libavformat/rtp.c

   1 /*
   2  * RTP input/output format
   3  * Copyright (c) 2002 Fabrice Bellard.
   4  *
   5  * This library is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2 of the License, or (at your option) any later version.
   9  *
  10  * This library is distributed in the hope that it will be useful,
  11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with this library; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  18  */
  19 #include "avformat.h"
  20
  21 #include <unistd.h>
  22 #include <sys/types.h>
  23 #include <sys/socket.h>
  24 #include <netinet/in.h>
  25 #ifndef __BEOS__
  26 # include <arpa/inet.h>
  27 #else
  28 # include "barpainet.h"
  29 #endif
  30 #include <netdb.h>
  31
  32 //#define DEBUG
  33
  34
  35 /* TODO: - add RTCP statistics reporting (should be optional).
  36
  37          - add support for h263/mpeg4 packetized output : IDEA: send a
  38          buffer to 'rtp_write_packet' contains all the packets for ONE
  39          frame. Each packet should have a four byte header containing
  40          the length in big endian format (same trick as
  41          'url_open_dyn_packet_buf')
  42 */
  43
  44 #define RTP_VERSION 2
  45
  46 #define RTP_MAX_SDES 256   /* maximum text length for SDES */
  47
  48 /* RTCP paquets use 0.5 % of the bandwidth */
  49 #define RTCP_TX_RATIO_NUM 5
  50 #define RTCP_TX_RATIO_DEN 1000
  51
  52 typedef enum {
  53   RTCP_SR   = 200,
  54   RTCP_RR   = 201,
  55   RTCP_SDES = 202,
  56   RTCP_BYE  = 203,
  57   RTCP_APP  = 204
  58 } rtcp_type_t;
  59
  60 typedef enum {
  61   RTCP_SDES_END    =  0,
  62   RTCP_SDES_CNAME  =  1,
  63   RTCP_SDES_NAME   =  2,
  64   RTCP_SDES_EMAIL  =  3,
  65   RTCP_SDES_PHONE  =  4,
  66   RTCP_SDES_LOC    =  5,
  67   RTCP_SDES_TOOL   =  6,
  68   RTCP_SDES_NOTE   =  7,
  69   RTCP_SDES_PRIV   =  8,
  70   RTCP_SDES_IMG    =  9,
  71   RTCP_SDES_DOOR   = 10,
  72   RTCP_SDES_SOURCE = 11
  73 } rtcp_sdes_type_t;
  74
  75 enum RTPPayloadType {
  76     RTP_PT_ULAW = 0,
  77     RTP_PT_GSM = 3,
  78     RTP_PT_G723 = 4,
  79     RTP_PT_ALAW = 8,
  80     RTP_PT_S16BE_STEREO = 10,
  81     RTP_PT_S16BE_MONO = 11,
  82     RTP_PT_MPEGAUDIO = 14,
  83     RTP_PT_JPEG = 26,
  84     RTP_PT_H261 = 31,
  85     RTP_PT_MPEGVIDEO = 32,
  86     RTP_PT_MPEG2TS = 33,
  87     RTP_PT_H263 = 34, /* old H263 encapsulation */
  88     RTP_PT_PRIVATE = 96,
  89 };
  90
  91 typedef struct RTPContext {
  92     int payload_type;
  93     uint32_t ssrc;
  94     uint16_t seq;
  95     uint32_t timestamp;
  96     uint32_t base_timestamp;
  97     uint32_t cur_timestamp;
  98     int max_payload_size;
  99     /* rtcp sender statistics receive */
 100     int64_t last_rtcp_ntp_time;
 101     int64_t first_rtcp_ntp_time;
 102     uint32_t last_rtcp_timestamp;
 103     /* rtcp sender statistics */
 104     unsigned int packet_count;
 105     unsigned int octet_count;
 106     unsigned int last_octet_count;
 107     int first_packet;
 108     /* buffer for output */
 109     uint8_t buf[RTP_MAX_PACKET_LENGTH];
 110     uint8_t *buf_ptr;
 111 } RTPContext;
 112
 113 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
 114 {
 115     switch(payload_type) {
 116     case RTP_PT_ULAW:
 117         codec->codec_id = CODEC_ID_PCM_MULAW;
 118         codec->channels = 1;
 119         codec->sample_rate = 8000;
 120         break;
 121     case RTP_PT_ALAW:
 122         codec->codec_id = CODEC_ID_PCM_ALAW;
 123         codec->channels = 1;
 124         codec->sample_rate = 8000;
 125         break;
 126     case RTP_PT_S16BE_STEREO:
 127         codec->codec_id = CODEC_ID_PCM_S16BE;
 128         codec->channels = 2;
 129         codec->sample_rate = 44100;
 130         break;
 131     case RTP_PT_S16BE_MONO:
 132         codec->codec_id = CODEC_ID_PCM_S16BE;
 133         codec->channels = 1;
 134         codec->sample_rate = 44100;
 135         break;
 136     case RTP_PT_MPEGAUDIO:
 137         codec->codec_id = CODEC_ID_MP2;
 138         break;
 139     case RTP_PT_JPEG:
 140         codec->codec_id = CODEC_ID_MJPEG;
 141         break;
 142     case RTP_PT_MPEGVIDEO:
 143         codec->codec_id = CODEC_ID_MPEG1VIDEO;
 144         break;
 145     default:
 146         return -1;
 147     }
 148     return 0;
 149 }
 150
 151 /* return < 0 if unknown payload type */
 152 int rtp_get_payload_type(AVCodecContext *codec)
 153 {
 154     int payload_type;
 155
 156     /* compute the payload type */
 157     payload_type = -1;
 158     switch(codec->codec_id) {
 159     case CODEC_ID_PCM_MULAW:
 160         payload_type = RTP_PT_ULAW;
 161         break;
 162     case CODEC_ID_PCM_ALAW:
 163         payload_type = RTP_PT_ALAW;
 164         break;
 165     case CODEC_ID_PCM_S16BE:
 166         if (codec->channels == 1) {
 167             payload_type = RTP_PT_S16BE_MONO;
 168         } else if (codec->channels == 2) {
 169             payload_type = RTP_PT_S16BE_STEREO;
 170         }
 171         break;
 172     case CODEC_ID_MP2:
 173     case CODEC_ID_MP3:
 174         payload_type = RTP_PT_MPEGAUDIO;
 175         break;
 176     case CODEC_ID_MJPEG:
 177         payload_type = RTP_PT_JPEG;
 178         break;
 179     case CODEC_ID_MPEG1VIDEO:
 180         payload_type = RTP_PT_MPEGVIDEO;
 181         break;
 182     default:
 183         break;
 184     }
 185     return payload_type;
 186 }
 187
 188 static inline uint32_t decode_be32(const uint8_t *p)
 189 {
 190     return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
 191 }
 192
 193 static inline uint64_t decode_be64(const uint8_t *p)
 194 {
 195     return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
 196 }
 197
 198 static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
 199 {
 200     RTPContext *s = s1->priv_data;
 201
 202     if (buf[1] != 200)
 203         return -1;
 204     s->last_rtcp_ntp_time = decode_be64(buf + 8);
 205     if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
 206         s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
 207     s->last_rtcp_timestamp = decode_be32(buf + 16);
 208     return 0;
 209 }
 210
 211 /**
 212  * Parse an RTP packet directly sent as raw data. Can only be used if
 213  * 'raw' is given as input file
 214  * @param s1 media file context
 215  * @param pkt returned packet
 216  * @param buf input buffer
 217  * @param len buffer len
 218  * @return zero if no error.
 219  */
 220 int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
 221                      const unsigned char *buf, int len)
 222 {
 223     RTPContext *s = s1->priv_data;
 224     unsigned int ssrc, h;
 225     int payload_type, seq, delta_timestamp;
 226     AVStream *st;
 227     uint32_t timestamp;
 228
 229     if (len < 12)
 230         return -1;
 231
 232     if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
 233         return -1;
 234     if (buf[1] >= 200 && buf[1] <= 204) {
 235         rtcp_parse_packet(s1, buf, len);
 236         return -1;
 237     }
 238     payload_type = buf[1] & 0x7f;
 239     seq  = (buf[2] << 8) | buf[3];
 240     timestamp = decode_be32(buf + 4);
 241     ssrc = decode_be32(buf + 8);
 242
 243     if (s->payload_type < 0) {
 244         s->payload_type = payload_type;
 245
 246         if (payload_type == RTP_PT_MPEG2TS) {
 247             /* XXX: special case : not a single codec but a whole stream */
 248             return -1;
 249         } else {
 250             st = av_new_stream(s1, 0);
 251             if (!st)
 252                 return -1;
 253             rtp_get_codec_info(&st->codec, payload_type);
 254         }
 255     }
 256
 257     /* NOTE: we can handle only one payload type */
 258     if (s->payload_type != payload_type)
 259         return -1;
 260 #if defined(DEBUG) || 1
 261     if (seq != ((s->seq + 1) & 0xffff)) {
 262         printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
 263                payload_type, seq, ((s->seq + 1) & 0xffff));
 264     }
 265     s->seq = seq;
 266 #endif
 267     len -= 12;
 268     buf += 12;
 269     st = s1->streams[0];
 270     switch(st->codec.codec_id) {
 271     case CODEC_ID_MP2:
 272         /* better than nothing: skip mpeg audio RTP header */
 273         if (len <= 4)
 274             return -1;
 275         h = decode_be32(buf);
 276         len -= 4;
 277         buf += 4;
 278         av_new_packet(pkt, len);
 279         memcpy(pkt->data, buf, len);
 280         break;
 281     case CODEC_ID_MPEG1VIDEO:
 282         /* better than nothing: skip mpeg audio RTP header */
 283         if (len <= 4)
 284             return -1;
 285         h = decode_be32(buf);
 286         buf += 4;
 287         len -= 4;
 288         if (h & (1 << 26)) {
 289             /* mpeg2 */
 290             if (len <= 4)
 291                 return -1;
 292             buf += 4;
 293             len -= 4;
 294         }
 295         av_new_packet(pkt, len);
 296         memcpy(pkt->data, buf, len);
 297         break;
 298     default:
 299         av_new_packet(pkt, len);
 300         memcpy(pkt->data, buf, len);
 301         break;
 302     }
 303
 304     switch(st->codec.codec_id) {
 305     case CODEC_ID_MP2:
 306     case CODEC_ID_MPEG1VIDEO:
 307         if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
 308             int64_t addend;
 309             /* XXX: is it really necessary to unify the timestamp base ? */
 310             /* compute pts from timestamp with received ntp_time */
 311             delta_timestamp = timestamp - s->last_rtcp_timestamp;
 312             /* convert to 90 kHz without overflow */
 313             addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
 314             addend = (addend * 5625) >> 14;
 315             pkt->pts = addend + delta_timestamp;
 316         }
 317         break;
 318     default:
 319         /* no timestamp info yet */
 320         break;
 321     }
 322     return 0;
 323 }
 324
 325 static int rtp_read_header(AVFormatContext *s1,
 326                            AVFormatParameters *ap)
 327 {
 328     RTPContext *s = s1->priv_data;
 329     s->payload_type = -1;
 330     s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
 331     s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
 332     return 0;
 333 }
 334
 335 static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
 336 {
 337     char buf[RTP_MAX_PACKET_LENGTH];
 338     int ret;
 339
 340     /* XXX: needs a better API for packet handling ? */
 341     for(;;) {
 342         ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
 343         if (ret < 0)
 344             return AVERROR_IO;
 345         if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
 346             break;
 347     }
 348     return 0;
 349 }
 350
 351 static int rtp_read_close(AVFormatContext *s1)
 352 {
 353     //    RTPContext *s = s1->priv_data;
 354     return 0;
 355 }
 356
 357 static int rtp_probe(AVProbeData *p)
 358 {
 359     if (strstart(p->filename, "rtp://", NULL))
 360         return AVPROBE_SCORE_MAX;
 361     return 0;
 362 }
 363
 364 /* rtp output */
 365
 366 static int rtp_write_header(AVFormatContext *s1)
 367 {
 368     RTPContext *s = s1->priv_data;
 369     int payload_type, max_packet_size;
 370     AVStream *st;
 371
 372     if (s1->nb_streams != 1)
 373         return -1;
 374     st = s1->streams[0];
 375
 376     payload_type = rtp_get_payload_type(&st->codec);
 377     if (payload_type < 0)
 378         payload_type = RTP_PT_PRIVATE; /* private payload type */
 379     s->payload_type = payload_type;
 380
 381     s->base_timestamp = random();
 382     s->timestamp = s->base_timestamp;
 383     s->ssrc = random();
 384     s->first_packet = 1;
 385
 386     max_packet_size = url_fget_max_packet_size(&s1->pb);
 387     if (max_packet_size <= 12)
 388         return AVERROR_IO;
 389     s->max_payload_size = max_packet_size - 12;
 390
 391     switch(st->codec.codec_id) {
 392     case CODEC_ID_MP2:
 393     case CODEC_ID_MP3:
 394         s->buf_ptr = s->buf + 4;
 395         s->cur_timestamp = 0;
 396         break;
 397     case CODEC_ID_MPEG1VIDEO:
 398         s->cur_timestamp = 0;
 399         break;
 400     default:
 401         s->buf_ptr = s->buf;
 402         break;
 403     }
 404
 405     return 0;
 406 }
 407
 408 /* send an rtcp sender report packet */
 409 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 410 {
 411     RTPContext *s = s1->priv_data;
 412 #if defined(DEBUG)
 413     printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
 414 #endif
 415     put_byte(&s1->pb, (RTP_VERSION << 6));
 416     put_byte(&s1->pb, 200);
 417     put_be16(&s1->pb, 6); /* length in words - 1 */
 418     put_be32(&s1->pb, s->ssrc);
 419     put_be64(&s1->pb, ntp_time);
 420     put_be32(&s1->pb, s->timestamp);
 421     put_be32(&s1->pb, s->packet_count);
 422     put_be32(&s1->pb, s->octet_count);
 423     put_flush_packet(&s1->pb);
 424 }
 425
 426 /* send an rtp packet. sequence number is incremented, but the caller
 427    must update the timestamp itself */
 428 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
 429 {
 430     RTPContext *s = s1->priv_data;
 431
 432 #ifdef DEBUG
 433     printf("rtp_send_data size=%d\n", len);
 434 #endif
 435
 436     /* build the RTP header */
 437     put_byte(&s1->pb, (RTP_VERSION << 6));
 438     put_byte(&s1->pb, s->payload_type & 0x7f);
 439     put_be16(&s1->pb, s->seq);
 440     put_be32(&s1->pb, s->timestamp);
 441     put_be32(&s1->pb, s->ssrc);
 442
 443     put_buffer(&s1->pb, buf1, len);
 444     put_flush_packet(&s1->pb);
 445
 446     s->seq++;
 447     s->octet_count += len;
 448     s->packet_count++;
 449 }
 450
 451 /* send an integer number of samples and compute time stamp and fill
 452    the rtp send buffer before sending. */
 453 static void rtp_send_samples(AVFormatContext *s1,
 454                              const uint8_t *buf1, int size, int sample_size)
 455 {
 456     RTPContext *s = s1->priv_data;
 457     int len, max_packet_size, n;
 458
 459     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
 460     /* not needed, but who nows */
 461     if ((size % sample_size) != 0)
 462         av_abort();
 463     while (size > 0) {
 464         len = (max_packet_size - (s->buf_ptr - s->buf));
 465         if (len > size)
 466             len = size;
 467
 468         /* copy data */
 469         memcpy(s->buf_ptr, buf1, len);
 470         s->buf_ptr += len;
 471         buf1 += len;
 472         size -= len;
 473         n = (s->buf_ptr - s->buf);
 474         /* if buffer full, then send it */
 475         if (n >= max_packet_size) {
 476             rtp_send_data(s1, s->buf, n);
 477             s->buf_ptr = s->buf;
 478             /* update timestamp */
 479             s->timestamp += n / sample_size;
 480         }
 481     }
 482 }
 483
 484 /* NOTE: we suppose that exactly one frame is given as argument here */
 485 /* XXX: test it */
 486 static void rtp_send_mpegaudio(AVFormatContext *s1,
 487                                const uint8_t *buf1, int size)
 488 {
 489     RTPContext *s = s1->priv_data;
 490     AVStream *st = s1->streams[0];
 491     int len, count, max_packet_size;
 492
 493     max_packet_size = s->max_payload_size;
 494
 495     /* test if we must flush because not enough space */
 496     len = (s->buf_ptr - s->buf);
 497     if ((len + size) > max_packet_size) {
 498         if (len > 4) {
 499             rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
 500             s->buf_ptr = s->buf + 4;
 501             /* 90 KHz time stamp */
 502             s->timestamp = s->base_timestamp +
 503                 (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
 504         }
 505     }
 506
 507     /* add the packet */
 508     if (size > max_packet_size) {
 509         /* big packet: fragment */
 510         count = 0;
 511         while (size > 0) {
 512             len = max_packet_size - 4;
 513             if (len > size)
 514                 len = size;
 515             /* build fragmented packet */
 516             s->buf[0] = 0;
 517             s->buf[1] = 0;
 518             s->buf[2] = count >> 8;
 519             s->buf[3] = count;
 520             memcpy(s->buf + 4, buf1, len);
 521             rtp_send_data(s1, s->buf, len + 4);
 522             size -= len;
 523             buf1 += len;
 524             count += len;
 525         }
 526     } else {
 527         if (s->buf_ptr == s->buf + 4) {
 528             /* no fragmentation possible */
 529             s->buf[0] = 0;
 530             s->buf[1] = 0;
 531             s->buf[2] = 0;
 532             s->buf[3] = 0;
 533         }
 534         memcpy(s->buf_ptr, buf1, size);
 535         s->buf_ptr += size;
 536     }
 537     s->cur_timestamp += st->codec.frame_size;
 538 }
 539
 540 /* NOTE: a single frame must be passed with sequence header if
 541    needed. XXX: use slices. */
 542 static void rtp_send_mpegvideo(AVFormatContext *s1,
 543                                const uint8_t *buf1, int size)
 544 {
 545     RTPContext *s = s1->priv_data;
 546     AVStream *st = s1->streams[0];
 547     int len, h, max_packet_size;
 548     uint8_t *q;
 549
 550     max_packet_size = s->max_payload_size;
 551
 552     while (size > 0) {
 553         /* XXX: more correct headers */
 554         h = 0;
 555         if (st->codec.sub_id == 2)
 556             h |= 1 << 26; /* mpeg 2 indicator */
 557         q = s->buf;
 558         *q++ = h >> 24;
 559         *q++ = h >> 16;
 560         *q++ = h >> 8;
 561         *q++ = h;
 562
 563         if (st->codec.sub_id == 2) {
 564             h = 0;
 565             *q++ = h >> 24;
 566             *q++ = h >> 16;
 567             *q++ = h >> 8;
 568             *q++ = h;
 569         }
 570
 571         len = max_packet_size - (q - s->buf);
 572         if (len > size)
 573             len = size;
 574
 575         memcpy(q, buf1, len);
 576         q += len;
 577
 578         /* 90 KHz time stamp */
 579         s->timestamp = s->base_timestamp +
 580             av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
 581         rtp_send_data(s1, s->buf, q - s->buf);
 582
 583         buf1 += len;
 584         size -= len;
 585     }
 586     s->cur_timestamp++;
 587 }
 588
 589 static void rtp_send_raw(AVFormatContext *s1,
 590                          const uint8_t *buf1, int size)
 591 {
 592     RTPContext *s = s1->priv_data;
 593     AVStream *st = s1->streams[0];
 594     int len, max_packet_size;
 595
 596     max_packet_size = s->max_payload_size;
 597
 598     while (size > 0) {
 599         len = max_packet_size;
 600         if (len > size)
 601             len = size;
 602
 603         /* 90 KHz time stamp */
 604         s->timestamp = s->base_timestamp +
 605             av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
 606         rtp_send_data(s1, buf1, len);
 607
 608         buf1 += len;
 609         size -= len;
 610     }
 611     s->cur_timestamp++;
 612 }
 613
 614 /* write an RTP packet. 'buf1' must contain a single specific frame. */
 615 static int rtp_write_packet(AVFormatContext *s1, int stream_index,
 616                             const uint8_t *buf1, int size, int64_t pts)
 617 {
 618     RTPContext *s = s1->priv_data;
 619     AVStream *st = s1->streams[0];
 620     int rtcp_bytes;
 621     int64_t ntp_time;
 622
 623 #ifdef DEBUG
 624     printf("%d: write len=%d\n", stream_index, size);
 625 #endif
 626
 627     /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
 628     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 629         RTCP_TX_RATIO_DEN;
 630     if (s->first_packet || rtcp_bytes >= 28) {
 631         /* compute NTP time */
 632         /* XXX: 90 kHz timestamp hardcoded */
 633         ntp_time = (pts << 28) / 5625;
 634         rtcp_send_sr(s1, ntp_time);
 635         s->last_octet_count = s->octet_count;
 636         s->first_packet = 0;
 637     }
 638
 639     switch(st->codec.codec_id) {
 640     case CODEC_ID_PCM_MULAW:
 641     case CODEC_ID_PCM_ALAW:
 642     case CODEC_ID_PCM_U8:
 643     case CODEC_ID_PCM_S8:
 644         rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
 645         break;
 646     case CODEC_ID_PCM_U16BE:
 647     case CODEC_ID_PCM_U16LE:
 648     case CODEC_ID_PCM_S16BE:
 649     case CODEC_ID_PCM_S16LE:
 650         rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
 651         break;
 652     case CODEC_ID_MP2:
 653     case CODEC_ID_MP3:
 654         rtp_send_mpegaudio(s1, buf1, size);
 655         break;
 656     case CODEC_ID_MPEG1VIDEO:
 657         rtp_send_mpegvideo(s1, buf1, size);
 658         break;
 659     default:
 660         /* better than nothing : send the codec raw data */
 661         rtp_send_raw(s1, buf1, size);
 662         break;
 663     }
 664     return 0;
 665 }
 666
 667 static int rtp_write_trailer(AVFormatContext *s1)
 668 {
 669     //    RTPContext *s = s1->priv_data;
 670     return 0;
 671 }
 672
 673 AVInputFormat rtp_demux = {
 674     "rtp",
 675     "RTP input format",
 676     sizeof(RTPContext),
 677     rtp_probe,
 678     rtp_read_header,
 679     rtp_read_packet,
 680     rtp_read_close,
 681     .flags = AVFMT_NOHEADER,
 682 };
 683
 684 AVOutputFormat rtp_mux = {
 685     "rtp",
 686     "RTP output format",
 687     NULL,
 688     NULL,
 689     sizeof(RTPContext),
 690     CODEC_ID_PCM_MULAW,
 691     CODEC_ID_NONE,
 692     rtp_write_header,
 693     rtp_write_packet,
 694     rtp_write_trailer,
 695 };
 696
 697 int rtp_init(void)
 698 {
 699     av_register_output_format(&rtp_mux);
 700     av_register_input_format(&rtp_demux);
 701     return 0;
 702 }