libav/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 #include <arpa/inet.h>
  26 #include <netdb.h>
  27
  28 //#define DEBUG
  29
  30
  31 /* TODO: - add RTCP statistics reporting (should be optional).
  32
  33          - add support for h263/mpeg4 packetized output : IDEA: send a
  34          buffer to 'rtp_write_packet' contains all the packets for ONE
  35          frame. Each packet should have a four byte header containing
  36          the length in big endian format (same trick as
  37          'url_open_dyn_packet_buf')
  38 */
  39
  40 #define RTP_VERSION 2
  41
  42 #define RTP_MAX_SDES 256   /* maximum text length for SDES */
  43
  44 /* RTCP paquets use 0.5 % of the bandwidth */
  45 #define RTCP_TX_RATIO_NUM 5
  46 #define RTCP_TX_RATIO_DEN 1000
  47
  48 typedef enum {
  49   RTCP_SR   = 200,
  50   RTCP_RR   = 201,
  51   RTCP_SDES = 202,
  52   RTCP_BYE  = 203,
  53   RTCP_APP  = 204
  54 } rtcp_type_t;
  55
  56 typedef enum {
  57   RTCP_SDES_END    =  0,
  58   RTCP_SDES_CNAME  =  1,
  59   RTCP_SDES_NAME   =  2,
  60   RTCP_SDES_EMAIL  =  3,
  61   RTCP_SDES_PHONE  =  4,
  62   RTCP_SDES_LOC    =  5,
  63   RTCP_SDES_TOOL   =  6,
  64   RTCP_SDES_NOTE   =  7,
  65   RTCP_SDES_PRIV   =  8,
  66   RTCP_SDES_IMG    =  9,
  67   RTCP_SDES_DOOR   = 10,
  68   RTCP_SDES_SOURCE = 11
  69 } rtcp_sdes_type_t;
  70
  71 enum RTPPayloadType {
  72     RTP_PT_ULAW = 0,
  73     RTP_PT_GSM = 3,
  74     RTP_PT_G723 = 4,
  75     RTP_PT_ALAW = 8,
  76     RTP_PT_S16BE_STEREO = 10,
  77     RTP_PT_S16BE_MONO = 11,
  78     RTP_PT_MPEGAUDIO = 14,
  79     RTP_PT_JPEG = 26,
  80     RTP_PT_H261 = 31,
  81     RTP_PT_MPEGVIDEO = 32,
  82     RTP_PT_MPEG2TS = 33,
  83     RTP_PT_H263 = 34, /* old H263 encapsulation */
  84 };
  85
  86 typedef struct RTPContext {
  87     int payload_type;
  88     UINT32 ssrc;
  89     UINT16 seq;
  90     UINT32 timestamp;
  91     UINT32 base_timestamp;
  92     UINT32 cur_timestamp;
  93     int max_payload_size;
  94     /* rtcp sender statistics receive */
  95     INT64 last_rtcp_ntp_time;
  96     UINT32 last_rtcp_timestamp;
  97     /* rtcp sender statistics */
  98     unsigned int packet_count;
  99     unsigned int octet_count;
 100     unsigned int last_octet_count;
 101     int first_packet;
 102     /* buffer for output */
 103     UINT8 buf[RTP_MAX_PACKET_LENGTH];
 104     UINT8 *buf_ptr;
 105 } RTPContext;
 106
 107 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
 108 {
 109     switch(payload_type) {
 110     case RTP_PT_ULAW:
 111         codec->codec_id = CODEC_ID_PCM_MULAW;
 112         codec->channels = 1;
 113         codec->sample_rate = 8000;
 114         break;
 115     case RTP_PT_ALAW:
 116         codec->codec_id = CODEC_ID_PCM_ALAW;
 117         codec->channels = 1;
 118         codec->sample_rate = 8000;
 119         break;
 120     case RTP_PT_S16BE_STEREO:
 121         codec->codec_id = CODEC_ID_PCM_S16BE;
 122         codec->channels = 2;
 123         codec->sample_rate = 44100;
 124         break;
 125     case RTP_PT_S16BE_MONO:
 126         codec->codec_id = CODEC_ID_PCM_S16BE;
 127         codec->channels = 1;
 128         codec->sample_rate = 44100;
 129         break;
 130     case RTP_PT_MPEGAUDIO:
 131         codec->codec_id = CODEC_ID_MP2;
 132         break;
 133     case RTP_PT_JPEG:
 134         codec->codec_id = CODEC_ID_MJPEG;
 135         break;
 136     case RTP_PT_MPEGVIDEO:
 137         codec->codec_id = CODEC_ID_MPEG1VIDEO;
 138         break;
 139     default:
 140         return -1;
 141     }
 142     return 0;
 143 }
 144
 145 /* return < 0 if unknown payload type */
 146 int rtp_get_payload_type(AVCodecContext *codec)
 147 {
 148     int payload_type;
 149
 150     /* compute the payload type */
 151     payload_type = -1;
 152     switch(codec->codec_id) {
 153     case CODEC_ID_PCM_MULAW:
 154         payload_type = RTP_PT_ULAW;
 155         break;
 156     case CODEC_ID_PCM_ALAW:
 157         payload_type = RTP_PT_ALAW;
 158         break;
 159     case CODEC_ID_PCM_S16BE:
 160         if (codec->channels == 1) {
 161             payload_type = RTP_PT_S16BE_MONO;
 162         } else if (codec->channels == 2) {
 163             payload_type = RTP_PT_S16BE_STEREO;
 164         }
 165         break;
 166     case CODEC_ID_MP2:
 167     case CODEC_ID_MP3LAME:
 168         payload_type = RTP_PT_MPEGAUDIO;
 169         break;
 170     case CODEC_ID_MJPEG:
 171         payload_type = RTP_PT_JPEG;
 172         break;
 173     case CODEC_ID_MPEG1VIDEO:
 174         payload_type = RTP_PT_MPEGVIDEO;
 175         break;
 176     default:
 177         break;
 178     }
 179     return payload_type;
 180 }
 181
 182 static inline UINT32 decode_be32(const UINT8 *p)
 183 {
 184     return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
 185 }
 186
 187 static inline UINT32 decode_be64(const UINT8 *p)
 188 {
 189     return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
 190 }
 191
 192 static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
 193 {
 194     RTPContext *s = s1->priv_data;
 195
 196     if (buf[1] != 200)
 197         return -1;
 198     s->last_rtcp_ntp_time = decode_be64(buf + 8);
 199     s->last_rtcp_timestamp = decode_be32(buf + 16);
 200     return 0;
 201 }
 202
 203 /**
 204  * Parse an RTP packet directly sent as raw data. Can only be used if
 205  * 'raw' is given as input file
 206  * @param s1 media file context
 207  * @param pkt returned packet
 208  * @param buf input buffer
 209  * @param len buffer len
 210  * @return zero if no error.
 211  */
 212 int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
 213                      const unsigned char *buf, int len)
 214 {
 215     RTPContext *s = s1->priv_data;
 216     unsigned int ssrc, h;
 217     int payload_type, seq, delta_timestamp;
 218     AVStream *st;
 219     UINT32 timestamp;
 220
 221     if (len < 12)
 222         return -1;
 223
 224     if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
 225         return -1;
 226     if (buf[1] >= 200 && buf[1] <= 204) {
 227         rtcp_parse_packet(s1, buf, len);
 228         return -1;
 229     }
 230     payload_type = buf[1] & 0x7f;
 231     seq  = (buf[2] << 8) | buf[3];
 232     timestamp = decode_be32(buf + 4);
 233     ssrc = decode_be32(buf + 8);
 234
 235     if (s->payload_type < 0) {
 236         s->payload_type = payload_type;
 237
 238         if (payload_type == RTP_PT_MPEG2TS) {
 239             /* XXX: special case : not a single codec but a whole stream */
 240             return -1;
 241         } else {
 242             st = av_new_stream(s1, 0);
 243             if (!st)
 244                 return -1;
 245             rtp_get_codec_info(&st->codec, payload_type);
 246         }
 247     }
 248
 249     /* NOTE: we can handle only one payload type */
 250     if (s->payload_type != payload_type)
 251         return -1;
 252 #if defined(DEBUG) || 1
 253     if (seq != ((s->seq + 1) & 0xffff)) {
 254         printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
 255                payload_type, seq, ((s->seq + 1) & 0xffff));
 256     }
 257     s->seq = seq;
 258 #endif
 259     len -= 12;
 260     buf += 12;
 261     st = s1->streams[0];
 262     switch(st->codec.codec_id) {
 263     case CODEC_ID_MP2:
 264         /* better than nothing: skip mpeg audio RTP header */
 265         if (len <= 4)
 266             return -1;
 267         h = decode_be32(buf);
 268         len -= 4;
 269         buf += 4;
 270         av_new_packet(pkt, len);
 271         memcpy(pkt->data, buf, len);
 272         break;
 273     case CODEC_ID_MPEG1VIDEO:
 274         /* better than nothing: skip mpeg audio RTP header */
 275         if (len <= 4)
 276             return -1;
 277         h = decode_be32(buf);
 278         buf += 4;
 279         len -= 4;
 280         if (h & (1 << 26)) {
 281             /* mpeg2 */
 282             if (len <= 4)
 283                 return -1;
 284             buf += 4;
 285             len -= 4;
 286         }
 287         av_new_packet(pkt, len);
 288         memcpy(pkt->data, buf, len);
 289         break;
 290     default:
 291         av_new_packet(pkt, len);
 292         memcpy(pkt->data, buf, len);
 293         break;
 294     }
 295
 296     if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
 297         /* compute pts from timestamp with received ntp_time */
 298         delta_timestamp = timestamp - s->last_rtcp_timestamp;
 299         /* XXX: do conversion, but not needed for mpeg at 90 KhZ */
 300         pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
 301     }
 302     return 0;
 303 }
 304
 305 static int rtp_read_header(AVFormatContext *s1,
 306                            AVFormatParameters *ap)
 307 {
 308     RTPContext *s = s1->priv_data;
 309     s->payload_type = -1;
 310     s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
 311     return 0;
 312 }
 313
 314 static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
 315 {
 316     char buf[RTP_MAX_PACKET_LENGTH];
 317     int ret;
 318
 319     /* XXX: needs a better API for packet handling ? */
 320     for(;;) {
 321         ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
 322         if (ret < 0)
 323             return AVERROR_IO;
 324         if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
 325             break;
 326     }
 327     return 0;
 328 }
 329
 330 static int rtp_read_close(AVFormatContext *s1)
 331 {
 332     //    RTPContext *s = s1->priv_data;
 333     return 0;
 334 }
 335
 336 static int rtp_probe(AVProbeData *p)
 337 {
 338     if (strstart(p->filename, "rtp://", NULL))
 339         return AVPROBE_SCORE_MAX;
 340     return 0;
 341 }
 342
 343 /* rtp output */
 344
 345 static int rtp_write_header(AVFormatContext *s1)
 346 {
 347     RTPContext *s = s1->priv_data;
 348     int payload_type, max_packet_size;
 349     AVStream *st;
 350
 351     if (s1->nb_streams != 1)
 352         return -1;
 353     st = s1->streams[0];
 354
 355     payload_type = rtp_get_payload_type(&st->codec);
 356     if (payload_type < 0)
 357         return -1;
 358     s->payload_type = payload_type;
 359
 360     s->base_timestamp = random();
 361     s->timestamp = s->base_timestamp;
 362     s->ssrc = random();
 363     s->first_packet = 1;
 364
 365     max_packet_size = url_fget_max_packet_size(&s1->pb);
 366     if (max_packet_size <= 12)
 367         return AVERROR_IO;
 368     s->max_payload_size = max_packet_size - 12;
 369
 370     switch(st->codec.codec_id) {
 371     case CODEC_ID_MP2:
 372     case CODEC_ID_MP3LAME:
 373         s->buf_ptr = s->buf + 4;
 374         s->cur_timestamp = 0;
 375         break;
 376     case CODEC_ID_MPEG1VIDEO:
 377         s->cur_timestamp = 0;
 378         break;
 379     default:
 380         s->buf_ptr = s->buf;
 381         break;
 382     }
 383
 384     return 0;
 385 }
 386
 387 /* send an rtcp sender report packet */
 388 static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
 389 {
 390     RTPContext *s = s1->priv_data;
 391 #if defined(DEBUG)
 392     printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
 393 #endif
 394     put_byte(&s1->pb, (RTP_VERSION << 6));
 395     put_byte(&s1->pb, 200);
 396     put_be16(&s1->pb, 6); /* length in words - 1 */
 397     put_be32(&s1->pb, s->ssrc);
 398     put_be64(&s1->pb, ntp_time);
 399     put_be32(&s1->pb, s->timestamp);
 400     put_be32(&s1->pb, s->packet_count);
 401     put_be32(&s1->pb, s->octet_count);
 402     put_flush_packet(&s1->pb);
 403 }
 404
 405 /* send an rtp packet. sequence number is incremented, but the caller
 406    must update the timestamp itself */
 407 static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
 408 {
 409     RTPContext *s = s1->priv_data;
 410
 411 #ifdef DEBUG
 412     printf("rtp_send_data size=%d\n", len);
 413 #endif
 414
 415     /* build the RTP header */
 416     put_byte(&s1->pb, (RTP_VERSION << 6));
 417     put_byte(&s1->pb, s->payload_type & 0x7f);
 418     put_be16(&s1->pb, s->seq);
 419     put_be32(&s1->pb, s->timestamp);
 420     put_be32(&s1->pb, s->ssrc);
 421
 422     put_buffer(&s1->pb, buf1, len);
 423     put_flush_packet(&s1->pb);
 424
 425     s->seq++;
 426     s->octet_count += len;
 427     s->packet_count++;
 428 }
 429
 430 /* send an integer number of samples and compute time stamp and fill
 431    the rtp send buffer before sending. */
 432 static void rtp_send_samples(AVFormatContext *s1,
 433                              UINT8 *buf1, int size, int sample_size)
 434 {
 435     RTPContext *s = s1->priv_data;
 436     int len, max_packet_size, n;
 437
 438     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
 439     /* not needed, but who nows */
 440     if ((size % sample_size) != 0)
 441         av_abort();
 442     while (size > 0) {
 443         len = (max_packet_size - (s->buf_ptr - s->buf));
 444         if (len > size)
 445             len = size;
 446
 447         /* copy data */
 448         memcpy(s->buf_ptr, buf1, len);
 449         s->buf_ptr += len;
 450         buf1 += len;
 451         size -= len;
 452         n = (s->buf_ptr - s->buf);
 453         /* if buffer full, then send it */
 454         if (n >= max_packet_size) {
 455             rtp_send_data(s1, s->buf, n);
 456             s->buf_ptr = s->buf;
 457             /* update timestamp */
 458             s->timestamp += n / sample_size;
 459         }
 460     }
 461 }
 462
 463 /* NOTE: we suppose that exactly one frame is given as argument here */
 464 /* XXX: test it */
 465 static void rtp_send_mpegaudio(AVFormatContext *s1,
 466                                UINT8 *buf1, int size)
 467 {
 468     RTPContext *s = s1->priv_data;
 469     AVStream *st = s1->streams[0];
 470     int len, count, max_packet_size;
 471
 472     max_packet_size = s->max_payload_size;
 473
 474     /* test if we must flush because not enough space */
 475     len = (s->buf_ptr - s->buf);
 476     if ((len + size) > max_packet_size) {
 477         if (len > 4) {
 478             rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
 479             s->buf_ptr = s->buf + 4;
 480             /* 90 KHz time stamp */
 481             s->timestamp = s->base_timestamp +
 482                 (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
 483         }
 484     }
 485
 486     /* add the packet */
 487     if (size > max_packet_size) {
 488         /* big packet: fragment */
 489         count = 0;
 490         while (size > 0) {
 491             len = max_packet_size - 4;
 492             if (len > size)
 493                 len = size;
 494             /* build fragmented packet */
 495             s->buf[0] = 0;
 496             s->buf[1] = 0;
 497             s->buf[2] = count >> 8;
 498             s->buf[3] = count;
 499             memcpy(s->buf + 4, buf1, len);
 500             rtp_send_data(s1, s->buf, len + 4);
 501             size -= len;
 502             buf1 += len;
 503             count += len;
 504         }
 505     } else {
 506         if (s->buf_ptr == s->buf + 4) {
 507             /* no fragmentation possible */
 508             s->buf[0] = 0;
 509             s->buf[1] = 0;
 510             s->buf[2] = 0;
 511             s->buf[3] = 0;
 512         }
 513         memcpy(s->buf_ptr, buf1, size);
 514         s->buf_ptr += size;
 515     }
 516     s->cur_timestamp += st->codec.frame_size;
 517 }
 518
 519 /* NOTE: a single frame must be passed with sequence header if
 520    needed. XXX: use slices. */
 521 static void rtp_send_mpegvideo(AVFormatContext *s1,
 522                                UINT8 *buf1, int size)
 523 {
 524     RTPContext *s = s1->priv_data;
 525     AVStream *st = s1->streams[0];
 526     int len, h, max_packet_size;
 527     UINT8 *q;
 528
 529     max_packet_size = s->max_payload_size;
 530
 531     while (size > 0) {
 532         /* XXX: more correct headers */
 533         h = 0;
 534         if (st->codec.sub_id == 2)
 535             h |= 1 << 26; /* mpeg 2 indicator */
 536         q = s->buf;
 537         *q++ = h >> 24;
 538         *q++ = h >> 16;
 539         *q++ = h >> 8;
 540         *q++ = h;
 541
 542         if (st->codec.sub_id == 2) {
 543             h = 0;
 544             *q++ = h >> 24;
 545             *q++ = h >> 16;
 546             *q++ = h >> 8;
 547             *q++ = h;
 548         }
 549
 550         len = max_packet_size - (q - s->buf);
 551         if (len > size)
 552             len = size;
 553
 554         memcpy(q, buf1, len);
 555         q += len;
 556
 557         /* 90 KHz time stamp */
 558         /* XXX: overflow */
 559         s->timestamp = s->base_timestamp +
 560             (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
 561         rtp_send_data(s1, s->buf, q - s->buf);
 562
 563         buf1 += len;
 564         size -= len;
 565     }
 566     s->cur_timestamp++;
 567 }
 568
 569 /* write an RTP packet. 'buf1' must contain a single specific frame. */
 570 static int rtp_write_packet(AVFormatContext *s1, int stream_index,
 571                             UINT8 *buf1, int size, int force_pts)
 572 {
 573     RTPContext *s = s1->priv_data;
 574     AVStream *st = s1->streams[0];
 575     int rtcp_bytes;
 576     INT64 ntp_time;
 577
 578 #ifdef DEBUG
 579     printf("%d: write len=%d\n", stream_index, size);
 580 #endif
 581
 582     /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
 583     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 584         RTCP_TX_RATIO_DEN;
 585     if (s->first_packet || rtcp_bytes >= 28) {
 586         /* compute NTP time */
 587         ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
 588         rtcp_send_sr(s1, ntp_time);
 589         s->last_octet_count = s->octet_count;
 590         s->first_packet = 0;
 591     }
 592
 593     switch(st->codec.codec_id) {
 594     case CODEC_ID_PCM_MULAW:
 595     case CODEC_ID_PCM_ALAW:
 596     case CODEC_ID_PCM_U8:
 597     case CODEC_ID_PCM_S8:
 598         rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
 599         break;
 600     case CODEC_ID_PCM_U16BE:
 601     case CODEC_ID_PCM_U16LE:
 602     case CODEC_ID_PCM_S16BE:
 603     case CODEC_ID_PCM_S16LE:
 604         rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
 605         break;
 606     case CODEC_ID_MP2:
 607     case CODEC_ID_MP3LAME:
 608         rtp_send_mpegaudio(s1, buf1, size);
 609         break;
 610     case CODEC_ID_MPEG1VIDEO:
 611         rtp_send_mpegvideo(s1, buf1, size);
 612         break;
 613     default:
 614         return AVERROR_IO;
 615     }
 616     return 0;
 617 }
 618
 619 static int rtp_write_trailer(AVFormatContext *s1)
 620 {
 621     //    RTPContext *s = s1->priv_data;
 622     return 0;
 623 }
 624
 625 AVInputFormat rtp_demux = {
 626     "rtp",
 627     "RTP input format",
 628     sizeof(RTPContext),
 629     rtp_probe,
 630     rtp_read_header,
 631     rtp_read_packet,
 632     rtp_read_close,
 633     flags: AVFMT_NOHEADER,
 634 };
 635
 636 AVOutputFormat rtp_mux = {
 637     "rtp",
 638     "RTP output format",
 639     NULL,
 640     NULL,
 641     sizeof(RTPContext),
 642     CODEC_ID_PCM_MULAW,
 643     CODEC_ID_NONE,
 644     rtp_write_header,
 645     rtp_write_packet,
 646     rtp_write_trailer,
 647 };
 648
 649 int rtp_init(void)
 650 {
 651     av_register_output_format(&rtp_mux);
 652     av_register_input_format(&rtp_demux);
 653     return 0;
 654 }