);
327  avio_flush(s1->pb);
328 
329  s->seq = (s->seq + 1) & 0xffff;
330  s->octet_count += len;
331  s->packet_count++;
332 }
333 
334 /* send an integer number of samples and compute time stamp and fill
335  the rtp send buffer before sending. */
337  const uint8_t *buf1, int size, int sample_size_bits)
338 {
339  RTPMuxContext *s = s1->priv_data;
340  int len, max_packet_size, n;
341  /* Calculate the number of bytes to get samples aligned on a byte border */
342  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
343 
344  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
345  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
346  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
347  return AVERROR(EINVAL);
348  n = 0;
349  while (size > 0) {
350  s->buf_ptr = s->buf;
351  len = FFMIN(max_packet_size, size);
352 
353  /* copy data */
354  memcpy(s->buf_ptr, buf1, len);
355  s->buf_ptr += len;
356  buf1 += len;
357  size -= len;
358  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
359  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
360  n += (s->buf_ptr - s->buf);
361  }
362  return 0;
363 }
364 
366  const uint8_t *buf1, int size)
367 {
368  RTPMuxContext *s = s1->priv_data;
369  int len, count, max_packet_size;
370 
371  max_packet_size = s->max_payload_size;
372 
373  /* test if we must flush because not enough space */
374  len = (s->buf_ptr - s->buf);
375  if ((len + size) > max_packet_size) {
376  if (len > 4) {
377  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
378  s->buf_ptr = s->buf + 4;
379  }
380  }
381  if (s->buf_ptr == s->buf + 4) {
382  s->timestamp = s->cur_timestamp;
383  }
384 
385  /* add the packet */
386  if (size > max_packet_size) {
387  /* big packet: fragment */
388  count = 0;
389  while (size > 0) {
390  len = max_packet_size - 4;
391  if (len > size)
392  len = size;
393  /* build fragmented packet */
394  s->buf[0] = 0;
395  s->buf[1] = 0;
396  s->buf[2] = count >> 8;
397  s->buf[3] = count;
398  memcpy(s->buf + 4, buf1, len);
399  ff_rtp_send_data(s1, s->buf, len + 4, 0);
400  size -= len;
401  buf1 += len;
402  count += len;
403  }
404  } else {
405  if (s->buf_ptr == s->buf + 4) {
406  /* no fragmentation possible */
407  s->buf[0] = 0;
408  s->buf[1] = 0;
409  s->buf[2] = 0;
410  s->buf[3] = 0;
411  }
412  memcpy(s->buf_ptr, buf1, size);
413  s->buf_ptr += size;
414  }
415 }
416 
418  const uint8_t *buf1, int size)
419 {
420  RTPMuxContext *s = s1->priv_data;
421  int len, max_packet_size;
422 
423  max_packet_size = s->max_payload_size;
424 
425  while (size > 0) {
426  len = max_packet_size;
427  if (len > size)
428  len = size;
429 
430  s->timestamp = s->cur_timestamp;
431  ff_rtp_send_data(s1, buf1, len, (len == size));
432 
433  buf1 += len;
434  size -= len;
435  }
436 }
437 
438 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
440  const uint8_t *buf1, int size)
441 {
442  RTPMuxContext *s = s1->priv_data;
443  int len, out_len;
444 
445  while (size >= TS_PACKET_SIZE) {
446  len = s->max_payload_size - (s->buf_ptr - s->buf);
447  if (len > size)
448  len = size;
449  memcpy(s->buf_ptr, buf1, len);
450  buf1 += len;
451  size -= len;
452  s->buf_ptr += len;
453 
454  out_len = s->buf_ptr - s->buf;
455  if (out_len >= s->max_payload_size) {
456  ff_rtp_send_data(s1, s->buf, out_len, 0);
457  s->buf_ptr = s->buf;
458  }
459  }
460 }
461 
462 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
463 {
464  RTPMuxContext *s = s1->priv_data;
465  AVStream *st = s1->streams[0];
466  int frame_duration = av_get_audio_frame_duration(st->codec, 0);
467  int frame_size = st->codec->block_align;
468  int frames = size / frame_size;
469 
470  while (frames > 0) {
471  int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
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, n * frame_size);
478  frames -= n;
479  s->num_frames += n;
480  s->buf_ptr += n * frame_size;
481  buf += n * frame_size;
482  s->cur_timestamp += n * 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_dlog(s1, "%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(s1, pkt->data, size);
555  break;
556  case AV_CODEC_ID_H263:
557  if (s->flags & FF_RTP_FLAG_RFC2190) {
558  int mb_info_size = 0;
559  const uint8_t *mb_info =
561  &mb_info_size);
562  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
563  break;
564  }
565  /* Fallthrough */
566  case AV_CODEC_ID_H263P:
567  ff_rtp_send_h263(s1, pkt->data, size);
568  break;
569  case AV_CODEC_ID_VORBIS:
570  case AV_CODEC_ID_THEORA:
571  ff_rtp_send_xiph(s1, pkt->data, size);
572  break;
573  case AV_CODEC_ID_VP8:
574  ff_rtp_send_vp8(s1, pkt->data, size);
575  break;
576  case AV_CODEC_ID_ILBC:
577  rtp_send_ilbc(s1, pkt->data, size);
578  break;
579  case AV_CODEC_ID_MJPEG:
580  ff_rtp_send_jpeg(s1, pkt->data, size);
581  break;
582  case AV_CODEC_ID_OPUS:
583  if (size > s->max_payload_size) {
584  av_log(s1, AV_LOG_ERROR,
585  "Packet size %d too large for max RTP payload size %d\n",
586  size, s->max_payload_size);
587  return AVERROR(EINVAL);
588  }
589  /* Intentional fallthrough */
590  default:
591  /* better than nothing : send the codec raw data */
592  rtp_send_raw(s1, pkt->data, size);
593  break;
594  }
595  return 0;
596 }
597 
599 {
600  RTPMuxContext *s = s1->priv_data;
601 
602  /* If the caller closes and recreates ->pb, this might actually
603  * be NULL here even if it was successfully allocated at the start. */
604  if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
605  rtcp_send_sr(s1, ff_ntp_time(), 1);
606  av_freep(&s->buf);
607 
608  return 0;
609 }
610 
612  .name = "rtp",
613  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
614  .priv_data_size = sizeof(RTPMuxContext),
615  .audio_codec = AV_CODEC_ID_PCM_MULAW,
616  .video_codec = AV_CODEC_ID_MPEG4,
620  .priv_class = &rtp_muxer_class,
621 };
unsigned int packet_size
Definition: avformat.h:1026
void avio_wb64(AVIOContext *s, uint64_t val)
Definition: aviobuf.c:330
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:62
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:1138
int size
AVOption.
Definition: opt.h:234
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:2829
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:336
#define NTP_OFFSET_US
Definition: internal.h:96
static int write_packet(AVFormatContext *s, AVPacket *pkt)
Definition: assenc.c:58
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 num
numerator
Definition: rational.h:44
int size
Definition: avcodec.h:974
#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
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#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:898
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1828
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:67
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
Format I/O context.
Definition: avformat.h:922
#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:38
uint8_t
AVOptions.
static const AVClass rtp_muxer_class
Definition: rtpenc.c:40
int id
Format-specific stream ID.
Definition: avformat.h:706
int max_frames_per_packet
Definition: rtpenc.h:52
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1164
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:990
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:87
uint8_t * data
Definition: avcodec.h:973
static int flags
Definition: log.c:44
Definition: rtp.h:99
uint8_t * buf
Definition: rtpenc.h:49
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:2507
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
Definition: aviobuf.c:165
static int write_trailer(AVFormatContext *s)
Definition: assenc.c:64
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
Definition: opt.h:264
#define FF_RTP_FLAG_RFC2190
Definition: rtpenc.h:68
uint64_t ff_ntp_time(void)
Get the current time since NTP epoch in microseconds.
Definition: utils.c:2655
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:89
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:129
static const uint8_t frame_size[4]
Definition: g723_1_data.h:47
int max_packet_size
Definition: avio.h:98
uint32_t ssrc
Definition: rtpenc.h:32
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:105
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_aac.c:25
void ff_rtp_send_jpeg(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_jpeg.c:27
#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:150
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:379
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:53
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:168
AVRational avg_frame_rate
Average framerate.
Definition: avformat.h:780
#define FF_RTP_FLAG_SKIP_RTCP
Definition: rtpenc.h:69
static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:439
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:718
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:978
int64_t av_rescale_q_rnd(int64_t a, AVRational bq, AVRational cq, enum AVRounding rnd)
Rescale a 64-bit integer by 2 rational numbers with specified rounding.
Definition: mathematics.c:121
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:30
int void avio_flush(AVIOContext *s)
Definition: aviobuf.c:180
#define FFMIN(a, b)
Definition: common.h:57
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
Definition: rtpenc.c:313
const char * name
Definition: avformat.h:446
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
Definition: rtpenc.c:462
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:365
preferred ID for MPEG-1/2 video decoding
Definition: avcodec.h:110
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
Stream structure.
Definition: avformat.h:699
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1811
int64_t first_rtcp_ntp_time
Definition: rtpenc.h:43
uint32_t cur_timestamp
Definition: rtpenc.h:37
NULL
Definition: eval.c:55
AVOutputFormat ff_rtp_muxer
Definition: rtpenc.c:611
enum AVMediaType codec_type
Definition: avcodec.h:1058
enum AVCodecID codec_id
Definition: avcodec.h:1067
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avutil.h:240
Definition: rtp.h:100
int sample_rate
samples per second
Definition: avcodec.h:1791
AVIOContext * pb
I/O context.
Definition: avformat.h:964
av_default_item_name
Definition: dnxhdenc.c:52
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:73
void avio_w8(AVIOContext *s, int b)
Definition: aviobuf.c:144
static int is_supported(enum AVCodecID id)
Definition: rtpenc.c:49
int first_packet
Definition: rtpenc.h:47
int extradata_size
Definition: avcodec.h:1165
Describe the class of an AVClass context structure.
Definition: log.h:33
rational number numerator/denominator
Definition: rational.h:43
int flags
Definition: rtpenc.h:60
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:2061
Round toward -infinity.
Definition: mathematics.h:52
uint8_t * buf_ptr
Definition: rtpenc.h:50
void avio_wb16(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:342
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
Definition: rational.h:122
#define NTP_TO_RTP_FORMAT(x)
Definition: rtp.h:113
void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc_h264.c:84
Main libavformat public API header.
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:417
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
Definition: rtpenc.c:267
#define RTCP_SR_SIZE
Definition: rtpenc.c:47
int den
denominator
Definition: rational.h:45
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:466
uint32_t timestamp
Definition: rtpenc.h:35
int len
int channels
number of audio channels
Definition: avcodec.h:1792
void * priv_data
Format private data.
Definition: avformat.h:950
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:380
static int rtp_write_trailer(AVFormatContext *s1)
Definition: rtpenc.c:598
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:268
uint8_t * av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:287
uint32_t av_get_random_seed(void)
Get random data.
Definition: random_seed.c:95
int stream_index
Definition: avcodec.h:975
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:741
const char * cname
Definition: rtpenc.h:33
This structure stores compressed data.
Definition: avcodec.h:950
#define FF_RTP_FLAG_SEND_BYE
Definition: rtpenc.h:71
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:966