rtp.c 20.6 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
 * 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;
        }
Fabrice Bellard's avatar
Fabrice Bellard committed
237 238 239 240 241 242 243 244 245 246 247 248
    } else {
        switch(st->codec.codec_id) {
        case CODEC_ID_MPEG1VIDEO:
        case CODEC_ID_MPEG2VIDEO:
        case CODEC_ID_MP2:
        case CODEC_ID_MP3:
        case CODEC_ID_MPEG4:
            st->need_parsing = 1;
            break;
        default:
            break;
        }
249 250 251 252 253 254 255
    }
    return s;
}

/**
 * Parse an RTP or RTCP packet directly sent as a buffer. 
 * @param s RTP parse context.
256
 * @param pkt returned packet
257
 * @param buf input buffer or NULL to read the next packets
258
 * @param len buffer len
259 260
 * @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).
261
 */
262 263
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
                     const uint8_t *buf, int len)
264 265
{
    unsigned int ssrc, h;
266
    int payload_type, seq, delta_timestamp, ret;
267
    AVStream *st;
268
    uint32_t timestamp;
269
    
270 271 272 273 274 275 276 277 278 279 280 281 282 283 284
    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;
    }

285 286 287 288 289 290
    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
291
        rtcp_parse_packet(s, buf, len);
292 293 294 295 296 297 298 299 300 301 302 303
        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)) {
304
        av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
305 306 307 308 309 310
               payload_type, seq, ((s->seq + 1) & 0xffff));
    }
    s->seq = seq;
#endif
    len -= 12;
    buf += 12;
311 312 313 314 315 316

    st = s->st;
    if (!st) {
        /* specific MPEG2TS demux support */
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
        if (ret < 0)
317
            return -1;
318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336
        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:
Fabrice Bellard's avatar
Fabrice Bellard committed
337
            /* better than nothing: skip mpeg video RTP header */
338 339
            if (len <= 4)
                return -1;
340
            h = decode_be32(buf);
341 342
            buf += 4;
            len -= 4;
343 344 345 346 347 348 349 350 351 352 353 354 355 356
            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;
357
        }
358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375
        
        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;
376
        }
377
        pkt->stream_index = s->st->index;
378 379 380 381
    }
    return 0;
}

382
void rtp_parse_close(RTPDemuxContext *s)
383
{
384 385
    if (s->payload_type == RTP_PT_MPEG2TS) {
        mpegts_parse_close(s->ts);
386
    }
387
    av_free(s);
388 389 390 391 392 393
}

/* rtp output */

static int rtp_write_header(AVFormatContext *s1)
{
394 395
    RTPDemuxContext *s = s1->priv_data;
    int payload_type, max_packet_size, n;
396 397 398 399 400 401 402 403
    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)
404
        payload_type = RTP_PT_PRIVATE; /* private payload type */
405 406 407 408 409 410 411 412 413 414 415 416 417 418
    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
419
    case CODEC_ID_MP3:
420 421 422 423 424 425
        s->buf_ptr = s->buf + 4;
        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG1VIDEO:
        s->cur_timestamp = 0;
        break;
426 427 428 429 430 431 432
    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;
433 434 435 436 437 438 439 440 441
    default:
        s->buf_ptr = s->buf;
        break;
    }

    return 0;
}

/* send an rtcp sender report packet */
442
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
443
{
444
    RTPDemuxContext *s = s1->priv_data;
445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460
#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 */
461
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
462
{
463
    RTPDemuxContext *s = s1->priv_data;
464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 483 484 485 486

#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,
487
                             const uint8_t *buf1, int size, int sample_size)
488
{
489
    RTPDemuxContext *s = s1->priv_data;
490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519
    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,
520
                               const uint8_t *buf1, int size)
521
{
522
    RTPDemuxContext *s = s1->priv_data;
523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573 574 575
    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,
576
                               const uint8_t *buf1, int size)
577
{
578
    RTPDemuxContext *s = s1->priv_data;
579 580
    AVStream *st = s1->streams[0];
    int len, h, max_packet_size;
581
    uint8_t *q;
582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612

    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 + 
613
            av_rescale((int64_t)s->cur_timestamp * st->codec.time_base.num, 90000, st->codec.time_base.den); //FIXME pass timestamps
614 615 616 617 618 619 620 621
        rtp_send_data(s1, s->buf, q - s->buf);

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

622
static void rtp_send_raw(AVFormatContext *s1,
623
                         const uint8_t *buf1, int size)
624
{
625
    RTPDemuxContext *s = s1->priv_data;
626 627 628 629 630 631 632 633 634 635 636 637
    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 + 
638
            av_rescale((int64_t)s->cur_timestamp * st->codec.time_base.num, 90000, st->codec.time_base.den); //FIXME pass timestamps
639 640 641 642 643 644 645 646
        rtp_send_data(s1, buf1, len);

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

647 648 649 650 651 652 653 654 655 656 657 658 659 660 661 662 663 664 665 666 667 668 669 670
/* 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;
        }
    }
}

671
/* write an RTP packet. 'buf1' must contain a single specific frame. */
672
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
673
{
674
    RTPDemuxContext *s = s1->priv_data;
675 676
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
677
    int64_t ntp_time;
678 679
    int size= pkt->size;
    uint8_t *buf1= pkt->data;
680 681
    
#ifdef DEBUG
682
    printf("%d: write len=%d\n", pkt->stream_index, size);
683 684 685 686 687 688 689
#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 */
690
        /* XXX: 90 kHz timestamp hardcoded */
691
        ntp_time = (pkt->pts << 28) / 5625;
692 693 694 695 696 697 698 699 700 701 702 703 704 705 706 707 708 709 710
        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
711
    case CODEC_ID_MP3:
712 713 714 715 716
        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
        rtp_send_mpegvideo(s1, buf1, size);
        break;
717 718 719
    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, buf1, size);
        break;
720
    default:
721 722 723
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
724 725 726 727 728 729
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
730
    //    RTPDemuxContext *s = s1->priv_data;
731 732 733 734 735 736 737 738
    return 0;
}

AVOutputFormat rtp_mux = {
    "rtp",
    "RTP output format",
    NULL,
    NULL,
739
    sizeof(RTPDemuxContext),
740 741 742 743 744 745 746 747 748 749 750 751
    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;
}