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ffmpeg / libavformat / rtpdec.c @ e9fce261

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1
/*
2
 * RTP input format
3
 * Copyright (c) 2002 Fabrice Bellard
4
 *
5
 * This file is part of FFmpeg.
6
 *
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 * 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

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

    
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#include "libavcodec/bitstream.h"
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#include "avformat.h"
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#include "mpegts.h"
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#include <unistd.h>
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#include "network.h"
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#include "rtpdec.h"
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#include "rtp_asf.h"
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#include "rtp_h264.h"
35

    
36
//#define DEBUG
37

    
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/* TODO: - add RTCP statistics reporting (should be optional).
39

40
         - 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
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         frame. Each packet should have a four byte header containing
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         the length in big endian format (same trick as
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         'url_open_dyn_packet_buf')
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*/
46

    
47
/* statistics functions */
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RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
49

    
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static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
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static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
52

    
53
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
54
{
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    handler->next= RTPFirstDynamicPayloadHandler;
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    RTPFirstDynamicPayloadHandler= handler;
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}
58

    
59
void av_register_rtp_dynamic_payload_handlers(void)
60
{
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    ff_register_dynamic_payload_handler(&mp4v_es_handler);
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    ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
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    ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
64

    
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    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
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    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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}
68

    
69
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
70
{
71
    if (buf[1] != 200)
72
        return -1;
73
    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
74
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
75
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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    s->last_rtcp_timestamp = AV_RB32(buf + 16);
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    return 0;
78
}
79

    
80
#define RTP_SEQ_MOD (1<<16)
81

    
82
/**
83
* called on parse open packet
84
*/
85
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
86
{
87
    memset(s, 0, sizeof(RTPStatistics));
88
    s->max_seq= base_sequence;
89
    s->probation= 1;
90
}
91

    
92
/**
93
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
94
*/
95
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
96
{
97
    s->max_seq= seq;
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    s->cycles= 0;
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    s->base_seq= seq -1;
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    s->bad_seq= RTP_SEQ_MOD + 1;
101
    s->received= 0;
102
    s->expected_prior= 0;
103
    s->received_prior= 0;
104
    s->jitter= 0;
105
    s->transit= 0;
106
}
107

    
108
/**
109
* returns 1 if we should handle this packet.
110
*/
111
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
112
{
113
    uint16_t udelta= seq - s->max_seq;
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    const int MAX_DROPOUT= 3000;
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    const int MAX_MISORDER = 100;
116
    const int MIN_SEQUENTIAL = 2;
117

    
118
    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
119
    if(s->probation)
120
    {
121
        if(seq==s->max_seq + 1) {
122
            s->probation--;
123
            s->max_seq= seq;
124
            if(s->probation==0) {
125
                rtp_init_sequence(s, seq);
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                s->received++;
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                return 1;
128
            }
129
        } else {
130
            s->probation= MIN_SEQUENTIAL - 1;
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            s->max_seq = seq;
132
        }
133
    } else if (udelta < MAX_DROPOUT) {
134
        // in order, with permissible gap
135
        if(seq < s->max_seq) {
136
            //sequence number wrapped; count antother 64k cycles
137
            s->cycles += RTP_SEQ_MOD;
138
        }
139
        s->max_seq= seq;
140
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
141
        // sequence made a large jump...
142
        if(seq==s->bad_seq) {
143
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
144
            rtp_init_sequence(s, seq);
145
        } else {
146
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
147
            return 0;
148
        }
149
    } else {
150
        // duplicate or reordered packet...
151
    }
152
    s->received++;
153
    return 1;
154
}
155

    
156
#if 0
157
/**
158
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
159
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
160
* never change.  I left this in in case someone else can see a way. (rdm)
161
*/
162
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
163
{
164
    uint32_t transit= arrival_timestamp - sent_timestamp;
165
    int d;
166
    s->transit= transit;
167
    d= FFABS(transit - s->transit);
168
    s->jitter += d - ((s->jitter + 8)>>4);
169
}
170
#endif
171

    
172
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
173
{
174
    ByteIOContext *pb;
175
    uint8_t *buf;
176
    int len;
177
    int rtcp_bytes;
178
    RTPStatistics *stats= &s->statistics;
179
    uint32_t lost;
180
    uint32_t extended_max;
181
    uint32_t expected_interval;
182
    uint32_t received_interval;
183
    uint32_t lost_interval;
184
    uint32_t expected;
185
    uint32_t fraction;
186
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
187

    
188
    if (!s->rtp_ctx || (count < 1))
189
        return -1;
190

    
191
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
192
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
193
    s->octet_count += count;
194
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
195
        RTCP_TX_RATIO_DEN;
196
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
197
    if (rtcp_bytes < 28)
198
        return -1;
199
    s->last_octet_count = s->octet_count;
200

    
201
    if (url_open_dyn_buf(&pb) < 0)
202
        return -1;
203

    
204
    // Receiver Report
205
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
206
    put_byte(pb, 201);
207
    put_be16(pb, 7); /* length in words - 1 */
208
    put_be32(pb, s->ssrc); // our own SSRC
209
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
210
    // some placeholders we should really fill...
211
    // RFC 1889/p64
212
    extended_max= stats->cycles + stats->max_seq;
213
    expected= extended_max - stats->base_seq + 1;
214
    lost= expected - stats->received;
215
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
216
    expected_interval= expected - stats->expected_prior;
217
    stats->expected_prior= expected;
218
    received_interval= stats->received - stats->received_prior;
219
    stats->received_prior= stats->received;
220
    lost_interval= expected_interval - received_interval;
221
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
222
    else fraction = (lost_interval<<8)/expected_interval;
223

    
224
    fraction= (fraction<<24) | lost;
225

    
226
    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
227
    put_be32(pb, extended_max); /* max sequence received */
228
    put_be32(pb, stats->jitter>>4); /* jitter */
229

    
230
    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
231
    {
232
        put_be32(pb, 0); /* last SR timestamp */
233
        put_be32(pb, 0); /* delay since last SR */
234
    } else {
235
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
236
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
237

    
238
        put_be32(pb, middle_32_bits); /* last SR timestamp */
239
        put_be32(pb, delay_since_last); /* delay since last SR */
240
    }
241

    
242
    // CNAME
243
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
244
    put_byte(pb, 202);
245
    len = strlen(s->hostname);
246
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
247
    put_be32(pb, s->ssrc);
248
    put_byte(pb, 0x01);
249
    put_byte(pb, len);
250
    put_buffer(pb, s->hostname, len);
251
    // padding
252
    for (len = (6 + len) % 4; len % 4; len++) {
253
        put_byte(pb, 0);
254
    }
255

    
256
    put_flush_packet(pb);
257
    len = url_close_dyn_buf(pb, &buf);
258
    if ((len > 0) && buf) {
259
        int result;
260
        dprintf(s->ic, "sending %d bytes of RR\n", len);
261
        result= url_write(s->rtp_ctx, buf, len);
262
        dprintf(s->ic, "result from url_write: %d\n", result);
263
        av_free(buf);
264
    }
265
    return 0;
266
}
267

    
268
/**
269
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
270
 * MPEG2TS streams to indicate that they should be demuxed inside the
271
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
272
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
273
 */
274
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
275
{
276
    RTPDemuxContext *s;
277

    
278
    s = av_mallocz(sizeof(RTPDemuxContext));
279
    if (!s)
280
        return NULL;
281
    s->payload_type = payload_type;
282
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
283
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
284
    s->ic = s1;
285
    s->st = st;
286
    s->rtp_payload_data = rtp_payload_data;
287
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
288
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
289
        s->ts = mpegts_parse_open(s->ic);
290
        if (s->ts == NULL) {
291
            av_free(s);
292
            return NULL;
293
        }
294
    } else {
295
        av_set_pts_info(st, 32, 1, 90000);
296
        switch(st->codec->codec_id) {
297
        case CODEC_ID_MPEG1VIDEO:
298
        case CODEC_ID_MPEG2VIDEO:
299
        case CODEC_ID_MP2:
300
        case CODEC_ID_MP3:
301
        case CODEC_ID_MPEG4:
302
        case CODEC_ID_H264:
303
            st->need_parsing = AVSTREAM_PARSE_FULL;
304
            break;
305
        default:
306
            if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
307
                av_set_pts_info(st, 32, 1, st->codec->sample_rate);
308
            }
309
            break;
310
        }
311
    }
312
    // needed to send back RTCP RR in RTSP sessions
313
    s->rtp_ctx = rtpc;
314
    gethostname(s->hostname, sizeof(s->hostname));
315
    return s;
316
}
317

    
318
void
319
rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
320
                               RTPDynamicProtocolHandler *handler)
321
{
322
    s->dynamic_protocol_context = ctx;
323
    s->parse_packet = handler->parse_packet;
324
}
325

    
326
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
327
{
328
    int au_headers_length, au_header_size, i;
329
    GetBitContext getbitcontext;
330
    RTPPayloadData *infos;
331

    
332
    infos = s->rtp_payload_data;
333

    
334
    if (infos == NULL)
335
        return -1;
336

    
337
    /* decode the first 2 bytes where the AUHeader sections are stored
338
       length in bits */
339
    au_headers_length = AV_RB16(buf);
340

    
341
    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
342
      return -1;
343

    
344
    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
345

    
346
    /* skip AU headers length section (2 bytes) */
347
    buf += 2;
348

    
349
    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
350

    
351
    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
352
    au_header_size = infos->sizelength + infos->indexlength;
353
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
354
        return -1;
355

    
356
    infos->nb_au_headers = au_headers_length / au_header_size;
357
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
358

    
359
    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
360
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
361
       but does when sending the whole as one big packet...  */
362
    infos->au_headers[0].size = 0;
363
    infos->au_headers[0].index = 0;
364
    for (i = 0; i < infos->nb_au_headers; ++i) {
365
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
366
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
367
    }
368

    
369
    infos->nb_au_headers = 1;
370

    
371
    return 0;
372
}
373

    
374
/**
375
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
376
 */
377
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
378
{
379
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
380
        int64_t addend;
381
        int delta_timestamp;
382

    
383
        /* compute pts from timestamp with received ntp_time */
384
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
385
        /* convert to the PTS timebase */
386
        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);
387
        pkt->pts = addend + delta_timestamp;
388
    }
389
}
390

    
391
/**
392
 * Parse an RTP or RTCP packet directly sent as a buffer.
393
 * @param s RTP parse context.
394
 * @param pkt returned packet
395
 * @param buf input buffer or NULL to read the next packets
396
 * @param len buffer len
397
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
398
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
399
 */
400
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
401
                     const uint8_t *buf, int len)
402
{
403
    unsigned int ssrc, h;
404
    int payload_type, seq, ret, flags = 0;
405
    AVStream *st;
406
    uint32_t timestamp;
407
    int rv= 0;
408

    
409
    if (!buf) {
410
        /* return the next packets, if any */
411
        if(s->st && s->parse_packet) {
412
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
413
            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
414
                                s->st, pkt, &timestamp, NULL, 0, flags);
415
            finalize_packet(s, pkt, timestamp);
416
            return rv;
417
        } else {
418
            // TODO: Move to a dynamic packet handler (like above)
419
            if (s->read_buf_index >= s->read_buf_size)
420
                return -1;
421
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
422
                                      s->read_buf_size - s->read_buf_index);
423
            if (ret < 0)
424
                return -1;
425
            s->read_buf_index += ret;
426
            if (s->read_buf_index < s->read_buf_size)
427
                return 1;
428
            else
429
                return 0;
430
        }
431
    }
432

    
433
    if (len < 12)
434
        return -1;
435

    
436
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
437
        return -1;
438
    if (buf[1] >= 200 && buf[1] <= 204) {
439
        rtcp_parse_packet(s, buf, len);
440
        return -1;
441
    }
442
    payload_type = buf[1] & 0x7f;
443
    if (buf[1] & 0x80)
444
        flags |= RTP_FLAG_MARKER;
445
    seq  = AV_RB16(buf + 2);
446
    timestamp = AV_RB32(buf + 4);
447
    ssrc = AV_RB32(buf + 8);
448
    /* store the ssrc in the RTPDemuxContext */
449
    s->ssrc = ssrc;
450

    
451
    /* NOTE: we can handle only one payload type */
452
    if (s->payload_type != payload_type)
453
        return -1;
454

    
455
    st = s->st;
456
    // only do something with this if all the rtp checks pass...
457
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
458
    {
459
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
460
               payload_type, seq, ((s->seq + 1) & 0xffff));
461
        return -1;
462
    }
463

    
464
    s->seq = seq;
465
    len -= 12;
466
    buf += 12;
467

    
468
    if (!st) {
469
        /* specific MPEG2TS demux support */
470
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
471
        if (ret < 0)
472
            return -1;
473
        if (ret < len) {
474
            s->read_buf_size = len - ret;
475
            memcpy(s->buf, buf + ret, s->read_buf_size);
476
            s->read_buf_index = 0;
477
            return 1;
478
        }
479
        return 0;
480
    } else if (s->parse_packet) {
481
        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
482
                             s->st, pkt, &timestamp, buf, len, flags);
483
    } else {
484
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
485
        switch(st->codec->codec_id) {
486
        case CODEC_ID_MP2:
487
            /* better than nothing: skip mpeg audio RTP header */
488
            if (len <= 4)
489
                return -1;
490
            h = AV_RB32(buf);
491
            len -= 4;
492
            buf += 4;
493
            av_new_packet(pkt, len);
494
            memcpy(pkt->data, buf, len);
495
            break;
496
        case CODEC_ID_MPEG1VIDEO:
497
        case CODEC_ID_MPEG2VIDEO:
498
            /* better than nothing: skip mpeg video RTP header */
499
            if (len <= 4)
500
                return -1;
501
            h = AV_RB32(buf);
502
            buf += 4;
503
            len -= 4;
504
            if (h & (1 << 26)) {
505
                /* mpeg2 */
506
                if (len <= 4)
507
                    return -1;
508
                buf += 4;
509
                len -= 4;
510
            }
511
            av_new_packet(pkt, len);
512
            memcpy(pkt->data, buf, len);
513
            break;
514
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
515
            // timestamps.
516
            // TODO: Put this into a dynamic packet handler...
517
        case CODEC_ID_AAC:
518
            if (rtp_parse_mp4_au(s, buf))
519
                return -1;
520
            {
521
                RTPPayloadData *infos = s->rtp_payload_data;
522
                if (infos == NULL)
523
                    return -1;
524
                buf += infos->au_headers_length_bytes + 2;
525
                len -= infos->au_headers_length_bytes + 2;
526

    
527
                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
528
                    one au_header */
529
                av_new_packet(pkt, infos->au_headers[0].size);
530
                memcpy(pkt->data, buf, infos->au_headers[0].size);
531
                buf += infos->au_headers[0].size;
532
                len -= infos->au_headers[0].size;
533
            }
534
            s->read_buf_size = len;
535
            rv= 0;
536
            break;
537
        default:
538
            av_new_packet(pkt, len);
539
            memcpy(pkt->data, buf, len);
540
            break;
541
        }
542

    
543
        pkt->stream_index = st->index;
544
    }
545

    
546
    // now perform timestamp things....
547
    finalize_packet(s, pkt, timestamp);
548

    
549
    return rv;
550
}
551

    
552
void rtp_parse_close(RTPDemuxContext *s)
553
{
554
    // TODO: fold this into the protocol specific data fields.
555
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
556
        mpegts_parse_close(s->ts);
557
    }
558
    av_free(s);
559
}