FFmpeg  2.1.1
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_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 %s\n", avcodec_get_name(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  if (st->codec->flags & CODEC_FLAG_BITEXACT) {
131  s->seq = 0;
132  } else
133  s->seq = av_get_random_seed() & 0x0fff;
134  } else
135  s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
136 
137  if (s1->packet_size) {
138  if (s1->pb->max_packet_size)
139  s1->packet_size = FFMIN(s1->packet_size,
140  s1->pb->max_packet_size);
141  } else
142  s1->packet_size = s1->pb->max_packet_size;
143  if (s1->packet_size <= 12) {
144  av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
145  return AVERROR(EIO);
146  }
147  s->buf = av_malloc(s1->packet_size);
148  if (s->buf == NULL) {
149  return AVERROR(ENOMEM);
150  }
151  s->max_payload_size = s1->packet_size - 12;
152 
153  s->max_frames_per_packet = 0;
154  if (s1->max_delay > 0) {
155  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
157  if (!frame_size)
158  frame_size = st->codec->frame_size;
159  if (frame_size == 0) {
160  av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
161  } else {
165  (AVRational){ frame_size, st->codec->sample_rate },
166  AV_ROUND_DOWN);
167  }
168  }
169  if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
170  /* FIXME: We should round down here... */
171  s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
172  }
173  }
174 
175  avpriv_set_pts_info(st, 32, 1, 90000);
176  switch(st->codec->codec_id) {
177  case AV_CODEC_ID_MP2:
178  case AV_CODEC_ID_MP3:
179  s->buf_ptr = s->buf + 4;
180  break;
183  break;
184  case AV_CODEC_ID_MPEG2TS:
185  n = s->max_payload_size / TS_PACKET_SIZE;
186  if (n < 1)
187  n = 1;
188  s->max_payload_size = n * TS_PACKET_SIZE;
189  s->buf_ptr = s->buf;
190  break;
191  case AV_CODEC_ID_H264:
192  /* check for H.264 MP4 syntax */
193  if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
194  s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
195  }
196  break;
197  case AV_CODEC_ID_VORBIS:
198  case AV_CODEC_ID_THEORA:
199  if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
200  s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
201  s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
202  s->num_frames = 0;
203  goto defaultcase;
205  /* Due to a historical error, the clock rate for G722 in RTP is
206  * 8000, even if the sample rate is 16000. See RFC 3551. */
207  avpriv_set_pts_info(st, 32, 1, 8000);
208  break;
209  case AV_CODEC_ID_OPUS:
210  if (st->codec->channels > 2) {
211  av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
212  goto fail;
213  }
214  /* The opus RTP RFC says that all opus streams should use 48000 Hz
215  * as clock rate, since all opus sample rates can be expressed in
216  * this clock rate, and sample rate changes on the fly are supported. */
217  avpriv_set_pts_info(st, 32, 1, 48000);
218  break;
219  case AV_CODEC_ID_ILBC:
220  if (st->codec->block_align != 38 && st->codec->block_align != 50) {
221  av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
222  goto fail;
223  }
224  if (!s->max_frames_per_packet)
225  s->max_frames_per_packet = 1;
226  s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
227  s->max_payload_size / st->codec->block_align);
228  goto defaultcase;
229  case AV_CODEC_ID_AMR_NB:
230  case AV_CODEC_ID_AMR_WB:
231  if (!s->max_frames_per_packet)
232  s->max_frames_per_packet = 12;
233  if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
234  n = 31;
235  else
236  n = 61;
237  /* max_header_toc_size + the largest AMR payload must fit */
238  if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
239  av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
240  goto fail;
241  }
242  if (st->codec->channels != 1) {
243  av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
244  goto fail;
245  }
246  case AV_CODEC_ID_AAC:
247  s->num_frames = 0;
248  default:
249 defaultcase:
250  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
251  avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
252  }
253  s->buf_ptr = s->buf;
254  break;
255  }
256 
257  return 0;
258 
259 fail:
260  av_freep(&s->buf);
261  return AVERROR(EINVAL);
262 }
263 
264 /* send an rtcp sender report packet */
265 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
266 {
267  RTPMuxContext *s = s1->priv_data;
268  uint32_t rtp_ts;
269 
270  av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
271 
272  s->last_rtcp_ntp_time = ntp_time;
273  rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
274  s1->streams[0]->time_base) + s->base_timestamp;
275  avio_w8(s1->pb, (RTP_VERSION << 6));
276  avio_w8(s1->pb, RTCP_SR);
277  avio_wb16(s1->pb, 6); /* length in words - 1 */
278  avio_wb32(s1->pb, s->ssrc);
279  avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
280  avio_wb32(s1->pb, rtp_ts);
281  avio_wb32(s1->pb, s->packet_count);
282  avio_wb32(s1->pb, s->octet_count);
283 
284  if (s->cname) {
285  int len = FFMIN(strlen(s->cname), 255);
286  avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
287  avio_w8(s1->pb, RTCP_SDES);
288  avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
289 
290  avio_wb32(s1->pb, s->ssrc);
291  avio_w8(s1->pb, 0x01); /* CNAME */
292  avio_w8(s1->pb, len);
293  avio_write(s1->pb, s->cname, len);
294  avio_w8(s1->pb, 0); /* END */
295  for (len = (7 + len) % 4; len % 4; len++)
296  avio_w8(s1->pb, 0);
297  }
298 
299  avio_flush(s1->pb);
300 }
301 
302 /* send an rtp packet. sequence number is incremented, but the caller
303  must update the timestamp itself */
304 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
305 {
306  RTPMuxContext *s = s1->priv_data;
307 
308  av_dlog(s1, "rtp_send_data size=%d\n", len);
309 
310  /* build the RTP header */
311  avio_w8(s1->pb, (RTP_VERSION << 6));
312  avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
313  avio_wb16(s1->pb, s->seq);
314  avio_wb32(s1->pb, s->timestamp);
315  avio_wb32(s1->pb, s->ssrc);
316 
317  avio_write(s1->pb, buf1, len);
318  avio_flush(s1->pb);
319 
320  s->seq = (s->seq + 1) & 0xffff;
321  s->octet_count += len;
322  s->packet_count++;
323 }
324 
325 /* send an integer number of samples and compute time stamp and fill
326  the rtp send buffer before sending. */
328  const uint8_t *buf1, int size, int sample_size_bits)
329 {
330  RTPMuxContext *s = s1->priv_data;
331  int len, max_packet_size, n;
332  /* Calculate the number of bytes to get samples aligned on a byte border */
333  int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
334 
335  max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
336  /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
337  if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
338  return AVERROR(EINVAL);
339  n = 0;
340  while (size > 0) {
341  s->buf_ptr = s->buf;
342  len = FFMIN(max_packet_size, size);
343 
344  /* copy data */
345  memcpy(s->buf_ptr, buf1, len);
346  s->buf_ptr += len;
347  buf1 += len;
348  size -= len;
349  s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
350  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
351  n += (s->buf_ptr - s->buf);
352  }
353  return 0;
354 }
355 
357  const uint8_t *buf1, int size)
358 {
359  RTPMuxContext *s = s1->priv_data;
360  int len, count, max_packet_size;
361 
362  max_packet_size = s->max_payload_size;
363 
364  /* test if we must flush because not enough space */
365  len = (s->buf_ptr - s->buf);
366  if ((len + size) > max_packet_size) {
367  if (len > 4) {
368  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
369  s->buf_ptr = s->buf + 4;
370  }
371  }
372  if (s->buf_ptr == s->buf + 4) {
373  s->timestamp = s->cur_timestamp;
374  }
375 
376  /* add the packet */
377  if (size > max_packet_size) {
378  /* big packet: fragment */
379  count = 0;
380  while (size > 0) {
381  len = max_packet_size - 4;
382  if (len > size)
383  len = size;
384  /* build fragmented packet */
385  s->buf[0] = 0;
386  s->buf[1] = 0;
387  s->buf[2] = count >> 8;
388  s->buf[3] = count;
389  memcpy(s->buf + 4, buf1, len);
390  ff_rtp_send_data(s1, s->buf, len + 4, 0);
391  size -= len;
392  buf1 += len;
393  count += len;
394  }
395  } else {
396  if (s->buf_ptr == s->buf + 4) {
397  /* no fragmentation possible */
398  s->buf[0] = 0;
399  s->buf[1] = 0;
400  s->buf[2] = 0;
401  s->buf[3] = 0;
402  }
403  memcpy(s->buf_ptr, buf1, size);
404  s->buf_ptr += size;
405  }
406 }
407 
409  const uint8_t *buf1, int size)
410 {
411  RTPMuxContext *s = s1->priv_data;
412  int len, max_packet_size;
413 
414  max_packet_size = s->max_payload_size;
415 
416  while (size > 0) {
417  len = max_packet_size;
418  if (len > size)
419  len = size;
420 
421  s->timestamp = s->cur_timestamp;
422  ff_rtp_send_data(s1, buf1, len, (len == size));
423 
424  buf1 += len;
425  size -= len;
426  }
427 }
428 
429 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
431  const uint8_t *buf1, int size)
432 {
433  RTPMuxContext *s = s1->priv_data;
434  int len, out_len;
435 
436  while (size >= TS_PACKET_SIZE) {
437  len = s->max_payload_size - (s->buf_ptr - s->buf);
438  if (len > size)
439  len = size;
440  memcpy(s->buf_ptr, buf1, len);
441  buf1 += len;
442  size -= len;
443  s->buf_ptr += len;
444 
445  out_len = s->buf_ptr - s->buf;
446  if (out_len >= s->max_payload_size) {
447  ff_rtp_send_data(s1, s->buf, out_len, 0);
448  s->buf_ptr = s->buf;
449  }
450  }
451 }
452 
453 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
454 {
455  RTPMuxContext *s = s1->priv_data;
456  AVStream *st = s1->streams[0];
457  int frame_duration = av_get_audio_frame_duration(st->codec, 0);
458  int frame_size = st->codec->block_align;
459  int frames = size / frame_size;
460 
461  while (frames > 0) {
462  int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
463 
464  if (!s->num_frames) {
465  s->buf_ptr = s->buf;
466  s->timestamp = s->cur_timestamp;
467  }
468  memcpy(s->buf_ptr, buf, n * frame_size);
469  frames -= n;
470  s->num_frames += n;
471  s->buf_ptr += n * frame_size;
472  buf += n * frame_size;
473  s->cur_timestamp += n * frame_duration;
474 
475  if (s->num_frames == s->max_frames_per_packet) {
476  ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
477  s->num_frames = 0;
478  }
479  }
480  return 0;
481 }
482 
484 {
485  RTPMuxContext *s = s1->priv_data;
486  AVStream *st = s1->streams[0];
487  int rtcp_bytes;
488  int size= pkt->size;
489 
490  av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
491 
492  rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
494  if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
495  (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
496  !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
497  rtcp_send_sr(s1, ff_ntp_time());
499  s->first_packet = 0;
500  }
501  s->cur_timestamp = s->base_timestamp + pkt->pts;
502 
503  switch(st->codec->codec_id) {
506  case AV_CODEC_ID_PCM_U8:
507  case AV_CODEC_ID_PCM_S8:
508  return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
513  return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
515  /* The actual sample size is half a byte per sample, but since the
516  * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
517  * the correct parameter for send_samples_bits is 8 bits per stream
518  * clock. */
519  return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
521  return rtp_send_samples(s1, pkt->data, size,
523  case AV_CODEC_ID_MP2:
524  case AV_CODEC_ID_MP3:
525  rtp_send_mpegaudio(s1, pkt->data, size);
526  break;
529  ff_rtp_send_mpegvideo(s1, pkt->data, size);
530  break;
531  case AV_CODEC_ID_AAC:
532  if (s->flags & FF_RTP_FLAG_MP4A_LATM)
533  ff_rtp_send_latm(s1, pkt->data, size);
534  else
535  ff_rtp_send_aac(s1, pkt->data, size);
536  break;
537  case AV_CODEC_ID_AMR_NB:
538  case AV_CODEC_ID_AMR_WB:
539  ff_rtp_send_amr(s1, pkt->data, size);
540  break;
541  case AV_CODEC_ID_MPEG2TS:
542  rtp_send_mpegts_raw(s1, pkt->data, size);
543  break;
544  case AV_CODEC_ID_H264:
545  ff_rtp_send_h264(s1, pkt->data, size);
546  break;
547  case AV_CODEC_ID_H263:
548  if (s->flags & FF_RTP_FLAG_RFC2190) {
549  int mb_info_size = 0;
550  const uint8_t *mb_info =
552  &mb_info_size);
553  ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
554  break;
555  }
556  /* Fallthrough */
557  case AV_CODEC_ID_H263P:
558  ff_rtp_send_h263(s1, pkt->data, size);
559  break;
560  case AV_CODEC_ID_VORBIS:
561  case AV_CODEC_ID_THEORA:
562  ff_rtp_send_xiph(s1, pkt->data, size);
563  break;
564  case AV_CODEC_ID_VP8:
565  ff_rtp_send_vp8(s1, pkt->data, size);
566  break;
567  case AV_CODEC_ID_ILBC:
568  rtp_send_ilbc(s1, pkt->data, size);
569  break;
570  case AV_CODEC_ID_MJPEG:
571  ff_rtp_send_jpeg(s1, pkt->data, size);
572  break;
573  case AV_CODEC_ID_OPUS:
574  if (size > s->max_payload_size) {
575  av_log(s1, AV_LOG_ERROR,
576  "Packet size %d too large for max RTP payload size %d\n",
577  size, s->max_payload_size);
578  return AVERROR(EINVAL);
579  }
580  /* Intentional fallthrough */
581  default:
582  /* better than nothing : send the codec raw data */
583  rtp_send_raw(s1, pkt->data, size);
584  break;
585  }
586  return 0;
587 }
588 
590 {
591  RTPMuxContext *s = s1->priv_data;
592 
593  av_freep(&s->buf);
594 
595  return 0;
596 }
597 
599  .name = "rtp",
600  .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
601  .priv_data_size = sizeof(RTPMuxContext),
602  .audio_codec = AV_CODEC_ID_PCM_MULAW,
603  .video_codec = AV_CODEC_ID_MPEG4,
607  .priv_class = &rtp_muxer_class,
608 };
unsigned int packet_size
Definition: avformat.h:1042
const char * s
Definition: avisynth_c.h:668
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:1137
int size
AVOption.
Definition: opt.h:253
#define LIBAVUTIL_VERSION_INT
Definition: avcodec.h:820
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:3922
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:327
#define NTP_OFFSET_US
Definition: internal.h:91
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: rtpenc.c:483
#define RTP_VERSION
Definition: rtp.h:78
int64_t last_rtcp_ntp_time
Definition: rtpenc.h:42
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:686
int size
Definition: avcodec.h:1064
#define RTCP_TX_RATIO_NUM
Definition: rtp.h:82
unsigned int last_octet_count
Definition: rtpenc.h:46
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
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)
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
Definition: aviobuf.c:173
int void avio_flush(AVIOContext *s)
Force flushing of buffered data to the output s.
Definition: aviobuf.c:193
#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:958
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1910
#define FF_RTP_FLAG_MP4A_LATM
Definition: rtpenc.h:67
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avcodec.h:1265
void av_freep(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:234
Format I/O context.
Definition: avformat.h:968
#define RTP_PT_PRIVATE
Definition: rtp.h:77
const char * av_default_item_name(void *ctx)
Return the context name.
Definition: log.c:145
uint8_t
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:55
static const AVClass rtp_muxer_class
Definition: rtpenc.c:40
int id
Format-specific stream ID.
Definition: avformat.h:674
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
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
Definition: rtp.h:99
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:714
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:2563
const OptionDef options[]
Definition: ffserver.c:4682
#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:3665
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) av_const
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:130
static const uint8_t frame_size[4]
Definition: g723_1_data.h:58
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:102
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: avcodec.h:4147
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_aac.c:25
void * priv_data
Format private data.
Definition: avformat.h:988
unsigned m
Definition: audioconvert.c:186
void ff_rtp_send_jpeg(AVFormatContext *s1, const uint8_t *buff, int size)
Definition: rtpenc_jpeg.c:27
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:151
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:398
const char * avcodec_get_name(enum AVCodecID id)
Get the name of a codec.
Definition: utils.c:2626
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1234
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_const av_gcd(int64_t a, int64_t b)
Return the greatest common divisor of a and b.
Definition: mathematics.c:55
#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:430
goto fail
Definition: avfilter.c:963
unsigned int nb_streams
A list of all streams in the file.
Definition: avformat.h:1015
int64_t av_rescale_q_rnd(int64_t a, AVRational bq, AVRational cq, enum AVRounding) av_const
Rescale a 64-bit integer by 2 rational numbers with specified rounding.
Definition: mathematics.c:122
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:31
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
Definition: rtpenc.c:304
uint8_t * av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:323
AVStream ** streams
Definition: avformat.h:1016
static int write_trailer(AVFormatContext *s1)
Definition: v4l2enc.c:93
void * av_malloc(size_t size) av_malloc_attrib 1(1)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:73
#define FFMIN(a, b)
Definition: avcodec.h:925
const char * name
Definition: avformat.h:395
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
Definition: rtpenc.c:453
int n
Definition: avisynth_c.h:588
void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc_mpv.c:29
void avio_w8(AVIOContext *s, int b)
Definition: aviobuf.c:151
static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:356
preferred ID for MPEG-1/2 video decoding
Definition: avcodec.h:107
Stream structure.
Definition: avformat.h:667
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1893
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:598
enum AVMediaType codec_type
Definition: avcodec.h:1154
enum AVCodecID codec_id
Definition: avcodec.h:1157
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
Definition: avcodec.h:2290
int sample_rate
samples per second
Definition: avcodec.h:1873
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
Definition: rtpenc.h:72
static int is_supported(enum AVCodecID id)
Definition: rtpenc.c:49
void avio_wb64(AVIOContext *s, uint64_t val)
Definition: aviobuf.c:361
int first_packet
Definition: rtpenc.h:47
AVIOContext * pb
I/O context.
Definition: avformat.h:1001
void * buf
Definition: avisynth_c.h:594
Describe the class of an AVClass context structure.
Definition: log.h:50
rational number numerator/denominator
Definition: rational.h:43
uint8_t * data
Definition: avcodec.h:1063
int flags
Definition: rtpenc.h:60
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
Definition: rtpenc.c:265
void avio_wb32(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:299
#define s1
Definition: regdef.h:38
int num_frames
Definition: rtpenc.h:39
void avio_wb16(AVIOContext *s, unsigned int val)
Definition: aviobuf.c:373
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:2946
Round toward -infinity.
Definition: mathematics.h:70
uint8_t * buf_ptr
Definition: rtpenc.h:50
static int flags
Definition: cpu.c:45
#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
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Definition: rtpenc.c:408
#define RTCP_SR_SIZE
Definition: rtpenc.c:47
Main libavformat public API header.
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
Definition: opt.h:283
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:511
uint32_t timestamp
Definition: rtpenc.h:35
int len
int channels
number of audio channels
Definition: avcodec.h:1874
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:492
static int rtp_write_trailer(AVFormatContext *s1)
Definition: rtpenc.c:589
#define AVERROR(e)
void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size, const uint8_t *mb_info, int mb_info_size)
void INT64 INT64 count
Definition: avisynth_c.h:594
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
Definition: random_seed.c:106
static AVPacket pkt
Definition: demuxing.c:52
int stream_index
Definition: avcodec.h:1065
This structure stores compressed data.
Definition: avcodec.h:1040
static int write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: v4l2enc.c:85
int64_t pts
Presentation timestamp in AVStream-&gt;time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1056