Statistics
| Branch: | Revision:

ffmpeg / libavformat / rtpdec.c @ 0369d2b0

History | View | Annotate | Download (17.6 KB)

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
9
 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
11
 *
12
 * 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.
16
 *
17
 * You should have received a copy of the GNU Lesser General Public
18
 * License along with FFmpeg; if not, write to the Free Software
19
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20
 */
21

    
22
/* needed for gethostname() */
23
#define _XOPEN_SOURCE 500
24

    
25
#include "libavcodec/bitstream.h"
26
#include "avformat.h"
27
#include "mpegts.h"
28

    
29
#include <unistd.h>
30
#include "network.h"
31

    
32
#include "rtp_internal.h"
33
#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
40
         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
43
         'url_open_dyn_packet_buf')
44
*/
45

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

    
49
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;
55
    RTPFirstDynamicPayloadHandler= handler;
56
}
57

    
58
void av_register_rtp_dynamic_payload_handlers(void)
59
{
60
    ff_register_dynamic_payload_handler(&mp4v_es_handler);
61
    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;
72
    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;
97
    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;
110
    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++;
123
                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, rtp_payload_data_t *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
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
315
{
316
    int au_headers_length, au_header_size, i;
317
    GetBitContext getbitcontext;
318
    rtp_payload_data_t *infos;
319

    
320
    infos = s->rtp_payload_data;
321

    
322
    if (infos == NULL)
323
        return -1;
324

    
325
    /* decode the first 2 bytes where the AUHeader sections are stored
326
       length in bits */
327
    au_headers_length = AV_RB16(buf);
328

    
329
    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
330
      return -1;
331

    
332
    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
333

    
334
    /* skip AU headers length section (2 bytes) */
335
    buf += 2;
336

    
337
    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
338

    
339
    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
340
    au_header_size = infos->sizelength + infos->indexlength;
341
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
342
        return -1;
343

    
344
    infos->nb_au_headers = au_headers_length / au_header_size;
345
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
346

    
347
    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
348
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
349
       but does when sending the whole as one big packet...  */
350
    infos->au_headers[0].size = 0;
351
    infos->au_headers[0].index = 0;
352
    for (i = 0; i < infos->nb_au_headers; ++i) {
353
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
354
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
355
    }
356

    
357
    infos->nb_au_headers = 1;
358

    
359
    return 0;
360
}
361

    
362
/**
363
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
364
 */
365
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
366
{
367
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
368
        int64_t addend;
369
        int delta_timestamp;
370

    
371
        /* compute pts from timestamp with received ntp_time */
372
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
373
        /* convert to the PTS timebase */
374
        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);
375
        pkt->pts = addend + delta_timestamp;
376
    }
377
    pkt->stream_index = s->st->index;
378
}
379

    
380
/**
381
 * Parse an RTP or RTCP packet directly sent as a buffer.
382
 * @param s RTP parse context.
383
 * @param pkt returned packet
384
 * @param buf input buffer or NULL to read the next packets
385
 * @param len buffer len
386
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
387
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
388
 */
389
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
390
                     const uint8_t *buf, int len)
391
{
392
    unsigned int ssrc, h;
393
    int payload_type, seq, ret, flags = 0;
394
    AVStream *st;
395
    uint32_t timestamp;
396
    int rv= 0;
397

    
398
    if (!buf) {
399
        /* return the next packets, if any */
400
        if(s->st && s->parse_packet) {
401
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
402
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0, flags);
403
            finalize_packet(s, pkt, timestamp);
404
            return rv;
405
        } else {
406
            // TODO: Move to a dynamic packet handler (like above)
407
            if (s->read_buf_index >= s->read_buf_size)
408
                return -1;
409
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
410
                                      s->read_buf_size - s->read_buf_index);
411
            if (ret < 0)
412
                return -1;
413
            s->read_buf_index += ret;
414
            if (s->read_buf_index < s->read_buf_size)
415
                return 1;
416
            else
417
                return 0;
418
        }
419
    }
420

    
421
    if (len < 12)
422
        return -1;
423

    
424
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
425
        return -1;
426
    if (buf[1] >= 200 && buf[1] <= 204) {
427
        rtcp_parse_packet(s, buf, len);
428
        return -1;
429
    }
430
    payload_type = buf[1] & 0x7f;
431
    seq  = AV_RB16(buf + 2);
432
    timestamp = AV_RB32(buf + 4);
433
    ssrc = AV_RB32(buf + 8);
434
    /* store the ssrc in the RTPDemuxContext */
435
    s->ssrc = ssrc;
436

    
437
    /* NOTE: we can handle only one payload type */
438
    if (s->payload_type != payload_type)
439
        return -1;
440

    
441
    st = s->st;
442
    // only do something with this if all the rtp checks pass...
443
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
444
    {
445
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
446
               payload_type, seq, ((s->seq + 1) & 0xffff));
447
        return -1;
448
    }
449

    
450
    s->seq = seq;
451
    len -= 12;
452
    buf += 12;
453

    
454
    if (!st) {
455
        /* specific MPEG2TS demux support */
456
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
457
        if (ret < 0)
458
            return -1;
459
        if (ret < len) {
460
            s->read_buf_size = len - ret;
461
            memcpy(s->buf, buf + ret, s->read_buf_size);
462
            s->read_buf_index = 0;
463
            return 1;
464
        }
465
    } else if (s->parse_packet) {
466
        rv = s->parse_packet(s, pkt, &timestamp, buf, len, flags);
467
    } else {
468
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
469
        switch(st->codec->codec_id) {
470
        case CODEC_ID_MP2:
471
            /* better than nothing: skip mpeg audio RTP header */
472
            if (len <= 4)
473
                return -1;
474
            h = AV_RB32(buf);
475
            len -= 4;
476
            buf += 4;
477
            av_new_packet(pkt, len);
478
            memcpy(pkt->data, buf, len);
479
            break;
480
        case CODEC_ID_MPEG1VIDEO:
481
        case CODEC_ID_MPEG2VIDEO:
482
            /* better than nothing: skip mpeg video RTP header */
483
            if (len <= 4)
484
                return -1;
485
            h = AV_RB32(buf);
486
            buf += 4;
487
            len -= 4;
488
            if (h & (1 << 26)) {
489
                /* mpeg2 */
490
                if (len <= 4)
491
                    return -1;
492
                buf += 4;
493
                len -= 4;
494
            }
495
            av_new_packet(pkt, len);
496
            memcpy(pkt->data, buf, len);
497
            break;
498
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
499
            // timestamps.
500
            // TODO: Put this into a dynamic packet handler...
501
        case CODEC_ID_AAC:
502
            if (rtp_parse_mp4_au(s, buf))
503
                return -1;
504
            {
505
                rtp_payload_data_t *infos = s->rtp_payload_data;
506
                if (infos == NULL)
507
                    return -1;
508
                buf += infos->au_headers_length_bytes + 2;
509
                len -= infos->au_headers_length_bytes + 2;
510

    
511
                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
512
                    one au_header */
513
                av_new_packet(pkt, infos->au_headers[0].size);
514
                memcpy(pkt->data, buf, infos->au_headers[0].size);
515
                buf += infos->au_headers[0].size;
516
                len -= infos->au_headers[0].size;
517
            }
518
            s->read_buf_size = len;
519
            rv= 0;
520
            break;
521
        default:
522
            av_new_packet(pkt, len);
523
            memcpy(pkt->data, buf, len);
524
            break;
525
        }
526

    
527
        // now perform timestamp things....
528
        finalize_packet(s, pkt, timestamp);
529
    }
530
    return rv;
531
}
532

    
533
void rtp_parse_close(RTPDemuxContext *s)
534
{
535
    // TODO: fold this into the protocol specific data fields.
536
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
537
        mpegts_parse_close(s->ts);
538
    }
539
    av_free(s);
540
}