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1
/*
2
 * RTP input format
3
 * Copyright (c) 2002 Fabrice Bellard.
4
 *
5
 * This file is part of FFmpeg.
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 *
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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
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 * 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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#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 "rtp_internal.h"
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#include "rtp_h264.h"
31

    
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//#define DEBUG
33

    
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/* TODO: - add RTCP statistics reporting (should be optional).
35

36
         - 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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*/
42

    
43
/* statistics functions */
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RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
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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};
48

    
49
static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
50
{
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    handler->next= RTPFirstDynamicPayloadHandler;
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    RTPFirstDynamicPayloadHandler= handler;
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}
54

    
55
void av_register_rtp_dynamic_payload_handlers(void)
56
{
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    register_dynamic_payload_handler(&mp4v_es_handler);
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    register_dynamic_payload_handler(&mpeg4_generic_handler);
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    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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}
61

    
62
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
63
{
64
    if (buf[1] != 200)
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        return -1;
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    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
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        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;
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}
72

    
73
#define RTP_SEQ_MOD (1<<16)
74

    
75
/**
76
* called on parse open packet
77
*/
78
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
79
{
80
    memset(s, 0, sizeof(RTPStatistics));
81
    s->max_seq= base_sequence;
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    s->probation= 1;
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}
84

    
85
/**
86
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
87
*/
88
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
89
{
90
    s->max_seq= seq;
91
    s->cycles= 0;
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    s->base_seq= seq -1;
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    s->bad_seq= RTP_SEQ_MOD + 1;
94
    s->received= 0;
95
    s->expected_prior= 0;
96
    s->received_prior= 0;
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    s->jitter= 0;
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    s->transit= 0;
99
}
100

    
101
/**
102
* returns 1 if we should handle this packet.
103
*/
104
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
105
{
106
    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;
109
    const int MIN_SEQUENTIAL = 2;
110

    
111
    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
112
    if(s->probation)
113
    {
114
        if(seq==s->max_seq + 1) {
115
            s->probation--;
116
            s->max_seq= seq;
117
            if(s->probation==0) {
118
                rtp_init_sequence(s, seq);
119
                s->received++;
120
                return 1;
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            }
122
        } else {
123
            s->probation= MIN_SEQUENTIAL - 1;
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            s->max_seq = seq;
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        }
126
    } else if (udelta < MAX_DROPOUT) {
127
        // in order, with permissible gap
128
        if(seq < s->max_seq) {
129
            //sequence number wrapped; count antother 64k cycles
130
            s->cycles += RTP_SEQ_MOD;
131
        }
132
        s->max_seq= seq;
133
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
134
        // sequence made a large jump...
135
        if(seq==s->bad_seq) {
136
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
137
            rtp_init_sequence(s, seq);
138
        } else {
139
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
140
            return 0;
141
        }
142
    } else {
143
        // duplicate or reordered packet...
144
    }
145
    s->received++;
146
    return 1;
147
}
148

    
149
#if 0
150
/**
151
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
152
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
153
* never change.  I left this in in case someone else can see a way. (rdm)
154
*/
155
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
156
{
157
    uint32_t transit= arrival_timestamp - sent_timestamp;
158
    int d;
159
    s->transit= transit;
160
    d= FFABS(transit - s->transit);
161
    s->jitter += d - ((s->jitter + 8)>>4);
162
}
163
#endif
164

    
165
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
166
{
167
    ByteIOContext *pb;
168
    uint8_t *buf;
169
    int len;
170
    int rtcp_bytes;
171
    RTPStatistics *stats= &s->statistics;
172
    uint32_t lost;
173
    uint32_t extended_max;
174
    uint32_t expected_interval;
175
    uint32_t received_interval;
176
    uint32_t lost_interval;
177
    uint32_t expected;
178
    uint32_t fraction;
179
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
180

    
181
    if (!s->rtp_ctx || (count < 1))
182
        return -1;
183

    
184
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
185
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
186
    s->octet_count += count;
187
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
188
        RTCP_TX_RATIO_DEN;
189
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
190
    if (rtcp_bytes < 28)
191
        return -1;
192
    s->last_octet_count = s->octet_count;
193

    
194
    if (url_open_dyn_buf(&pb) < 0)
195
        return -1;
196

    
197
    // Receiver Report
198
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
199
    put_byte(pb, 201);
200
    put_be16(pb, 7); /* length in words - 1 */
201
    put_be32(pb, s->ssrc); // our own SSRC
202
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
203
    // some placeholders we should really fill...
204
    // RFC 1889/p64
205
    extended_max= stats->cycles + stats->max_seq;
206
    expected= extended_max - stats->base_seq + 1;
207
    lost= expected - stats->received;
208
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
209
    expected_interval= expected - stats->expected_prior;
210
    stats->expected_prior= expected;
211
    received_interval= stats->received - stats->received_prior;
212
    stats->received_prior= stats->received;
213
    lost_interval= expected_interval - received_interval;
214
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
215
    else fraction = (lost_interval<<8)/expected_interval;
216

    
217
    fraction= (fraction<<24) | lost;
218

    
219
    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
220
    put_be32(pb, extended_max); /* max sequence received */
221
    put_be32(pb, stats->jitter>>4); /* jitter */
222

    
223
    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
224
    {
225
        put_be32(pb, 0); /* last SR timestamp */
226
        put_be32(pb, 0); /* delay since last SR */
227
    } else {
228
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
229
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
230

    
231
        put_be32(pb, middle_32_bits); /* last SR timestamp */
232
        put_be32(pb, delay_since_last); /* delay since last SR */
233
    }
234

    
235
    // CNAME
236
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
237
    put_byte(pb, 202);
238
    len = strlen(s->hostname);
239
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
240
    put_be32(pb, s->ssrc);
241
    put_byte(pb, 0x01);
242
    put_byte(pb, len);
243
    put_buffer(pb, s->hostname, len);
244
    // padding
245
    for (len = (6 + len) % 4; len % 4; len++) {
246
        put_byte(pb, 0);
247
    }
248

    
249
    put_flush_packet(pb);
250
    len = url_close_dyn_buf(pb, &buf);
251
    if ((len > 0) && buf) {
252
        int result;
253
        dprintf(s->ic, "sending %d bytes of RR\n", len);
254
        result= url_write(s->rtp_ctx, buf, len);
255
        dprintf(s->ic, "result from url_write: %d\n", result);
256
        av_free(buf);
257
    }
258
    return 0;
259
}
260

    
261
/**
262
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
263
 * MPEG2TS streams to indicate that they should be demuxed inside the
264
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
265
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
266
 */
267
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
268
{
269
    RTPDemuxContext *s;
270

    
271
    s = av_mallocz(sizeof(RTPDemuxContext));
272
    if (!s)
273
        return NULL;
274
    s->payload_type = payload_type;
275
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
276
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
277
    s->ic = s1;
278
    s->st = st;
279
    s->rtp_payload_data = rtp_payload_data;
280
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
281
    av_set_pts_info(s->st, 32, 1, 90000);
282
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
283
        s->ts = mpegts_parse_open(s->ic);
284
        if (s->ts == NULL) {
285
            av_free(s);
286
            return NULL;
287
        }
288
    } else {
289
        switch(st->codec->codec_id) {
290
        case CODEC_ID_MPEG1VIDEO:
291
        case CODEC_ID_MPEG2VIDEO:
292
        case CODEC_ID_MP2:
293
        case CODEC_ID_MP3:
294
        case CODEC_ID_MPEG4:
295
        case CODEC_ID_H264:
296
            st->need_parsing = AVSTREAM_PARSE_FULL;
297
            break;
298
        default:
299
            if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
300
                av_set_pts_info(st, 32, 1, st->codec->sample_rate);
301
            }
302
            break;
303
        }
304
    }
305
    // needed to send back RTCP RR in RTSP sessions
306
    s->rtp_ctx = rtpc;
307
    gethostname(s->hostname, sizeof(s->hostname));
308
    return s;
309
}
310

    
311
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
312
{
313
    int au_headers_length, au_header_size, i;
314
    GetBitContext getbitcontext;
315
    rtp_payload_data_t *infos;
316

    
317
    infos = s->rtp_payload_data;
318

    
319
    if (infos == NULL)
320
        return -1;
321

    
322
    /* decode the first 2 bytes where the AUHeader sections are stored
323
       length in bits */
324
    au_headers_length = AV_RB16(buf);
325

    
326
    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
327
      return -1;
328

    
329
    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
330

    
331
    /* skip AU headers length section (2 bytes) */
332
    buf += 2;
333

    
334
    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
335

    
336
    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
337
    au_header_size = infos->sizelength + infos->indexlength;
338
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
339
        return -1;
340

    
341
    infos->nb_au_headers = au_headers_length / au_header_size;
342
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
343

    
344
    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
345
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
346
       but does when sending the whole as one big packet...  */
347
    infos->au_headers[0].size = 0;
348
    infos->au_headers[0].index = 0;
349
    for (i = 0; i < infos->nb_au_headers; ++i) {
350
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
351
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
352
    }
353

    
354
    infos->nb_au_headers = 1;
355

    
356
    return 0;
357
}
358

    
359
/**
360
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
361
 */
362
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
363
{
364
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
365
        int64_t addend;
366
        int delta_timestamp;
367

    
368
        /* compute pts from timestamp with received ntp_time */
369
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
370
        /* convert to the PTS timebase */
371
        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);
372
        pkt->pts = addend + delta_timestamp;
373
    }
374
    pkt->stream_index = s->st->index;
375
}
376

    
377
/**
378
 * Parse an RTP or RTCP packet directly sent as a buffer.
379
 * @param s RTP parse context.
380
 * @param pkt returned packet
381
 * @param buf input buffer or NULL to read the next packets
382
 * @param len buffer len
383
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
384
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
385
 */
386
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
387
                     const uint8_t *buf, int len)
388
{
389
    unsigned int ssrc, h;
390
    int payload_type, seq, ret, flags = 0;
391
    AVStream *st;
392
    uint32_t timestamp;
393
    int rv= 0;
394

    
395
    if (!buf) {
396
        /* return the next packets, if any */
397
        if(s->st && s->parse_packet) {
398
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
399
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0, flags);
400
            finalize_packet(s, pkt, timestamp);
401
            return rv;
402
        } else {
403
            // TODO: Move to a dynamic packet handler (like above)
404
            if (s->read_buf_index >= s->read_buf_size)
405
                return -1;
406
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
407
                                      s->read_buf_size - s->read_buf_index);
408
            if (ret < 0)
409
                return -1;
410
            s->read_buf_index += ret;
411
            if (s->read_buf_index < s->read_buf_size)
412
                return 1;
413
            else
414
                return 0;
415
        }
416
    }
417

    
418
    if (len < 12)
419
        return -1;
420

    
421
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
422
        return -1;
423
    if (buf[1] >= 200 && buf[1] <= 204) {
424
        rtcp_parse_packet(s, buf, len);
425
        return -1;
426
    }
427
    payload_type = buf[1] & 0x7f;
428
    seq  = AV_RB16(buf + 2);
429
    timestamp = AV_RB32(buf + 4);
430
    ssrc = AV_RB32(buf + 8);
431
    /* store the ssrc in the RTPDemuxContext */
432
    s->ssrc = ssrc;
433

    
434
    /* NOTE: we can handle only one payload type */
435
    if (s->payload_type != payload_type)
436
        return -1;
437

    
438
    st = s->st;
439
    // only do something with this if all the rtp checks pass...
440
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
441
    {
442
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
443
               payload_type, seq, ((s->seq + 1) & 0xffff));
444
        return -1;
445
    }
446

    
447
    s->seq = seq;
448
    len -= 12;
449
    buf += 12;
450

    
451
    if (!st) {
452
        /* specific MPEG2TS demux support */
453
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
454
        if (ret < 0)
455
            return -1;
456
        if (ret < len) {
457
            s->read_buf_size = len - ret;
458
            memcpy(s->buf, buf + ret, s->read_buf_size);
459
            s->read_buf_index = 0;
460
            return 1;
461
        }
462
    } else if (s->parse_packet) {
463
        rv = s->parse_packet(s, pkt, &timestamp, buf, len, flags);
464
    } else {
465
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
466
        switch(st->codec->codec_id) {
467
        case CODEC_ID_MP2:
468
            /* better than nothing: skip mpeg audio RTP header */
469
            if (len <= 4)
470
                return -1;
471
            h = AV_RB32(buf);
472
            len -= 4;
473
            buf += 4;
474
            av_new_packet(pkt, len);
475
            memcpy(pkt->data, buf, len);
476
            break;
477
        case CODEC_ID_MPEG1VIDEO:
478
        case CODEC_ID_MPEG2VIDEO:
479
            /* better than nothing: skip mpeg video RTP header */
480
            if (len <= 4)
481
                return -1;
482
            h = AV_RB32(buf);
483
            buf += 4;
484
            len -= 4;
485
            if (h & (1 << 26)) {
486
                /* mpeg2 */
487
                if (len <= 4)
488
                    return -1;
489
                buf += 4;
490
                len -= 4;
491
            }
492
            av_new_packet(pkt, len);
493
            memcpy(pkt->data, buf, len);
494
            break;
495
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
496
            // timestamps.
497
            // TODO: Put this into a dynamic packet handler...
498
        case CODEC_ID_AAC:
499
            if (rtp_parse_mp4_au(s, buf))
500
                return -1;
501
            {
502
                rtp_payload_data_t *infos = s->rtp_payload_data;
503
                if (infos == NULL)
504
                    return -1;
505
                buf += infos->au_headers_length_bytes + 2;
506
                len -= infos->au_headers_length_bytes + 2;
507

    
508
                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
509
                    one au_header */
510
                av_new_packet(pkt, infos->au_headers[0].size);
511
                memcpy(pkt->data, buf, infos->au_headers[0].size);
512
                buf += infos->au_headers[0].size;
513
                len -= infos->au_headers[0].size;
514
            }
515
            s->read_buf_size = len;
516
            rv= 0;
517
            break;
518
        default:
519
            av_new_packet(pkt, len);
520
            memcpy(pkt->data, buf, len);
521
            break;
522
        }
523

    
524
        // now perform timestamp things....
525
        finalize_packet(s, pkt, timestamp);
526
    }
527
    return rv;
528
}
529

    
530
void rtp_parse_close(RTPDemuxContext *s)
531
{
532
    // TODO: fold this into the protocol specific data fields.
533
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
534
        mpegts_parse_close(s->ts);
535
    }
536
    av_free(s);
537
}