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ffmpeg / libavformat / rtpdec.c @ eafb17d1

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1
/*
2
 * RTP input format
3
 * Copyright (c) 2002 Fabrice Bellard
4
 *
5
 * This file is part of FFmpeg.
6
 *
7
 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
13
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15
 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
/* needed for gethostname() */
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#define _XOPEN_SOURCE 600
24

    
25
#include "libavcodec/bitstream.h"
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#include "avformat.h"
27
#include "mpegts.h"
28

    
29
#include <unistd.h>
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#include "network.h"
31

    
32
#include "rtpdec.h"
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#include "rtp_h264.h"
34

    
35
//#define DEBUG
36

    
37
/* TODO: - add RTCP statistics reporting (should be optional).
38

39
         - add support for h263/mpeg4 packetized output : IDEA: send a
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         buffer to 'rtp_write_packet' contains all the packets for ONE
41
         frame. Each packet should have a four byte header containing
42
         the length in big endian format (same trick as
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         'url_open_dyn_packet_buf')
44
*/
45

    
46
/* statistics functions */
47
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
48

    
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static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
50
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
51

    
52
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
53
{
54
    handler->next= RTPFirstDynamicPayloadHandler;
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    RTPFirstDynamicPayloadHandler= handler;
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}
57

    
58
void av_register_rtp_dynamic_payload_handlers(void)
59
{
60
    ff_register_dynamic_payload_handler(&mp4v_es_handler);
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    ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
62
    ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
63
}
64

    
65
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
66
{
67
    if (buf[1] != 200)
68
        return -1;
69
    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
70
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
71
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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    s->last_rtcp_timestamp = AV_RB32(buf + 16);
73
    return 0;
74
}
75

    
76
#define RTP_SEQ_MOD (1<<16)
77

    
78
/**
79
* called on parse open packet
80
*/
81
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
82
{
83
    memset(s, 0, sizeof(RTPStatistics));
84
    s->max_seq= base_sequence;
85
    s->probation= 1;
86
}
87

    
88
/**
89
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
90
*/
91
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
92
{
93
    s->max_seq= seq;
94
    s->cycles= 0;
95
    s->base_seq= seq -1;
96
    s->bad_seq= RTP_SEQ_MOD + 1;
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    s->received= 0;
98
    s->expected_prior= 0;
99
    s->received_prior= 0;
100
    s->jitter= 0;
101
    s->transit= 0;
102
}
103

    
104
/**
105
* returns 1 if we should handle this packet.
106
*/
107
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
108
{
109
    uint16_t udelta= seq - s->max_seq;
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    const int MAX_DROPOUT= 3000;
111
    const int MAX_MISORDER = 100;
112
    const int MIN_SEQUENTIAL = 2;
113

    
114
    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
115
    if(s->probation)
116
    {
117
        if(seq==s->max_seq + 1) {
118
            s->probation--;
119
            s->max_seq= seq;
120
            if(s->probation==0) {
121
                rtp_init_sequence(s, seq);
122
                s->received++;
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                return 1;
124
            }
125
        } else {
126
            s->probation= MIN_SEQUENTIAL - 1;
127
            s->max_seq = seq;
128
        }
129
    } else if (udelta < MAX_DROPOUT) {
130
        // in order, with permissible gap
131
        if(seq < s->max_seq) {
132
            //sequence number wrapped; count antother 64k cycles
133
            s->cycles += RTP_SEQ_MOD;
134
        }
135
        s->max_seq= seq;
136
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
137
        // sequence made a large jump...
138
        if(seq==s->bad_seq) {
139
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
140
            rtp_init_sequence(s, seq);
141
        } else {
142
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
143
            return 0;
144
        }
145
    } else {
146
        // duplicate or reordered packet...
147
    }
148
    s->received++;
149
    return 1;
150
}
151

    
152
#if 0
153
/**
154
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
155
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
156
* never change.  I left this in in case someone else can see a way. (rdm)
157
*/
158
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
159
{
160
    uint32_t transit= arrival_timestamp - sent_timestamp;
161
    int d;
162
    s->transit= transit;
163
    d= FFABS(transit - s->transit);
164
    s->jitter += d - ((s->jitter + 8)>>4);
165
}
166
#endif
167

    
168
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
169
{
170
    ByteIOContext *pb;
171
    uint8_t *buf;
172
    int len;
173
    int rtcp_bytes;
174
    RTPStatistics *stats= &s->statistics;
175
    uint32_t lost;
176
    uint32_t extended_max;
177
    uint32_t expected_interval;
178
    uint32_t received_interval;
179
    uint32_t lost_interval;
180
    uint32_t expected;
181
    uint32_t fraction;
182
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
183

    
184
    if (!s->rtp_ctx || (count < 1))
185
        return -1;
186

    
187
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
188
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
189
    s->octet_count += count;
190
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
191
        RTCP_TX_RATIO_DEN;
192
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
193
    if (rtcp_bytes < 28)
194
        return -1;
195
    s->last_octet_count = s->octet_count;
196

    
197
    if (url_open_dyn_buf(&pb) < 0)
198
        return -1;
199

    
200
    // Receiver Report
201
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
202
    put_byte(pb, 201);
203
    put_be16(pb, 7); /* length in words - 1 */
204
    put_be32(pb, s->ssrc); // our own SSRC
205
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
206
    // some placeholders we should really fill...
207
    // RFC 1889/p64
208
    extended_max= stats->cycles + stats->max_seq;
209
    expected= extended_max - stats->base_seq + 1;
210
    lost= expected - stats->received;
211
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
212
    expected_interval= expected - stats->expected_prior;
213
    stats->expected_prior= expected;
214
    received_interval= stats->received - stats->received_prior;
215
    stats->received_prior= stats->received;
216
    lost_interval= expected_interval - received_interval;
217
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
218
    else fraction = (lost_interval<<8)/expected_interval;
219

    
220
    fraction= (fraction<<24) | lost;
221

    
222
    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
223
    put_be32(pb, extended_max); /* max sequence received */
224
    put_be32(pb, stats->jitter>>4); /* jitter */
225

    
226
    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
227
    {
228
        put_be32(pb, 0); /* last SR timestamp */
229
        put_be32(pb, 0); /* delay since last SR */
230
    } else {
231
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
232
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
233

    
234
        put_be32(pb, middle_32_bits); /* last SR timestamp */
235
        put_be32(pb, delay_since_last); /* delay since last SR */
236
    }
237

    
238
    // CNAME
239
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
240
    put_byte(pb, 202);
241
    len = strlen(s->hostname);
242
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
243
    put_be32(pb, s->ssrc);
244
    put_byte(pb, 0x01);
245
    put_byte(pb, len);
246
    put_buffer(pb, s->hostname, len);
247
    // padding
248
    for (len = (6 + len) % 4; len % 4; len++) {
249
        put_byte(pb, 0);
250
    }
251

    
252
    put_flush_packet(pb);
253
    len = url_close_dyn_buf(pb, &buf);
254
    if ((len > 0) && buf) {
255
        int result;
256
        dprintf(s->ic, "sending %d bytes of RR\n", len);
257
        result= url_write(s->rtp_ctx, buf, len);
258
        dprintf(s->ic, "result from url_write: %d\n", result);
259
        av_free(buf);
260
    }
261
    return 0;
262
}
263

    
264
/**
265
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
266
 * MPEG2TS streams to indicate that they should be demuxed inside the
267
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
268
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
269
 */
270
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
271
{
272
    RTPDemuxContext *s;
273

    
274
    s = av_mallocz(sizeof(RTPDemuxContext));
275
    if (!s)
276
        return NULL;
277
    s->payload_type = payload_type;
278
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
279
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
280
    s->ic = s1;
281
    s->st = st;
282
    s->rtp_payload_data = rtp_payload_data;
283
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
284
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
285
        s->ts = mpegts_parse_open(s->ic);
286
        if (s->ts == NULL) {
287
            av_free(s);
288
            return NULL;
289
        }
290
    } else {
291
        av_set_pts_info(st, 32, 1, 90000);
292
        switch(st->codec->codec_id) {
293
        case CODEC_ID_MPEG1VIDEO:
294
        case CODEC_ID_MPEG2VIDEO:
295
        case CODEC_ID_MP2:
296
        case CODEC_ID_MP3:
297
        case CODEC_ID_MPEG4:
298
        case CODEC_ID_H264:
299
            st->need_parsing = AVSTREAM_PARSE_FULL;
300
            break;
301
        default:
302
            if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
303
                av_set_pts_info(st, 32, 1, st->codec->sample_rate);
304
            }
305
            break;
306
        }
307
    }
308
    // needed to send back RTCP RR in RTSP sessions
309
    s->rtp_ctx = rtpc;
310
    gethostname(s->hostname, sizeof(s->hostname));
311
    return s;
312
}
313

    
314
void
315
rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
316
                               RTPDynamicProtocolHandler *handler)
317
{
318
    s->dynamic_protocol_context = ctx;
319
    s->parse_packet = handler->parse_packet;
320
}
321

    
322
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
323
{
324
    int au_headers_length, au_header_size, i;
325
    GetBitContext getbitcontext;
326
    RTPPayloadData *infos;
327

    
328
    infos = s->rtp_payload_data;
329

    
330
    if (infos == NULL)
331
        return -1;
332

    
333
    /* decode the first 2 bytes where the AUHeader sections are stored
334
       length in bits */
335
    au_headers_length = AV_RB16(buf);
336

    
337
    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
338
      return -1;
339

    
340
    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
341

    
342
    /* skip AU headers length section (2 bytes) */
343
    buf += 2;
344

    
345
    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
346

    
347
    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
348
    au_header_size = infos->sizelength + infos->indexlength;
349
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
350
        return -1;
351

    
352
    infos->nb_au_headers = au_headers_length / au_header_size;
353
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
354

    
355
    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
356
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
357
       but does when sending the whole as one big packet...  */
358
    infos->au_headers[0].size = 0;
359
    infos->au_headers[0].index = 0;
360
    for (i = 0; i < infos->nb_au_headers; ++i) {
361
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
362
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
363
    }
364

    
365
    infos->nb_au_headers = 1;
366

    
367
    return 0;
368
}
369

    
370
/**
371
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
372
 */
373
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
374
{
375
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
376
        int64_t addend;
377
        int delta_timestamp;
378

    
379
        /* compute pts from timestamp with received ntp_time */
380
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
381
        /* convert to the PTS timebase */
382
        addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
383
        pkt->pts = addend + delta_timestamp;
384
    }
385
}
386

    
387
/**
388
 * Parse an RTP or RTCP packet directly sent as a buffer.
389
 * @param s RTP parse context.
390
 * @param pkt returned packet
391
 * @param buf input buffer or NULL to read the next packets
392
 * @param len buffer len
393
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
394
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
395
 */
396
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
397
                     const uint8_t *buf, int len)
398
{
399
    unsigned int ssrc, h;
400
    int payload_type, seq, ret, flags = 0;
401
    AVStream *st;
402
    uint32_t timestamp;
403
    int rv= 0;
404

    
405
    if (!buf) {
406
        /* return the next packets, if any */
407
        if(s->st && s->parse_packet) {
408
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
409
            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
410
                                s->st, pkt, &timestamp, NULL, 0, flags);
411
            finalize_packet(s, pkt, timestamp);
412
            return rv;
413
        } else {
414
            // TODO: Move to a dynamic packet handler (like above)
415
            if (s->read_buf_index >= s->read_buf_size)
416
                return -1;
417
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
418
                                      s->read_buf_size - s->read_buf_index);
419
            if (ret < 0)
420
                return -1;
421
            s->read_buf_index += ret;
422
            if (s->read_buf_index < s->read_buf_size)
423
                return 1;
424
            else
425
                return 0;
426
        }
427
    }
428

    
429
    if (len < 12)
430
        return -1;
431

    
432
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
433
        return -1;
434
    if (buf[1] >= 200 && buf[1] <= 204) {
435
        rtcp_parse_packet(s, buf, len);
436
        return -1;
437
    }
438
    payload_type = buf[1] & 0x7f;
439
    if (buf[1] & 0x80)
440
        flags |= RTP_FLAG_MARKER;
441
    seq  = AV_RB16(buf + 2);
442
    timestamp = AV_RB32(buf + 4);
443
    ssrc = AV_RB32(buf + 8);
444
    /* store the ssrc in the RTPDemuxContext */
445
    s->ssrc = ssrc;
446

    
447
    /* NOTE: we can handle only one payload type */
448
    if (s->payload_type != payload_type)
449
        return -1;
450

    
451
    st = s->st;
452
    // only do something with this if all the rtp checks pass...
453
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
454
    {
455
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
456
               payload_type, seq, ((s->seq + 1) & 0xffff));
457
        return -1;
458
    }
459

    
460
    s->seq = seq;
461
    len -= 12;
462
    buf += 12;
463

    
464
    if (!st) {
465
        /* specific MPEG2TS demux support */
466
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
467
        if (ret < 0)
468
            return -1;
469
        if (ret < len) {
470
            s->read_buf_size = len - ret;
471
            memcpy(s->buf, buf + ret, s->read_buf_size);
472
            s->read_buf_index = 0;
473
            return 1;
474
        }
475
        return 0;
476
    } else if (s->parse_packet) {
477
        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
478
                             s->st, pkt, &timestamp, buf, len, flags);
479
    } else {
480
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
481
        switch(st->codec->codec_id) {
482
        case CODEC_ID_MP2:
483
            /* better than nothing: skip mpeg audio RTP header */
484
            if (len <= 4)
485
                return -1;
486
            h = AV_RB32(buf);
487
            len -= 4;
488
            buf += 4;
489
            av_new_packet(pkt, len);
490
            memcpy(pkt->data, buf, len);
491
            break;
492
        case CODEC_ID_MPEG1VIDEO:
493
        case CODEC_ID_MPEG2VIDEO:
494
            /* better than nothing: skip mpeg video RTP header */
495
            if (len <= 4)
496
                return -1;
497
            h = AV_RB32(buf);
498
            buf += 4;
499
            len -= 4;
500
            if (h & (1 << 26)) {
501
                /* mpeg2 */
502
                if (len <= 4)
503
                    return -1;
504
                buf += 4;
505
                len -= 4;
506
            }
507
            av_new_packet(pkt, len);
508
            memcpy(pkt->data, buf, len);
509
            break;
510
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
511
            // timestamps.
512
            // TODO: Put this into a dynamic packet handler...
513
        case CODEC_ID_AAC:
514
            if (rtp_parse_mp4_au(s, buf))
515
                return -1;
516
            {
517
                RTPPayloadData *infos = s->rtp_payload_data;
518
                if (infos == NULL)
519
                    return -1;
520
                buf += infos->au_headers_length_bytes + 2;
521
                len -= infos->au_headers_length_bytes + 2;
522

    
523
                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
524
                    one au_header */
525
                av_new_packet(pkt, infos->au_headers[0].size);
526
                memcpy(pkt->data, buf, infos->au_headers[0].size);
527
                buf += infos->au_headers[0].size;
528
                len -= infos->au_headers[0].size;
529
            }
530
            s->read_buf_size = len;
531
            rv= 0;
532
            break;
533
        default:
534
            av_new_packet(pkt, len);
535
            memcpy(pkt->data, buf, len);
536
            break;
537
        }
538

    
539
        pkt->stream_index = st->index;
540
    }
541

    
542
    // now perform timestamp things....
543
    finalize_packet(s, pkt, timestamp);
544

    
545
    return rv;
546
}
547

    
548
void rtp_parse_close(RTPDemuxContext *s)
549
{
550
    // TODO: fold this into the protocol specific data fields.
551
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
552
        mpegts_parse_close(s->ts);
553
    }
554
    av_free(s);
555
}