rtpdec.c 17.9 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 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 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404
/*
 * RTP input format
 * Copyright (c) 2002 Fabrice Bellard.
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
#include "avformat.h"
#include "mpegts.h"
#include "bitstream.h"

#include <unistd.h>
#include "network.h"

#include "rtp_internal.h"
#include "rtp_h264.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')
*/

/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;

static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};

static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
    handler->next= RTPFirstDynamicPayloadHandler;
    RTPFirstDynamicPayloadHandler= handler;
}

void av_register_rtp_dynamic_payload_handlers(void)
{
    register_dynamic_payload_handler(&mp4v_es_handler);
    register_dynamic_payload_handler(&mpeg4_generic_handler);
    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
}

static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
    if (buf[1] != 200)
        return -1;
    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
    s->last_rtcp_timestamp = AV_RB32(buf + 16);
    return 0;
}

#define RTP_SEQ_MOD (1<<16)

/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
    memset(s, 0, sizeof(RTPStatistics));
    s->max_seq= base_sequence;
    s->probation= 1;
}

/**
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
    s->max_seq= seq;
    s->cycles= 0;
    s->base_seq= seq -1;
    s->bad_seq= RTP_SEQ_MOD + 1;
    s->received= 0;
    s->expected_prior= 0;
    s->received_prior= 0;
    s->jitter= 0;
    s->transit= 0;
}

/**
* returns 1 if we should handle this packet.
*/
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
    uint16_t udelta= seq - s->max_seq;
    const int MAX_DROPOUT= 3000;
    const int MAX_MISORDER = 100;
    const int MIN_SEQUENTIAL = 2;

    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
    if(s->probation)
    {
        if(seq==s->max_seq + 1) {
            s->probation--;
            s->max_seq= seq;
            if(s->probation==0) {
                rtp_init_sequence(s, seq);
                s->received++;
                return 1;
            }
        } else {
            s->probation= MIN_SEQUENTIAL - 1;
            s->max_seq = seq;
        }
    } else if (udelta < MAX_DROPOUT) {
        // in order, with permissible gap
        if(seq < s->max_seq) {
            //sequence number wrapped; count antother 64k cycles
            s->cycles += RTP_SEQ_MOD;
        }
        s->max_seq= seq;
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
        // sequence made a large jump...
        if(seq==s->bad_seq) {
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
            rtp_init_sequence(s, seq);
        } else {
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
            return 0;
        }
    } else {
        // duplicate or reordered packet...
    }
    s->received++;
    return 1;
}

#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
* never change.  I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
    uint32_t transit= arrival_timestamp - sent_timestamp;
    int d;
    s->transit= transit;
    d= FFABS(transit - s->transit);
    s->jitter += d - ((s->jitter + 8)>>4);
}
#endif

int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
    ByteIOContext *pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;
    RTPStatistics *stats= &s->statistics;
    uint32_t lost;
    uint32_t extended_max;
    uint32_t expected_interval;
    uint32_t received_interval;
    uint32_t lost_interval;
    uint32_t expected;
    uint32_t fraction;
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?

    if (!s->rtp_ctx || (count < 1))
        return -1;

    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    s->octet_count += count;
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
    if (rtcp_bytes < 28)
        return -1;
    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)
        return -1;

    // Receiver Report
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(pb, 201);
    put_be16(pb, 7); /* length in words - 1 */
    put_be32(pb, s->ssrc); // our own SSRC
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
    // some placeholders we should really fill...
    // RFC 1889/p64
    extended_max= stats->cycles + stats->max_seq;
    expected= extended_max - stats->base_seq + 1;
    lost= expected - stats->received;
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
    expected_interval= expected - stats->expected_prior;
    stats->expected_prior= expected;
    received_interval= stats->received - stats->received_prior;
    stats->received_prior= stats->received;
    lost_interval= expected_interval - received_interval;
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
    else fraction = (lost_interval<<8)/expected_interval;

    fraction= (fraction<<24) | lost;

    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
    put_be32(pb, extended_max); /* max sequence received */
    put_be32(pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
    {
        put_be32(pb, 0); /* last SR timestamp */
        put_be32(pb, 0); /* delay since last SR */
    } else {
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;

        put_be32(pb, middle_32_bits); /* last SR timestamp */
        put_be32(pb, delay_since_last); /* delay since last SR */
    }

    // CNAME
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(pb, 202);
    len = strlen(s->hostname);
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
    put_be32(pb, s->ssrc);
    put_byte(pb, 0x01);
    put_byte(pb, len);
    put_buffer(pb, s->hostname, len);
    // padding
    for (len = (6 + len) % 4; len % 4; len++) {
        put_byte(pb, 0);
    }

    put_flush_packet(pb);
    len = url_close_dyn_buf(pb, &buf);
    if ((len > 0) && buf) {
        int result;
#if defined(DEBUG)
        printf("sending %d bytes of RR\n", len);
#endif
        result= url_write(s->rtp_ctx, buf, len);
#if defined(DEBUG)
        printf("result from url_write: %d\n", result);
#endif
        av_free(buf);
    }
    return 0;
}

/**
 * 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)
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
    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;
    s->rtp_payload_data = rtp_payload_data;
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;
        }
    } 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:
        case CODEC_ID_H264:
            st->need_parsing = AVSTREAM_PARSE_FULL;
            break;
        default:
            break;
        }
    }
    // needed to send back RTCP RR in RTSP sessions
    s->rtp_ctx = rtpc;
    gethostname(s->hostname, sizeof(s->hostname));
    return s;
}

static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
    int au_headers_length, au_header_size, i;
    GetBitContext getbitcontext;
    rtp_payload_data_t *infos;

    infos = s->rtp_payload_data;

    if (infos == NULL)
        return -1;

    /* decode the first 2 bytes where are stored the AUHeader sections
       length in bits */
    au_headers_length = AV_RB16(buf);

    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
      return -1;

    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;

    /* skip AU headers length section (2 bytes) */
    buf += 2;

    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);

    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
    au_header_size = infos->sizelength + infos->indexlength;
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
        return -1;

    infos->nb_au_headers = au_headers_length / au_header_size;
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);

    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
       but does when sending the whole as one big packet...  */
    infos->au_headers[0].size = 0;
    infos->au_headers[0].index = 0;
    for (i = 0; i < infos->nb_au_headers; ++i) {
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
    }

    infos->nb_au_headers = 1;

    return 0;
}

/**
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
 */
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
    switch(s->st->codec->codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
        case CODEC_ID_MPEG2VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;

                int delta_timestamp;
                /* 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;
        case CODEC_ID_AAC:
        case CODEC_ID_H264:
        case CODEC_ID_MPEG4:
            pkt->pts = timestamp;
            break;
        default:
            /* no timestamp info yet */
            break;
    }
    pkt->stream_index = s->st->index;
}

/**
 * Parse an RTP or RTCP packet directly sent as a buffer.
 * @param s RTP parse context.
 * @param pkt returned packet
 * @param buf input buffer or NULL to read the next packets
 * @param len buffer len
 * @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).
 */
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
                     const uint8_t *buf, int len)
{
    unsigned int ssrc, h;
405
    int payload_type, seq, ret, flags = 0;
406 407 408 409 410 411 412 413
    AVStream *st;
    uint32_t timestamp;
    int rv= 0;

    if (!buf) {
        /* return the next packets, if any */
        if(s->st && s->parse_packet) {
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
414
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0, flags);
415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476
            finalize_packet(s, pkt, timestamp);
            return rv;
        } else {
            // TODO: Move to a dynamic packet handler (like above)
            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;
        }
    }

    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
        rtcp_parse_packet(s, buf, len);
        return -1;
    }
    payload_type = buf[1] & 0x7f;
    seq  = AV_RB16(buf + 2);
    timestamp = AV_RB32(buf + 4);
    ssrc = AV_RB32(buf + 8);
    /* store the ssrc in the RTPDemuxContext */
    s->ssrc = ssrc;

    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;

    st = s->st;
    // only do something with this if all the rtp checks pass...
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
    {
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
               payload_type, seq, ((s->seq + 1) & 0xffff));
        return -1;
    }

    s->seq = seq;
    len -= 12;
    buf += 12;

    if (!st) {
        /* specific MPEG2TS demux support */
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
        if (ret < 0)
            return -1;
        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;
        }
477
    } else if (s->parse_packet) {
478
        rv = s->parse_packet(s, pkt, &timestamp, buf, len, flags);
479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533
    } else {
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
        switch(st->codec->codec_id) {
        case CODEC_ID_MP2:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
            h = AV_RB32(buf);
            len -= 4;
            buf += 4;
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        case CODEC_ID_MPEG1VIDEO:
        case CODEC_ID_MPEG2VIDEO:
            /* better than nothing: skip mpeg video RTP header */
            if (len <= 4)
                return -1;
            h = AV_RB32(buf);
            buf += 4;
            len -= 4;
            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;
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
            // timestamps.
            // TODO: Put this into a dynamic packet handler...
        case CODEC_ID_AAC:
            if (rtp_parse_mp4_au(s, buf))
                return -1;
            {
                rtp_payload_data_t *infos = s->rtp_payload_data;
                if (infos == NULL)
                    return -1;
                buf += infos->au_headers_length_bytes + 2;
                len -= infos->au_headers_length_bytes + 2;

                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
                    one au_header */
                av_new_packet(pkt, infos->au_headers[0].size);
                memcpy(pkt->data, buf, infos->au_headers[0].size);
                buf += infos->au_headers[0].size;
                len -= infos->au_headers[0].size;
            }
            s->read_buf_size = len;
            rv= 0;
            break;
        default:
534 535
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552
            break;
        }

        // now perform timestamp things....
        finalize_packet(s, pkt, timestamp);
    }
    return rv;
}

void rtp_parse_close(RTPDemuxContext *s)
{
    // TODO: fold this into the protocol specific data fields.
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
        mpegts_parse_close(s->ts);
    }
    av_free(s);
}