ffmpeg/libavformat/rtpenc.c

   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[] = {
  32     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
  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:
  57     case AV_CODEC_ID_MPEG1VIDEO:
  58     case AV_CODEC_ID_MPEG2VIDEO:
  59     case AV_CODEC_ID_MPEG4:
  60     case AV_CODEC_ID_AAC:
  61     case AV_CODEC_ID_MP2:
  62     case AV_CODEC_ID_MP3:
  63     case AV_CODEC_ID_PCM_ALAW:
  64     case AV_CODEC_ID_PCM_MULAW:
  65     case AV_CODEC_ID_PCM_S8:
  66     case AV_CODEC_ID_PCM_S16BE:
  67     case AV_CODEC_ID_PCM_S16LE:
  68     case AV_CODEC_ID_PCM_U16BE:
  69     case AV_CODEC_ID_PCM_U16LE:
  70     case AV_CODEC_ID_PCM_U8:
  71     case AV_CODEC_ID_MPEG2TS:
  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:
  77     case AV_CODEC_ID_ADPCM_G722:
  78     case AV_CODEC_ID_ADPCM_G726:
  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
  89 static int rtp_write_header(AVFormatContext *s1)
  90 {
  91     RTPMuxContext *s = s1->priv_data;
  92     int n;
  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
 117     s->base_timestamp = av_get_random_seed();
 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;
 123     s->first_rtcp_ntp_time = ff_ntp_time();
 124     if (s1->start_time_realtime != 0  &&  s1->start_time_realtime != AV_NOPTS_VALUE)
 125         /* Round the NTP time to whole milliseconds. */
 126         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
 127                                  NTP_OFFSET_US;
 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     s->max_frames_per_packet = 0;
 156     if (s1->max_delay > 0) {
 157         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
 158             int frame_size = av_get_audio_frame_duration(st->codec, 0);
 159             if (!frame_size)
 160                 frame_size = st->codec->frame_size;
 161             if (frame_size == 0) {
 162                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
 163             } else {
 164                 s->max_frames_per_packet =
 165                         av_rescale_q_rnd(s1->max_delay,
 166                                          AV_TIME_BASE_Q,
 167                                          (AVRational){ frame_size, st->codec->sample_rate },
 168                                          AV_ROUND_DOWN);
 169             }
 170         }
 171         if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
 172             /* FIXME: We should round down here... */
 173             if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
 174                 s->max_frames_per_packet = av_rescale_q(s1->max_delay,
 175                                                         (AVRational){1, 1000000},
 176                                                         av_inv_q(st->avg_frame_rate));
 177             } else
 178                 s->max_frames_per_packet = 1;
 179         }
 180     }
 181
 182     avpriv_set_pts_info(st, 32, 1, 90000);
 183     switch(st->codec->codec_id) {
 184     case AV_CODEC_ID_MP2:
 185     case AV_CODEC_ID_MP3:
 186         s->buf_ptr = s->buf + 4;
 187         break;
 188     case AV_CODEC_ID_MPEG1VIDEO:
 189     case AV_CODEC_ID_MPEG2VIDEO:
 190         break;
 191     case AV_CODEC_ID_MPEG2TS:
 192         n = s->max_payload_size / TS_PACKET_SIZE;
 193         if (n < 1)
 194             n = 1;
 195         s->max_payload_size = n * TS_PACKET_SIZE;
 196         s->buf_ptr = s->buf;
 197         break;
 198     case AV_CODEC_ID_H264:
 199         /* check for H.264 MP4 syntax */
 200         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
 201             s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
 202         }
 203         break;
 204     case AV_CODEC_ID_HEVC:
 205         /* Only check for the standardized hvcC version of extradata, keeping
 206          * things simple and similar to the avcC/H264 case above, instead
 207          * of trying to handle the pre-standardization versions (as in
 208          * libavcodec/hevc.c). */
 209         if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
 210             s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
 211         }
 212         break;
 213     case AV_CODEC_ID_VORBIS:
 214     case AV_CODEC_ID_THEORA:
 215         if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
 216         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
 217         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
 218         s->num_frames = 0;
 219         goto defaultcase;
 220     case AV_CODEC_ID_ADPCM_G722:
 221         /* Due to a historical error, the clock rate for G722 in RTP is
 222          * 8000, even if the sample rate is 16000. See RFC 3551. */
 223         avpriv_set_pts_info(st, 32, 1, 8000);
 224         break;
 225     case AV_CODEC_ID_OPUS:
 226         if (st->codec->channels > 2) {
 227             av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
 228             goto fail;
 229         }
 230         /* The opus RTP RFC says that all opus streams should use 48000 Hz
 231          * as clock rate, since all opus sample rates can be expressed in
 232          * this clock rate, and sample rate changes on the fly are supported. */
 233         avpriv_set_pts_info(st, 32, 1, 48000);
 234         break;
 235     case AV_CODEC_ID_ILBC:
 236         if (st->codec->block_align != 38 && st->codec->block_align != 50) {
 237             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
 238             goto fail;
 239         }
 240         if (!s->max_frames_per_packet)
 241             s->max_frames_per_packet = 1;
 242         s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
 243                                          s->max_payload_size / st->codec->block_align);
 244         goto defaultcase;
 245     case AV_CODEC_ID_AMR_NB:
 246     case AV_CODEC_ID_AMR_WB:
 247         if (!s->max_frames_per_packet)
 248             s->max_frames_per_packet = 12;
 249         if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
 250             n = 31;
 251         else
 252             n = 61;
 253         /* max_header_toc_size + the largest AMR payload must fit */
 254         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
 255             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
 256             goto fail;
 257         }
 258         if (st->codec->channels != 1) {
 259             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
 260             goto fail;
 261         }
 262     case AV_CODEC_ID_AAC:
 263         s->num_frames = 0;
 264     default:
 265 defaultcase:
 266         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
 267             avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
 268         }
 269         s->buf_ptr = s->buf;
 270         break;
 271     }
 272
 273     return 0;
 274
 275 fail:
 276     av_freep(&s->buf);
 277     return AVERROR(EINVAL);
 278 }
 279
 280 /* send an rtcp sender report packet */
 281 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
 282 {
 283     RTPMuxContext *s = s1->priv_data;
 284     uint32_t rtp_ts;
 285
 286     av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
 287
 288     s->last_rtcp_ntp_time = ntp_time;
 289     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
 290                           s1->streams[0]->time_base) + s->base_timestamp;
 291     avio_w8(s1->pb, RTP_VERSION << 6);
 292     avio_w8(s1->pb, RTCP_SR);
 293     avio_wb16(s1->pb, 6); /* length in words - 1 */
 294     avio_wb32(s1->pb, s->ssrc);
 295     avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
 296     avio_wb32(s1->pb, rtp_ts);
 297     avio_wb32(s1->pb, s->packet_count);
 298     avio_wb32(s1->pb, s->octet_count);
 299
 300     if (s->cname) {
 301         int len = FFMIN(strlen(s->cname), 255);
 302         avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
 303         avio_w8(s1->pb, RTCP_SDES);
 304         avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
 305
 306         avio_wb32(s1->pb, s->ssrc);
 307         avio_w8(s1->pb, 0x01); /* CNAME */
 308         avio_w8(s1->pb, len);
 309         avio_write(s1->pb, s->cname, len);
 310         avio_w8(s1->pb, 0); /* END */
 311         for (len = (7 + len) % 4; len % 4; len++)
 312             avio_w8(s1->pb, 0);
 313     }
 314
 315     if (bye) {
 316         avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
 317         avio_w8(s1->pb, RTCP_BYE);
 318         avio_wb16(s1->pb, 1); /* length in words - 1 */
 319         avio_wb32(s1->pb, s->ssrc);
 320     }
 321
 322     avio_flush(s1->pb);
 323 }
 324
 325 /* send an rtp packet. sequence number is incremented, but the caller
 326    must update the timestamp itself */
 327 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 328 {
 329     RTPMuxContext *s = s1->priv_data;
 330
 331     av_dlog(s1, "rtp_send_data size=%d\n", len);
 332
 333     /* build the RTP header */
 334     avio_w8(s1->pb, RTP_VERSION << 6);
 335     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
 336     avio_wb16(s1->pb, s->seq);
 337     avio_wb32(s1->pb, s->timestamp);
 338     avio_wb32(s1->pb, s->ssrc);
 339
 340     avio_write(s1->pb, buf1, len);
 341     avio_flush(s1->pb);
 342
 343     s->seq = (s->seq + 1) & 0xffff;
 344     s->octet_count += len;
 345     s->packet_count++;
 346 }
 347
 348 /* send an integer number of samples and compute time stamp and fill
 349    the rtp send buffer before sending. */
 350 static int rtp_send_samples(AVFormatContext *s1,
 351                             const uint8_t *buf1, int size, int sample_size_bits)
 352 {
 353     RTPMuxContext *s = s1->priv_data;
 354     int len, max_packet_size, n;
 355     /* Calculate the number of bytes to get samples aligned on a byte border */
 356     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
 357
 358     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
 359     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
 360     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
 361         return AVERROR(EINVAL);
 362     n = 0;
 363     while (size > 0) {
 364         s->buf_ptr = s->buf;
 365         len = FFMIN(max_packet_size, size);
 366
 367         /* copy data */
 368         memcpy(s->buf_ptr, buf1, len);
 369         s->buf_ptr += len;
 370         buf1 += len;
 371         size -= len;
 372         s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
 373         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 374         n += (s->buf_ptr - s->buf);
 375     }
 376     return 0;
 377 }
 378
 379 static void rtp_send_mpegaudio(AVFormatContext *s1,
 380                                const uint8_t *buf1, int size)
 381 {
 382     RTPMuxContext *s = s1->priv_data;
 383     int len, count, max_packet_size;
 384
 385     max_packet_size = s->max_payload_size;
 386
 387     /* test if we must flush because not enough space */
 388     len = (s->buf_ptr - s->buf);
 389     if ((len + size) > max_packet_size) {
 390         if (len > 4) {
 391             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 392             s->buf_ptr = s->buf + 4;
 393         }
 394     }
 395     if (s->buf_ptr == s->buf + 4) {
 396         s->timestamp = s->cur_timestamp;
 397     }
 398
 399     /* add the packet */
 400     if (size > max_packet_size) {
 401         /* big packet: fragment */
 402         count = 0;
 403         while (size > 0) {
 404             len = max_packet_size - 4;
 405             if (len > size)
 406                 len = size;
 407             /* build fragmented packet */
 408             s->buf[0] = 0;
 409             s->buf[1] = 0;
 410             s->buf[2] = count >> 8;
 411             s->buf[3] = count;
 412             memcpy(s->buf + 4, buf1, len);
 413             ff_rtp_send_data(s1, s->buf, len + 4, 0);
 414             size -= len;
 415             buf1 += len;
 416             count += len;
 417         }
 418     } else {
 419         if (s->buf_ptr == s->buf + 4) {
 420             /* no fragmentation possible */
 421             s->buf[0] = 0;
 422             s->buf[1] = 0;
 423             s->buf[2] = 0;
 424             s->buf[3] = 0;
 425         }
 426         memcpy(s->buf_ptr, buf1, size);
 427         s->buf_ptr += size;
 428     }
 429 }
 430
 431 static void rtp_send_raw(AVFormatContext *s1,
 432                          const uint8_t *buf1, int size)
 433 {
 434     RTPMuxContext *s = s1->priv_data;
 435     int len, max_packet_size;
 436
 437     max_packet_size = s->max_payload_size;
 438
 439     while (size > 0) {
 440         len = max_packet_size;
 441         if (len > size)
 442             len = size;
 443
 444         s->timestamp = s->cur_timestamp;
 445         ff_rtp_send_data(s1, buf1, len, (len == size));
 446
 447         buf1 += len;
 448         size -= len;
 449     }
 450 }
 451
 452 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
 453 static void rtp_send_mpegts_raw(AVFormatContext *s1,
 454                                 const uint8_t *buf1, int size)
 455 {
 456     RTPMuxContext *s = s1->priv_data;
 457     int len, out_len;
 458
 459     while (size >= TS_PACKET_SIZE) {
 460         len = s->max_payload_size - (s->buf_ptr - s->buf);
 461         if (len > size)
 462             len = size;
 463         memcpy(s->buf_ptr, buf1, len);
 464         buf1 += len;
 465         size -= len;
 466         s->buf_ptr += len;
 467
 468         out_len = s->buf_ptr - s->buf;
 469         if (out_len >= s->max_payload_size) {
 470             ff_rtp_send_data(s1, s->buf, out_len, 0);
 471             s->buf_ptr = s->buf;
 472         }
 473     }
 474 }
 475
 476 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
 477 {
 478     RTPMuxContext *s = s1->priv_data;
 479     AVStream *st = s1->streams[0];
 480     int frame_duration = av_get_audio_frame_duration(st->codec, 0);
 481     int frame_size = st->codec->block_align;
 482     int frames = size / frame_size;
 483
 484     while (frames > 0) {
 485         int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
 486
 487         if (!s->num_frames) {
 488             s->buf_ptr = s->buf;
 489             s->timestamp = s->cur_timestamp;
 490         }
 491         memcpy(s->buf_ptr, buf, n * frame_size);
 492         frames           -= n;
 493         s->num_frames    += n;
 494         s->buf_ptr       += n * frame_size;
 495         buf              += n * frame_size;
 496         s->cur_timestamp += n * frame_duration;
 497
 498         if (s->num_frames == s->max_frames_per_packet) {
 499             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
 500             s->num_frames = 0;
 501         }
 502     }
 503     return 0;
 504 }
 505
 506 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 507 {
 508     RTPMuxContext *s = s1->priv_data;
 509     AVStream *st = s1->streams[0];
 510     int rtcp_bytes;
 511     int size= pkt->size;
 512
 513     av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
 514
 515     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 516         RTCP_TX_RATIO_DEN;
 517     if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
 518                             (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
 519         !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
 520         rtcp_send_sr(s1, ff_ntp_time(), 0);
 521         s->last_octet_count = s->octet_count;
 522         s->first_packet = 0;
 523     }
 524     s->cur_timestamp = s->base_timestamp + pkt->pts;
 525
 526     switch(st->codec->codec_id) {
 527     case AV_CODEC_ID_PCM_MULAW:
 528     case AV_CODEC_ID_PCM_ALAW:
 529     case AV_CODEC_ID_PCM_U8:
 530     case AV_CODEC_ID_PCM_S8:
 531         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
 532     case AV_CODEC_ID_PCM_U16BE:
 533     case AV_CODEC_ID_PCM_U16LE:
 534     case AV_CODEC_ID_PCM_S16BE:
 535     case AV_CODEC_ID_PCM_S16LE:
 536         return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
 537     case AV_CODEC_ID_ADPCM_G722:
 538         /* The actual sample size is half a byte per sample, but since the
 539          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
 540          * the correct parameter for send_samples_bits is 8 bits per stream
 541          * clock. */
 542         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
 543     case AV_CODEC_ID_ADPCM_G726:
 544         return rtp_send_samples(s1, pkt->data, size,
 545                                 st->codec->bits_per_coded_sample * st->codec->channels);
 546     case AV_CODEC_ID_MP2:
 547     case AV_CODEC_ID_MP3:
 548         rtp_send_mpegaudio(s1, pkt->data, size);
 549         break;
 550     case AV_CODEC_ID_MPEG1VIDEO:
 551     case AV_CODEC_ID_MPEG2VIDEO:
 552         ff_rtp_send_mpegvideo(s1, pkt->data, size);
 553         break;
 554     case AV_CODEC_ID_AAC:
 555         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
 556             ff_rtp_send_latm(s1, pkt->data, size);
 557         else
 558             ff_rtp_send_aac(s1, pkt->data, size);
 559         break;
 560     case AV_CODEC_ID_AMR_NB:
 561     case AV_CODEC_ID_AMR_WB:
 562         ff_rtp_send_amr(s1, pkt->data, size);
 563         break;
 564     case AV_CODEC_ID_MPEG2TS:
 565         rtp_send_mpegts_raw(s1, pkt->data, size);
 566         break;
 567     case AV_CODEC_ID_H264:
 568         ff_rtp_send_h264(s1, pkt->data, size);
 569         break;
 570     case AV_CODEC_ID_H261:
 571         ff_rtp_send_h261(s1, pkt->data, size);
 572         break;
 573     case AV_CODEC_ID_H263:
 574         if (s->flags & FF_RTP_FLAG_RFC2190) {
 575             int mb_info_size = 0;
 576             const uint8_t *mb_info =
 577                 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
 578                                         &mb_info_size);
 579             if (!mb_info) {
 580                 av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n");
 581                 return AVERROR(ENOMEM);
 582             }
 583             ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
 584             break;
 585         }
 586         /* Fallthrough */
 587     case AV_CODEC_ID_H263P:
 588         ff_rtp_send_h263(s1, pkt->data, size);
 589         break;
 590     case AV_CODEC_ID_HEVC:
 591         ff_rtp_send_hevc(s1, pkt->data, size);
 592         break;
 593     case AV_CODEC_ID_VORBIS:
 594     case AV_CODEC_ID_THEORA:
 595         ff_rtp_send_xiph(s1, pkt->data, size);
 596         break;
 597     case AV_CODEC_ID_VP8:
 598         ff_rtp_send_vp8(s1, pkt->data, size);
 599         break;
 600     case AV_CODEC_ID_ILBC:
 601         rtp_send_ilbc(s1, pkt->data, size);
 602         break;
 603     case AV_CODEC_ID_MJPEG:
 604         ff_rtp_send_jpeg(s1, pkt->data, size);
 605         break;
 606     case AV_CODEC_ID_OPUS:
 607         if (size > s->max_payload_size) {
 608             av_log(s1, AV_LOG_ERROR,
 609                    "Packet size %d too large for max RTP payload size %d\n",
 610                    size, s->max_payload_size);
 611             return AVERROR(EINVAL);
 612         }
 613         /* Intentional fallthrough */
 614     default:
 615         /* better than nothing : send the codec raw data */
 616         rtp_send_raw(s1, pkt->data, size);
 617         break;
 618     }
 619     return 0;
 620 }
 621
 622 static int rtp_write_trailer(AVFormatContext *s1)
 623 {
 624     RTPMuxContext *s = s1->priv_data;
 625
 626     /* If the caller closes and recreates ->pb, this might actually
 627      * be NULL here even if it was successfully allocated at the start. */
 628     if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
 629         rtcp_send_sr(s1, ff_ntp_time(), 1);
 630     av_freep(&s->buf);
 631
 632     return 0;
 633 }
 634
 635 AVOutputFormat ff_rtp_muxer = {
 636     .name              = "rtp",
 637     .long_name         = NULL_IF_CONFIG_SMALL("RTP output"),
 638     .priv_data_size    = sizeof(RTPMuxContext),
 639     .audio_codec       = AV_CODEC_ID_PCM_MULAW,
 640     .video_codec       = AV_CODEC_ID_MPEG4,
 641     .write_header      = rtp_write_header,
 642     .write_packet      = rtp_write_packet,
 643     .write_trailer     = rtp_write_trailer,
 644     .priv_class        = &rtp_muxer_class,
 645 };