rtp.c 20.3 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19
/*
 * RTP input/output format
 * Copyright (c) 2002 Fabrice Bellard.
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */
#include "avformat.h"
20
#include "mpegts.h"
21 22 23

#include <unistd.h>
#include <sys/types.h>
Michael Niedermayer's avatar
Michael Niedermayer committed
24
#include <sys/socket.h>
25
#include <netinet/in.h>
26 27 28 29 30
#ifndef __BEOS__
# include <arpa/inet.h>
#else
# include "barpainet.h"
#endif
31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75
#include <netdb.h>

//#define DEBUG


/* TODO: - add RTCP statistics reporting (should be optional).

         - add support for h263/mpeg4 packetized output : IDEA: send a
         buffer to 'rtp_write_packet' contains all the packets for ONE
         frame. Each packet should have a four byte header containing
         the length in big endian format (same trick as
         'url_open_dyn_packet_buf') 
*/

#define RTP_VERSION 2

#define RTP_MAX_SDES 256   /* maximum text length for SDES */

/* RTCP paquets use 0.5 % of the bandwidth */
#define RTCP_TX_RATIO_NUM 5
#define RTCP_TX_RATIO_DEN 1000

typedef enum {
  RTCP_SR   = 200,
  RTCP_RR   = 201,
  RTCP_SDES = 202,
  RTCP_BYE  = 203,
  RTCP_APP  = 204
} rtcp_type_t;

typedef enum {
  RTCP_SDES_END    =  0,
  RTCP_SDES_CNAME  =  1,
  RTCP_SDES_NAME   =  2,
  RTCP_SDES_EMAIL  =  3,
  RTCP_SDES_PHONE  =  4,
  RTCP_SDES_LOC    =  5,
  RTCP_SDES_TOOL   =  6,
  RTCP_SDES_NOTE   =  7,
  RTCP_SDES_PRIV   =  8, 
  RTCP_SDES_IMG    =  9,
  RTCP_SDES_DOOR   = 10,
  RTCP_SDES_SOURCE = 11
} rtcp_sdes_type_t;

76 77 78
struct RTPDemuxContext {
    AVFormatContext *ic;
    AVStream *st;
79
    int payload_type;
80 81 82 83 84
    uint32_t ssrc;
    uint16_t seq;
    uint32_t timestamp;
    uint32_t base_timestamp;
    uint32_t cur_timestamp;
85
    int max_payload_size;
86 87 88 89
    MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
    int read_buf_index;
    int read_buf_size;
    
90
    /* rtcp sender statistics receive */
91
    int64_t last_rtcp_ntp_time;
92
    int64_t first_rtcp_ntp_time;
93
    uint32_t last_rtcp_timestamp;
94 95 96 97 98 99
    /* rtcp sender statistics */
    unsigned int packet_count;
    unsigned int octet_count;
    unsigned int last_octet_count;
    int first_packet;
    /* buffer for output */
100 101
    uint8_t buf[RTP_MAX_PACKET_LENGTH];
    uint8_t *buf_ptr;
102
};
103 104 105 106 107

int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
    switch(payload_type) {
    case RTP_PT_ULAW:
108
        codec->codec_type = CODEC_TYPE_AUDIO;
109 110 111 112 113
        codec->codec_id = CODEC_ID_PCM_MULAW;
        codec->channels = 1;
        codec->sample_rate = 8000;
        break;
    case RTP_PT_ALAW:
114
        codec->codec_type = CODEC_TYPE_AUDIO;
115 116 117 118 119
        codec->codec_id = CODEC_ID_PCM_ALAW;
        codec->channels = 1;
        codec->sample_rate = 8000;
        break;
    case RTP_PT_S16BE_STEREO:
120
        codec->codec_type = CODEC_TYPE_AUDIO;
121 122 123 124 125
        codec->codec_id = CODEC_ID_PCM_S16BE;
        codec->channels = 2;
        codec->sample_rate = 44100;
        break;
    case RTP_PT_S16BE_MONO:
126
        codec->codec_type = CODEC_TYPE_AUDIO;
127 128 129 130 131
        codec->codec_id = CODEC_ID_PCM_S16BE;
        codec->channels = 1;
        codec->sample_rate = 44100;
        break;
    case RTP_PT_MPEGAUDIO:
132
        codec->codec_type = CODEC_TYPE_AUDIO;
133 134 135
        codec->codec_id = CODEC_ID_MP2;
        break;
    case RTP_PT_JPEG:
136
        codec->codec_type = CODEC_TYPE_VIDEO;
137 138 139
        codec->codec_id = CODEC_ID_MJPEG;
        break;
    case RTP_PT_MPEGVIDEO:
140
        codec->codec_type = CODEC_TYPE_VIDEO;
141 142
        codec->codec_id = CODEC_ID_MPEG1VIDEO;
        break;
143 144 145 146
    case RTP_PT_MPEG2TS:
        codec->codec_type = CODEC_TYPE_DATA;
        codec->codec_id = CODEC_ID_MPEG2TS;
        break;
147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174
    default:
        return -1;
    }
    return 0;
}

/* return < 0 if unknown payload type */
int rtp_get_payload_type(AVCodecContext *codec)
{
    int payload_type;

    /* compute the payload type */
    payload_type = -1;
    switch(codec->codec_id) {
    case CODEC_ID_PCM_MULAW:
        payload_type = RTP_PT_ULAW;
        break;
    case CODEC_ID_PCM_ALAW:
        payload_type = RTP_PT_ALAW;
        break;
    case CODEC_ID_PCM_S16BE:
        if (codec->channels == 1) {
            payload_type = RTP_PT_S16BE_MONO;
        } else if (codec->channels == 2) {
            payload_type = RTP_PT_S16BE_STEREO;
        }
        break;
    case CODEC_ID_MP2:
Fabrice Bellard's avatar
Fabrice Bellard committed
175
    case CODEC_ID_MP3:
176 177 178 179 180 181 182 183
        payload_type = RTP_PT_MPEGAUDIO;
        break;
    case CODEC_ID_MJPEG:
        payload_type = RTP_PT_JPEG;
        break;
    case CODEC_ID_MPEG1VIDEO:
        payload_type = RTP_PT_MPEGVIDEO;
        break;
184 185 186
    case CODEC_ID_MPEG2TS:
        payload_type = RTP_PT_MPEG2TS;
        break;
187 188 189 190 191 192
    default:
        break;
    }
    return payload_type;
}

193
static inline uint32_t decode_be32(const uint8_t *p)
194 195 196 197
{
    return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
}

198
static inline uint64_t decode_be64(const uint8_t *p)
199
{
200
    return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
201 202
}

203
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
204 205 206 207
{
    if (buf[1] != 200)
        return -1;
    s->last_rtcp_ntp_time = decode_be64(buf + 8);
208 209
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
210 211 212 213 214
    s->last_rtcp_timestamp = decode_be32(buf + 16);
    return 0;
}

/**
215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) 
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
{
    RTPDemuxContext *s;

    s = av_mallocz(sizeof(RTPDemuxContext));
    if (!s)
        return NULL;
    s->payload_type = payload_type;
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->ic = s1;
    s->st = st;
    if (payload_type == RTP_PT_MPEG2TS) {
        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;
        }
    }
    return s;
}

/**
 * Parse an RTP or RTCP packet directly sent as a buffer. 
 * @param s RTP parse context.
244
 * @param pkt returned packet
245
 * @param buf input buffer or NULL to read the next packets
246
 * @param len buffer len
247 248
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow 
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
249
 */
250 251
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
                     const uint8_t *buf, int len)
252 253
{
    unsigned int ssrc, h;
254
    int payload_type, seq, delta_timestamp, ret;
255
    AVStream *st;
256
    uint32_t timestamp;
257
    
258 259 260 261 262 263 264 265 266 267 268 269 270 271 272
    if (!buf) {
        /* return the next packets, if any */
        if (s->read_buf_index >= s->read_buf_size)
            return -1;
        ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, 
                                  s->read_buf_size - s->read_buf_index);
        if (ret < 0)
            return -1;
        s->read_buf_index += ret;
        if (s->read_buf_index < s->read_buf_size)
            return 1;
        else
            return 0;
    }

273 274 275 276 277 278
    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
279
        rtcp_parse_packet(s, buf, len);
280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298
        return -1;
    }
    payload_type = buf[1] & 0x7f;
    seq  = (buf[2] << 8) | buf[3];
    timestamp = decode_be32(buf + 4);
    ssrc = decode_be32(buf + 8);
    
    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
#if defined(DEBUG) || 1
    if (seq != ((s->seq + 1) & 0xffff)) {
        printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
               payload_type, seq, ((s->seq + 1) & 0xffff));
    }
    s->seq = seq;
#endif
    len -= 12;
    buf += 12;
299 300 301 302 303 304

    st = s->st;
    if (!st) {
        /* specific MPEG2TS demux support */
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
        if (ret < 0)
305
            return -1;
306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325
        if (ret < len) {
            s->read_buf_size = len - ret;
            memcpy(s->buf, buf + ret, s->read_buf_size);
            s->read_buf_index = 0;
            return 1;
        }
    } else {
        switch(st->codec.codec_id) {
        case CODEC_ID_MP2:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
            h = decode_be32(buf);
            len -= 4;
            buf += 4;
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        case CODEC_ID_MPEG1VIDEO:
            /* better than nothing: skip mpeg audio RTP header */
326 327
            if (len <= 4)
                return -1;
328
            h = decode_be32(buf);
329 330
            buf += 4;
            len -= 4;
331 332 333 334 335 336 337 338 339 340 341 342 343 344
            if (h & (1 << 26)) {
                /* mpeg2 */
                if (len <= 4)
                    return -1;
                buf += 4;
                len -= 4;
            }
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        default:
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
345
        }
346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363
        
        switch(st->codec.codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;
                /* XXX: is it really necessary to unify the timestamp base ? */
                /* compute pts from timestamp with received ntp_time */
                delta_timestamp = timestamp - s->last_rtcp_timestamp;
                /* convert to 90 kHz without overflow */
                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
                addend = (addend * 5625) >> 14;
                pkt->pts = addend + delta_timestamp;
            }
            break;
        default:
            /* no timestamp info yet */
            break;
364
        }
365
        pkt->stream_index = s->st->index;
366 367 368 369
    }
    return 0;
}

370
void rtp_parse_close(RTPDemuxContext *s)
371
{
372 373
    if (s->payload_type == RTP_PT_MPEG2TS) {
        mpegts_parse_close(s->ts);
374
    }
375
    av_free(s);
376 377 378 379 380 381
}

/* rtp output */

static int rtp_write_header(AVFormatContext *s1)
{
382 383
    RTPDemuxContext *s = s1->priv_data;
    int payload_type, max_packet_size, n;
384 385 386 387 388 389 390 391
    AVStream *st;

    if (s1->nb_streams != 1)
        return -1;
    st = s1->streams[0];

    payload_type = rtp_get_payload_type(&st->codec);
    if (payload_type < 0)
392
        payload_type = RTP_PT_PRIVATE; /* private payload type */
393 394 395 396 397 398 399 400 401 402 403 404 405 406
    s->payload_type = payload_type;

    s->base_timestamp = random();
    s->timestamp = s->base_timestamp;
    s->ssrc = random();
    s->first_packet = 1;

    max_packet_size = url_fget_max_packet_size(&s1->pb);
    if (max_packet_size <= 12)
        return AVERROR_IO;
    s->max_payload_size = max_packet_size - 12;

    switch(st->codec.codec_id) {
    case CODEC_ID_MP2:
Fabrice Bellard's avatar
Fabrice Bellard committed
407
    case CODEC_ID_MP3:
408 409 410 411 412 413
        s->buf_ptr = s->buf + 4;
        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG1VIDEO:
        s->cur_timestamp = 0;
        break;
414 415 416 417 418 419 420
    case CODEC_ID_MPEG2TS:
        n = s->max_payload_size / TS_PACKET_SIZE;
        if (n < 1)
            n = 1;
        s->max_payload_size = n * TS_PACKET_SIZE;
        s->buf_ptr = s->buf;
        break;
421 422 423 424 425 426 427 428 429
    default:
        s->buf_ptr = s->buf;
        break;
    }

    return 0;
}

/* send an rtcp sender report packet */
430
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
431
{
432
    RTPDemuxContext *s = s1->priv_data;
433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448
#if defined(DEBUG)
    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
    put_byte(&s1->pb, (RTP_VERSION << 6));
    put_byte(&s1->pb, 200);
    put_be16(&s1->pb, 6); /* length in words - 1 */
    put_be32(&s1->pb, s->ssrc);
    put_be64(&s1->pb, ntp_time);
    put_be32(&s1->pb, s->timestamp);
    put_be32(&s1->pb, s->packet_count);
    put_be32(&s1->pb, s->octet_count);
    put_flush_packet(&s1->pb);
}

/* send an rtp packet. sequence number is incremented, but the caller
   must update the timestamp itself */
449
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
450
{
451
    RTPDemuxContext *s = s1->priv_data;
452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474

#ifdef DEBUG
    printf("rtp_send_data size=%d\n", len);
#endif

    /* build the RTP header */
    put_byte(&s1->pb, (RTP_VERSION << 6));
    put_byte(&s1->pb, s->payload_type & 0x7f);
    put_be16(&s1->pb, s->seq);
    put_be32(&s1->pb, s->timestamp);
    put_be32(&s1->pb, s->ssrc);
    
    put_buffer(&s1->pb, buf1, len);
    put_flush_packet(&s1->pb);
    
    s->seq++;
    s->octet_count += len;
    s->packet_count++;
}

/* send an integer number of samples and compute time stamp and fill
   the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
475
                             const uint8_t *buf1, int size, int sample_size)
476
{
477
    RTPDemuxContext *s = s1->priv_data;
478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507
    int len, max_packet_size, n;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)
        av_abort();
    while (size > 0) {
        len = (max_packet_size - (s->buf_ptr - s->buf));
        if (len > size)
            len = size;

        /* copy data */
        memcpy(s->buf_ptr, buf1, len);
        s->buf_ptr += len;
        buf1 += len;
        size -= len;
        n = (s->buf_ptr - s->buf);
        /* if buffer full, then send it */
        if (n >= max_packet_size) {
            rtp_send_data(s1, s->buf, n);
            s->buf_ptr = s->buf;
            /* update timestamp */
            s->timestamp += n / sample_size;
        }
    }
} 

/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
508
                               const uint8_t *buf1, int size)
509
{
510
    RTPDemuxContext *s = s1->priv_data;
511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563
    AVStream *st = s1->streams[0];
    int len, count, max_packet_size;

    max_packet_size = s->max_payload_size;

    /* test if we must flush because not enough space */
    len = (s->buf_ptr - s->buf);
    if ((len + size) > max_packet_size) {
        if (len > 4) {
            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
            s->buf_ptr = s->buf + 4;
            /* 90 KHz time stamp */
            s->timestamp = s->base_timestamp + 
                (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
        }
    }

    /* add the packet */
    if (size > max_packet_size) {
        /* big packet: fragment */
        count = 0;
        while (size > 0) {
            len = max_packet_size - 4;
            if (len > size)
                len = size;
            /* build fragmented packet */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = count >> 8;
            s->buf[3] = count;
            memcpy(s->buf + 4, buf1, len);
            rtp_send_data(s1, s->buf, len + 4);
            size -= len;
            buf1 += len;
            count += len;
        }
    } else {
        if (s->buf_ptr == s->buf + 4) {
            /* no fragmentation possible */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = 0;
            s->buf[3] = 0;
        }
        memcpy(s->buf_ptr, buf1, size);
        s->buf_ptr += size;
    }
    s->cur_timestamp += st->codec.frame_size;
}

/* NOTE: a single frame must be passed with sequence header if
   needed. XXX: use slices. */
static void rtp_send_mpegvideo(AVFormatContext *s1,
564
                               const uint8_t *buf1, int size)
565
{
566
    RTPDemuxContext *s = s1->priv_data;
567 568
    AVStream *st = s1->streams[0];
    int len, h, max_packet_size;
569
    uint8_t *q;
570 571 572 573 574 575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        /* XXX: more correct headers */
        h = 0;
        if (st->codec.sub_id == 2)
            h |= 1 << 26; /* mpeg 2 indicator */
        q = s->buf;
        *q++ = h >> 24;
        *q++ = h >> 16;
        *q++ = h >> 8;
        *q++ = h;

        if (st->codec.sub_id == 2) {
            h = 0;
            *q++ = h >> 24;
            *q++ = h >> 16;
            *q++ = h >> 8;
            *q++ = h;
        }
        
        len = max_packet_size - (q - s->buf);
        if (len > size)
            len = size;

        memcpy(q, buf1, len);
        q += len;

        /* 90 KHz time stamp */
        s->timestamp = s->base_timestamp + 
601
            av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
602 603 604 605 606 607 608 609
        rtp_send_data(s1, s->buf, q - s->buf);

        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

610
static void rtp_send_raw(AVFormatContext *s1,
611
                         const uint8_t *buf1, int size)
612
{
613
    RTPDemuxContext *s = s1->priv_data;
614 615 616 617 618 619 620 621 622 623 624 625
    AVStream *st = s1->streams[0];
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        len = max_packet_size;
        if (len > size)
            len = size;

        /* 90 KHz time stamp */
        s->timestamp = s->base_timestamp + 
626
            av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
627 628 629 630 631 632 633 634
        rtp_send_data(s1, buf1, len);

        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652 653 654 655 656 657 658
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    int len, out_len;

    while (size >= TS_PACKET_SIZE) {
        len = s->max_payload_size - (s->buf_ptr - s->buf);
        if (len > size)
            len = size;
        memcpy(s->buf_ptr, buf1, len);
        buf1 += len;
        size -= len;
        s->buf_ptr += len;
        
        out_len = s->buf_ptr - s->buf;
        if (out_len >= s->max_payload_size) {
            rtp_send_data(s1, s->buf, out_len);
            s->buf_ptr = s->buf;
        }
    }
}

659 660
/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, int stream_index,
661
                            const uint8_t *buf1, int size, int64_t pts)
662
{
663
    RTPDemuxContext *s = s1->priv_data;
664 665
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
666
    int64_t ntp_time;
667 668 669 670 671 672 673 674 675 676
    
#ifdef DEBUG
    printf("%d: write len=%d\n", stream_index, size);
#endif

    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / 
        RTCP_TX_RATIO_DEN;
    if (s->first_packet || rtcp_bytes >= 28) {
        /* compute NTP time */
677
        /* XXX: 90 kHz timestamp hardcoded */
678
        ntp_time = (pts << 28) / 5625;
679 680 681 682 683 684 685 686 687 688 689 690 691 692 693 694 695 696 697
        rtcp_send_sr(s1, ntp_time); 
        s->last_octet_count = s->octet_count;
        s->first_packet = 0;
    }

    switch(st->codec.codec_id) {
    case CODEC_ID_PCM_MULAW:
    case CODEC_ID_PCM_ALAW:
    case CODEC_ID_PCM_U8:
    case CODEC_ID_PCM_S8:
        rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
        break;
    case CODEC_ID_PCM_U16BE:
    case CODEC_ID_PCM_U16LE:
    case CODEC_ID_PCM_S16BE:
    case CODEC_ID_PCM_S16LE:
        rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
        break;
    case CODEC_ID_MP2:
Fabrice Bellard's avatar
Fabrice Bellard committed
698
    case CODEC_ID_MP3:
699 700 701 702 703
        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
        rtp_send_mpegvideo(s1, buf1, size);
        break;
704 705 706
    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, buf1, size);
        break;
707
    default:
708 709 710
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
711 712 713 714 715 716
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
717
    //    RTPDemuxContext *s = s1->priv_data;
718 719 720 721 722 723 724 725
    return 0;
}

AVOutputFormat rtp_mux = {
    "rtp",
    "RTP output format",
    NULL,
    NULL,
726
    sizeof(RTPDemuxContext),
727 728 729 730 731 732 733 734 735 736 737 738
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
    rtp_write_trailer,
};

int rtp_init(void)
{
    av_register_output_format(&rtp_mux);
    return 0;
}