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 "mux.h"
  26 #include "libavutil/mathematics.h"
  27 #include "libavutil/mem.h"
  28 #include "libavutil/random_seed.h"
  29 #include "libavutil/opt.h"
  30
  31 #include "rtpenc.h"
  32
  33 static const AVOption options[] = {
  34     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
  35     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
  36     { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
  37     { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
  38     { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
  39     { NULL },
  40 };
  41
  42 static const AVClass rtp_muxer_class = {
  43     .class_name = "RTP muxer",
  44     .item_name  = av_default_item_name,
  45     .option     = options,
  46     .version    = LIBAVUTIL_VERSION_INT,
  47 };
  48
  49 #define RTCP_SR_SIZE 28
  50
  51 static int is_supported(enum AVCodecID id)
  52 {
  53     switch(id) {
  54     case AV_CODEC_ID_DIRAC:
  55     case AV_CODEC_ID_H261:
  56     case AV_CODEC_ID_H263:
  57     case AV_CODEC_ID_H263P:
  58     case AV_CODEC_ID_H264:
  59     case AV_CODEC_ID_HEVC:
  60     case AV_CODEC_ID_MPEG1VIDEO:
  61     case AV_CODEC_ID_MPEG2VIDEO:
  62     case AV_CODEC_ID_MPEG4:
  63     case AV_CODEC_ID_AAC:
  64     case AV_CODEC_ID_MP2:
  65     case AV_CODEC_ID_MP3:
  66     case AV_CODEC_ID_PCM_ALAW:
  67     case AV_CODEC_ID_PCM_MULAW:
  68     case AV_CODEC_ID_PCM_S8:
  69     case AV_CODEC_ID_PCM_S16BE:
  70     case AV_CODEC_ID_PCM_S16LE:
  71     case AV_CODEC_ID_PCM_S24BE:
  72     case AV_CODEC_ID_PCM_U16BE:
  73     case AV_CODEC_ID_PCM_U16LE:
  74     case AV_CODEC_ID_PCM_U8:
  75     case AV_CODEC_ID_MPEG2TS:
  76     case AV_CODEC_ID_AMR_NB:
  77     case AV_CODEC_ID_AMR_WB:
  78     case AV_CODEC_ID_VORBIS:
  79     case AV_CODEC_ID_THEORA:
  80     case AV_CODEC_ID_VP8:
  81     case AV_CODEC_ID_VP9:
  82     case AV_CODEC_ID_AV1:
  83     case AV_CODEC_ID_ADPCM_G722:
  84     case AV_CODEC_ID_ADPCM_G726:
  85     case AV_CODEC_ID_ADPCM_G726LE:
  86     case AV_CODEC_ID_ILBC:
  87     case AV_CODEC_ID_MJPEG:
  88     case AV_CODEC_ID_SPEEX:
  89     case AV_CODEC_ID_OPUS:
  90     case AV_CODEC_ID_RAWVIDEO:
  91     case AV_CODEC_ID_BITPACKED:
  92     case AV_CODEC_ID_G728:
  93         return 1;
  94     default:
  95         return 0;
  96     }
  97 }
  98
  99 static int rtp_write_header(AVFormatContext *s1)
 100 {
 101     RTPMuxContext *s = s1->priv_data;
 102     int n, ret = AVERROR(EINVAL);
 103     AVStream *st;
 104
 105     if (s1->nb_streams != 1) {
 106         av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
 107         return AVERROR(EINVAL);
 108     }
 109     st = s1->streams[0];
 110     if (!is_supported(st->codecpar->codec_id)) {
 111         av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
 112
 113         return -1;
 114     }
 115
 116     if (s->payload_type < 0) {
 117         /* Re-validate non-dynamic payload types */
 118         if (st->id < RTP_PT_PRIVATE)
 119             st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
 120
 121         s->payload_type = st->id;
 122     } else {
 123         /* private option takes priority */
 124         st->id = s->payload_type;
 125     }
 126
 127     s->base_timestamp = av_get_random_seed();
 128     s->timestamp = s->base_timestamp;
 129     s->cur_timestamp = 0;
 130     if (!s->ssrc)
 131         s->ssrc = av_get_random_seed();
 132     s->first_packet = 1;
 133     s->first_rtcp_ntp_time = ff_ntp_time();
 134     if (s1->start_time_realtime != 0  &&  s1->start_time_realtime != AV_NOPTS_VALUE)
 135         /* Round the NTP time to whole milliseconds. */
 136         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
 137                                  NTP_OFFSET_US;
 138     // Pick a random sequence start number, but in the lower end of the
 139     // available range, so that any wraparound doesn't happen immediately.
 140     // (Immediate wraparound would be an issue for SRTP.)
 141     if (s->seq < 0) {
 142         if (s1->flags & AVFMT_FLAG_BITEXACT) {
 143             s->seq = 0;
 144         } else
 145             s->seq = av_get_random_seed() & 0x0fff;
 146     } else
 147         s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
 148
 149     if (s1->packet_size) {
 150         if (s1->pb->max_packet_size)
 151             s1->packet_size = FFMIN(s1->packet_size,
 152                                     s1->pb->max_packet_size);
 153     } else
 154         s1->packet_size = s1->pb->max_packet_size;
 155     if (s1->packet_size <= 12) {
 156         av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
 157         return AVERROR(EIO);
 158     }
 159     s->buf = av_malloc(s1->packet_size);
 160     if (!s->buf) {
 161         return AVERROR(ENOMEM);
 162     }
 163     s->max_payload_size = s1->packet_size - 12;
 164
 165     if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
 166         avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
 167     } else {
 168         avpriv_set_pts_info(st, 32, 1, 90000);
 169     }
 170     s->buf_ptr = s->buf;
 171     switch(st->codecpar->codec_id) {
 172     case AV_CODEC_ID_MP2:
 173     case AV_CODEC_ID_MP3:
 174         s->buf_ptr = s->buf + 4;
 175         avpriv_set_pts_info(st, 32, 1, 90000);
 176         break;
 177     case AV_CODEC_ID_MPEG1VIDEO:
 178     case AV_CODEC_ID_MPEG2VIDEO:
 179         break;
 180     case AV_CODEC_ID_MPEG2TS:
 181         n = s->max_payload_size / TS_PACKET_SIZE;
 182         if (n < 1)
 183             n = 1;
 184         s->max_payload_size = n * TS_PACKET_SIZE;
 185         break;
 186     case AV_CODEC_ID_DIRAC:
 187         if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
 188             av_log(s, AV_LOG_ERROR,
 189                    "Packetizing VC-2 is experimental and does not use all values "
 190                    "of the specification "
 191                    "(even though most receivers may handle it just fine). "
 192                    "Please set -strict experimental in order to enable it.\n");
 193             ret = AVERROR_EXPERIMENTAL;
 194             goto fail;
 195         }
 196         break;
 197     case AV_CODEC_ID_H261:
 198         if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
 199             av_log(s, AV_LOG_ERROR,
 200                    "Packetizing H.261 is experimental and produces incorrect "
 201                    "packetization for cases where GOBs don't fit into packets "
 202                    "(even though most receivers may handle it just fine). "
 203                    "Please set -f_strict experimental in order to enable it.\n");
 204             ret = AVERROR_EXPERIMENTAL;
 205             goto fail;
 206         }
 207         break;
 208     case AV_CODEC_ID_H264:
 209         /* check for H.264 MP4 syntax */
 210         if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
 211             s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
 212         }
 213         break;
 214     case AV_CODEC_ID_HEVC:
 215         /* Only check for the standardized hvcC version of extradata, keeping
 216          * things simple and similar to the avcC/H.264 case above, instead
 217          * of trying to handle the pre-standardization versions (as in
 218          * libavcodec/hevc.c). */
 219         if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
 220             s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
 221         }
 222         break;
 223     case AV_CODEC_ID_MJPEG:
 224     case AV_CODEC_ID_BITPACKED:
 225     case AV_CODEC_ID_RAWVIDEO:
 226         if (st->codecpar->width <= 0 || st->codecpar->height <= 0) {
 227             av_log(s1, AV_LOG_ERROR, "dimensions not set\n");
 228             return AVERROR(EINVAL);
 229         }
 230         break;
 231     case AV_CODEC_ID_VP9:
 232         if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
 233             av_log(s, AV_LOG_ERROR,
 234                    "Packetizing VP9 is experimental and its specification is "
 235                    "still in draft state. "
 236                    "Please set -strict experimental in order to enable it.\n");
 237             ret = AVERROR_EXPERIMENTAL;
 238             goto fail;
 239         }
 240         break;
 241     case AV_CODEC_ID_AV1:
 242         if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
 243             av_log(s, AV_LOG_ERROR,
 244                    "Packetizing AV1 is experimental and its specification is "
 245                    "still in draft state. "
 246                    "Please set -strict experimental in order to enable it.\n");
 247             ret = AVERROR_EXPERIMENTAL;
 248             goto fail;
 249         }
 250         break;
 251     case AV_CODEC_ID_VORBIS:
 252     case AV_CODEC_ID_THEORA:
 253         s->max_frames_per_packet = 15;
 254         break;
 255     case AV_CODEC_ID_ADPCM_G722:
 256         /* Due to a historical error, the clock rate for G722 in RTP is
 257          * 8000, even if the sample rate is 16000. See RFC 3551. */
 258         avpriv_set_pts_info(st, 32, 1, 8000);
 259         break;
 260     case AV_CODEC_ID_OPUS:
 261         if (st->codecpar->ch_layout.nb_channels > 2) {
 262             av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
 263             goto fail;
 264         }
 265         /* The opus RTP RFC says that all opus streams should use 48000 Hz
 266          * as clock rate, since all opus sample rates can be expressed in
 267          * this clock rate, and sample rate changes on the fly are supported. */
 268         avpriv_set_pts_info(st, 32, 1, 48000);
 269         break;
 270     case AV_CODEC_ID_ILBC:
 271         if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
 272             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
 273             goto fail;
 274         }
 275         s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
 276         break;
 277     case AV_CODEC_ID_AMR_NB:
 278     case AV_CODEC_ID_AMR_WB:
 279         s->max_frames_per_packet = 50;
 280         if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
 281             n = 31;
 282         else
 283             n = 61;
 284         /* max_header_toc_size + the largest AMR payload must fit */
 285         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
 286             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
 287             goto fail;
 288         }
 289         if (st->codecpar->ch_layout.nb_channels != 1) {
 290             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
 291             goto fail;
 292         }
 293         break;
 294     case AV_CODEC_ID_AAC:
 295         s->max_frames_per_packet = 50;
 296         break;
 297     default:
 298         break;
 299     }
 300
 301     return 0;
 302
 303 fail:
 304     av_freep(&s->buf);
 305     return ret;
 306 }
 307
 308 /* send an rtcp sender report packet */
 309 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
 310 {
 311     RTPMuxContext *s = s1->priv_data;
 312     uint32_t rtp_ts;
 313
 314     av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
 315
 316     s->last_rtcp_ntp_time = ntp_time;
 317     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
 318                           s1->streams[0]->time_base) + s->base_timestamp;
 319     avio_w8(s1->pb, RTP_VERSION << 6);
 320     avio_w8(s1->pb, RTCP_SR);
 321     avio_wb16(s1->pb, 6); /* length in words - 1 */
 322     avio_wb32(s1->pb, s->ssrc);
 323     avio_wb32(s1->pb, ntp_time / 1000000);
 324     avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
 325     avio_wb32(s1->pb, rtp_ts);
 326     avio_wb32(s1->pb, s->packet_count);
 327     avio_wb32(s1->pb, s->octet_count);
 328
 329     if (s->cname) {
 330         int len = FFMIN(strlen(s->cname), 255);
 331         avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
 332         avio_w8(s1->pb, RTCP_SDES);
 333         avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
 334
 335         avio_wb32(s1->pb, s->ssrc);
 336         avio_w8(s1->pb, 0x01); /* CNAME */
 337         avio_w8(s1->pb, len);
 338         avio_write(s1->pb, s->cname, len);
 339         avio_w8(s1->pb, 0); /* END */
 340         for (len = (7 + len) % 4; len % 4; len++)
 341             avio_w8(s1->pb, 0);
 342     }
 343
 344     if (bye) {
 345         avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
 346         avio_w8(s1->pb, RTCP_BYE);
 347         avio_wb16(s1->pb, 1); /* length in words - 1 */
 348         avio_wb32(s1->pb, s->ssrc);
 349     }
 350
 351     avio_flush(s1->pb);
 352 }
 353
 354 /* send an rtp packet. sequence number is incremented, but the caller
 355    must update the timestamp itself */
 356 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 357 {
 358     RTPMuxContext *s = s1->priv_data;
 359
 360     av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
 361
 362     /* build the RTP header */
 363     avio_w8(s1->pb, RTP_VERSION << 6);
 364     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
 365     avio_wb16(s1->pb, s->seq);
 366     avio_wb32(s1->pb, s->timestamp);
 367     avio_wb32(s1->pb, s->ssrc);
 368
 369     avio_write(s1->pb, buf1, len);
 370     avio_flush(s1->pb);
 371
 372     s->seq = (s->seq + 1) & 0xffff;
 373     s->octet_count += len;
 374     s->packet_count++;
 375 }
 376
 377 /* send an integer number of samples and compute time stamp and fill
 378    the rtp send buffer before sending. */
 379 static int rtp_send_samples(AVFormatContext *s1,
 380                             const uint8_t *buf1, int size, int sample_size_bits)
 381 {
 382     RTPMuxContext *s = s1->priv_data;
 383     int len, max_packet_size, n;
 384     /* Calculate the number of bytes to get samples aligned on a byte border */
 385     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
 386
 387     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
 388     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
 389     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
 390         return AVERROR(EINVAL);
 391     n = 0;
 392     while (size > 0) {
 393         s->buf_ptr = s->buf;
 394         len = FFMIN(max_packet_size, size);
 395
 396         /* copy data */
 397         memcpy(s->buf_ptr, buf1, len);
 398         s->buf_ptr += len;
 399         buf1 += len;
 400         size -= len;
 401         s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
 402         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 403         n += (s->buf_ptr - s->buf);
 404     }
 405     return 0;
 406 }
 407
 408 static void rtp_send_mpegaudio(AVFormatContext *s1,
 409                                const uint8_t *buf1, int size)
 410 {
 411     RTPMuxContext *s = s1->priv_data;
 412     int len, count, max_packet_size;
 413
 414     max_packet_size = s->max_payload_size;
 415
 416     /* test if we must flush because not enough space */
 417     len = (s->buf_ptr - s->buf);
 418     if ((len + size) > max_packet_size) {
 419         if (len > 4) {
 420             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 421             s->buf_ptr = s->buf + 4;
 422         }
 423     }
 424     if (s->buf_ptr == s->buf + 4) {
 425         s->timestamp = s->cur_timestamp;
 426     }
 427
 428     /* add the packet */
 429     if (size > max_packet_size) {
 430         /* big packet: fragment */
 431         count = 0;
 432         while (size > 0) {
 433             len = max_packet_size - 4;
 434             if (len > size)
 435                 len = size;
 436             /* build fragmented packet */
 437             s->buf[0] = 0;
 438             s->buf[1] = 0;
 439             s->buf[2] = count >> 8;
 440             s->buf[3] = count;
 441             memcpy(s->buf + 4, buf1, len);
 442             ff_rtp_send_data(s1, s->buf, len + 4, 0);
 443             size -= len;
 444             buf1 += len;
 445             count += len;
 446         }
 447     } else {
 448         if (s->buf_ptr == s->buf + 4) {
 449             /* no fragmentation possible */
 450             s->buf[0] = 0;
 451             s->buf[1] = 0;
 452             s->buf[2] = 0;
 453             s->buf[3] = 0;
 454         }
 455         memcpy(s->buf_ptr, buf1, size);
 456         s->buf_ptr += size;
 457     }
 458 }
 459
 460 static void rtp_send_raw(AVFormatContext *s1,
 461                          const uint8_t *buf1, int size)
 462 {
 463     RTPMuxContext *s = s1->priv_data;
 464     int len, max_packet_size;
 465
 466     max_packet_size = s->max_payload_size;
 467
 468     while (size > 0) {
 469         len = max_packet_size;
 470         if (len > size)
 471             len = size;
 472
 473         s->timestamp = s->cur_timestamp;
 474         ff_rtp_send_data(s1, buf1, len, (len == size));
 475
 476         buf1 += len;
 477         size -= len;
 478     }
 479 }
 480
 481 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
 482 static void rtp_send_mpegts_raw(AVFormatContext *s1,
 483                                 const uint8_t *buf1, int size)
 484 {
 485     RTPMuxContext *s = s1->priv_data;
 486     int len, out_len;
 487
 488     s->timestamp = s->cur_timestamp;
 489     while (size >= TS_PACKET_SIZE) {
 490         len = s->max_payload_size - (s->buf_ptr - s->buf);
 491         if (len > size)
 492             len = size;
 493         memcpy(s->buf_ptr, buf1, len);
 494         buf1 += len;
 495         size -= len;
 496         s->buf_ptr += len;
 497
 498         out_len = s->buf_ptr - s->buf;
 499         if (out_len >= s->max_payload_size) {
 500             ff_rtp_send_data(s1, s->buf, out_len, 0);
 501             s->buf_ptr = s->buf;
 502         }
 503     }
 504 }
 505
 506 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
 507 {
 508     RTPMuxContext *s = s1->priv_data;
 509     AVStream *st = s1->streams[0];
 510     int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
 511     int frame_size = st->codecpar->block_align;
 512     int frames = size / frame_size;
 513
 514     while (frames > 0) {
 515         if (s->num_frames > 0 &&
 516             av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
 517                           s1->max_delay, AV_TIME_BASE_Q) >= 0) {
 518             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
 519             s->num_frames = 0;
 520         }
 521
 522         if (!s->num_frames) {
 523             s->buf_ptr = s->buf;
 524             s->timestamp = s->cur_timestamp;
 525         }
 526         memcpy(s->buf_ptr, buf, frame_size);
 527         frames--;
 528         s->num_frames++;
 529         s->buf_ptr       += frame_size;
 530         buf              += frame_size;
 531         s->cur_timestamp += frame_duration;
 532
 533         if (s->num_frames == s->max_frames_per_packet) {
 534             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
 535             s->num_frames = 0;
 536         }
 537     }
 538     return 0;
 539 }
 540
 541 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 542 {
 543     RTPMuxContext *s = s1->priv_data;
 544     AVStream *st = s1->streams[0];
 545     int rtcp_bytes;
 546     int size= pkt->size;
 547
 548     av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
 549
 550     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 551         RTCP_TX_RATIO_DEN;
 552     if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
 553                             (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
 554         !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
 555         rtcp_send_sr(s1, ff_ntp_time(), 0);
 556         s->last_octet_count = s->octet_count;
 557         s->first_packet = 0;
 558     }
 559     s->cur_timestamp = s->base_timestamp + pkt->pts;
 560
 561     switch(st->codecpar->codec_id) {
 562     case AV_CODEC_ID_PCM_MULAW:
 563     case AV_CODEC_ID_PCM_ALAW:
 564     case AV_CODEC_ID_PCM_U8:
 565     case AV_CODEC_ID_PCM_S8:
 566         return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels);
 567     case AV_CODEC_ID_PCM_U16BE:
 568     case AV_CODEC_ID_PCM_U16LE:
 569     case AV_CODEC_ID_PCM_S16BE:
 570     case AV_CODEC_ID_PCM_S16LE:
 571         return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->ch_layout.nb_channels);
 572     case AV_CODEC_ID_PCM_S24BE:
 573         return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->ch_layout.nb_channels);
 574     case AV_CODEC_ID_ADPCM_G722:
 575         /* The actual sample size is half a byte per sample, but since the
 576          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
 577          * the correct parameter for send_samples_bits is 8 bits per stream
 578          * clock. */
 579         return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels);
 580     case AV_CODEC_ID_ADPCM_G726:
 581     case AV_CODEC_ID_ADPCM_G726LE:
 582         return rtp_send_samples(s1, pkt->data, size,
 583                                 st->codecpar->bits_per_coded_sample * st->codecpar->ch_layout.nb_channels);
 584     case AV_CODEC_ID_MP2:
 585     case AV_CODEC_ID_MP3:
 586         rtp_send_mpegaudio(s1, pkt->data, size);
 587         break;
 588     case AV_CODEC_ID_MPEG1VIDEO:
 589     case AV_CODEC_ID_MPEG2VIDEO:
 590         ff_rtp_send_mpegvideo(s1, pkt->data, size);
 591         break;
 592     case AV_CODEC_ID_AAC:
 593         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
 594             ff_rtp_send_latm(s1, pkt->data, size);
 595         else
 596             ff_rtp_send_aac(s1, pkt->data, size);
 597         break;
 598     case AV_CODEC_ID_AMR_NB:
 599     case AV_CODEC_ID_AMR_WB:
 600         ff_rtp_send_amr(s1, pkt->data, size);
 601         break;
 602     case AV_CODEC_ID_AV1:
 603         ff_rtp_send_av1(s1, pkt->data, size, (pkt->flags & AV_PKT_FLAG_KEY) ? 1 : 0);
 604         break;
 605     case AV_CODEC_ID_MPEG2TS:
 606         rtp_send_mpegts_raw(s1, pkt->data, size);
 607         break;
 608     case AV_CODEC_ID_DIRAC:
 609         ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
 610         break;
 611     case AV_CODEC_ID_H264:
 612         ff_rtp_send_h264_hevc(s1, pkt->data, size);
 613         break;
 614     case AV_CODEC_ID_H261:
 615         ff_rtp_send_h261(s1, pkt->data, size);
 616         break;
 617     case AV_CODEC_ID_H263:
 618         if (s->flags & FF_RTP_FLAG_RFC2190) {
 619             size_t mb_info_size;
 620             const uint8_t *mb_info =
 621                 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
 622                                         &mb_info_size);
 623             ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
 624             break;
 625         }
 626         /* Fallthrough */
 627     case AV_CODEC_ID_H263P:
 628         ff_rtp_send_h263(s1, pkt->data, size);
 629         break;
 630     case AV_CODEC_ID_HEVC:
 631         ff_rtp_send_h264_hevc(s1, pkt->data, size);
 632         break;
 633     case AV_CODEC_ID_VORBIS:
 634     case AV_CODEC_ID_THEORA:
 635         ff_rtp_send_xiph(s1, pkt->data, size);
 636         break;
 637     case AV_CODEC_ID_VP8:
 638         ff_rtp_send_vp8(s1, pkt->data, size);
 639         break;
 640     case AV_CODEC_ID_VP9:
 641         ff_rtp_send_vp9(s1, pkt->data, size);
 642         break;
 643     case AV_CODEC_ID_ILBC:
 644         rtp_send_ilbc(s1, pkt->data, size);
 645         break;
 646     case AV_CODEC_ID_MJPEG:
 647         ff_rtp_send_jpeg(s1, pkt->data, size);
 648         break;
 649     case AV_CODEC_ID_BITPACKED:
 650     case AV_CODEC_ID_RAWVIDEO: {
 651         int interlaced = st->codecpar->field_order != AV_FIELD_PROGRESSIVE;
 652
 653         ff_rtp_send_raw_rfc4175(s1, pkt->data, size, interlaced, 0);
 654         if (interlaced)
 655             ff_rtp_send_raw_rfc4175(s1, pkt->data, size, interlaced, 1);
 656         break;
 657         }
 658     case AV_CODEC_ID_OPUS:
 659         if (size > s->max_payload_size) {
 660             av_log(s1, AV_LOG_ERROR,
 661                    "Packet size %d too large for max RTP payload size %d\n",
 662                    size, s->max_payload_size);
 663             return AVERROR(EINVAL);
 664         }
 665         /* Intentional fallthrough */
 666     default:
 667         /* better than nothing : send the codec raw data */
 668         rtp_send_raw(s1, pkt->data, size);
 669         break;
 670     }
 671     return 0;
 672 }
 673
 674 static int rtp_write_trailer(AVFormatContext *s1)
 675 {
 676     RTPMuxContext *s = s1->priv_data;
 677
 678     /* If the caller closes and recreates ->pb, this might actually
 679      * be NULL here even if it was successfully allocated at the start. */
 680     if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
 681         rtcp_send_sr(s1, ff_ntp_time(), 1);
 682     av_freep(&s->buf);
 683
 684     return 0;
 685 }
 686
 687 const FFOutputFormat ff_rtp_muxer = {
 688     .p.name            = "rtp",
 689     .p.long_name       = NULL_IF_CONFIG_SMALL("RTP output"),
 690     .priv_data_size    = sizeof(RTPMuxContext),
 691     .p.audio_codec     = AV_CODEC_ID_PCM_MULAW,
 692     .p.video_codec     = AV_CODEC_ID_MPEG4,
 693     .write_header      = rtp_write_header,
 694     .write_packet      = rtp_write_packet,
 695     .write_trailer     = rtp_write_trailer,
 696     .p.priv_class      = &rtp_muxer_class,
 697     .p.flags           = AVFMT_NODIMENSIONS | AVFMT_TS_NONSTRICT,
 698 };