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1
/*
2
 * RTP input/output format
3
 * Copyright (c) 2002 Fabrice Bellard.
4
 *
5
 * This library is free software; you can redistribute it and/or
6
 * modify it under the terms of the GNU Lesser General Public
7
 * License as published by the Free Software Foundation; either
8
 * version 2 of the License, or (at your option) any later version.
9
 *
10
 * This library is distributed in the hope that it will be useful,
11
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
13
 * Lesser General Public License for more details.
14
 *
15
 * You should have received a copy of the GNU Lesser General Public
16
 * License along with this library; if not, write to the Free Software
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 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
18
 */
19
#include "avformat.h"
20

    
21
#include <unistd.h>
22
#include <sys/types.h>
23
#include <sys/socket.h>
24
#include <netinet/in.h>
25
#include <arpa/inet.h>
26
#include <netdb.h>
27

    
28
//#define DEBUG
29

    
30

    
31
/* TODO: - add RTCP statistics reporting (should be optional).
32

33
         - add support for h263/mpeg4 packetized output : IDEA: send a
34
         buffer to 'rtp_write_packet' contains all the packets for ONE
35
         frame. Each packet should have a four byte header containing
36
         the length in big endian format (same trick as
37
         'url_open_dyn_packet_buf') 
38
*/
39

    
40
#define RTP_VERSION 2
41

    
42
#define RTP_MAX_SDES 256   /* maximum text length for SDES */
43

    
44
/* RTCP paquets use 0.5 % of the bandwidth */
45
#define RTCP_TX_RATIO_NUM 5
46
#define RTCP_TX_RATIO_DEN 1000
47

    
48
typedef enum {
49
  RTCP_SR   = 200,
50
  RTCP_RR   = 201,
51
  RTCP_SDES = 202,
52
  RTCP_BYE  = 203,
53
  RTCP_APP  = 204
54
} rtcp_type_t;
55

    
56
typedef enum {
57
  RTCP_SDES_END    =  0,
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  RTCP_SDES_CNAME  =  1,
59
  RTCP_SDES_NAME   =  2,
60
  RTCP_SDES_EMAIL  =  3,
61
  RTCP_SDES_PHONE  =  4,
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  RTCP_SDES_LOC    =  5,
63
  RTCP_SDES_TOOL   =  6,
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  RTCP_SDES_NOTE   =  7,
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  RTCP_SDES_PRIV   =  8, 
66
  RTCP_SDES_IMG    =  9,
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  RTCP_SDES_DOOR   = 10,
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  RTCP_SDES_SOURCE = 11
69
} rtcp_sdes_type_t;
70

    
71
enum RTPPayloadType {
72
    RTP_PT_ULAW = 0,
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    RTP_PT_GSM = 3,
74
    RTP_PT_G723 = 4,
75
    RTP_PT_ALAW = 8,
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    RTP_PT_S16BE_STEREO = 10,
77
    RTP_PT_S16BE_MONO = 11,
78
    RTP_PT_MPEGAUDIO = 14,
79
    RTP_PT_JPEG = 26,
80
    RTP_PT_H261 = 31,
81
    RTP_PT_MPEGVIDEO = 32,
82
    RTP_PT_MPEG2TS = 33,
83
    RTP_PT_H263 = 34, /* old H263 encapsulation */
84
};
85

    
86
typedef struct RTPContext {
87
    int payload_type;
88
    UINT32 ssrc;
89
    UINT16 seq;
90
    UINT32 timestamp;
91
    UINT32 base_timestamp;
92
    UINT32 cur_timestamp;
93
    int max_payload_size;
94
    /* rtcp sender statistics receive */
95
    INT64 last_rtcp_ntp_time;
96
    UINT32 last_rtcp_timestamp;
97
    /* rtcp sender statistics */
98
    unsigned int packet_count;
99
    unsigned int octet_count;
100
    unsigned int last_octet_count;
101
    int first_packet;
102
    /* buffer for output */
103
    UINT8 buf[RTP_MAX_PACKET_LENGTH];
104
    UINT8 *buf_ptr;
105
} RTPContext;
106

    
107
int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
108
{
109
    switch(payload_type) {
110
    case RTP_PT_ULAW:
111
        codec->codec_id = CODEC_ID_PCM_MULAW;
112
        codec->channels = 1;
113
        codec->sample_rate = 8000;
114
        break;
115
    case RTP_PT_ALAW:
116
        codec->codec_id = CODEC_ID_PCM_ALAW;
117
        codec->channels = 1;
118
        codec->sample_rate = 8000;
119
        break;
120
    case RTP_PT_S16BE_STEREO:
121
        codec->codec_id = CODEC_ID_PCM_S16BE;
122
        codec->channels = 2;
123
        codec->sample_rate = 44100;
124
        break;
125
    case RTP_PT_S16BE_MONO:
126
        codec->codec_id = CODEC_ID_PCM_S16BE;
127
        codec->channels = 1;
128
        codec->sample_rate = 44100;
129
        break;
130
    case RTP_PT_MPEGAUDIO:
131
        codec->codec_id = CODEC_ID_MP2;
132
        break;
133
    case RTP_PT_JPEG:
134
        codec->codec_id = CODEC_ID_MJPEG;
135
        break;
136
    case RTP_PT_MPEGVIDEO:
137
        codec->codec_id = CODEC_ID_MPEG1VIDEO;
138
        break;
139
    default:
140
        return -1;
141
    }
142
    return 0;
143
}
144

    
145
/* return < 0 if unknown payload type */
146
int rtp_get_payload_type(AVCodecContext *codec)
147
{
148
    int payload_type;
149

    
150
    /* compute the payload type */
151
    payload_type = -1;
152
    switch(codec->codec_id) {
153
    case CODEC_ID_PCM_MULAW:
154
        payload_type = RTP_PT_ULAW;
155
        break;
156
    case CODEC_ID_PCM_ALAW:
157
        payload_type = RTP_PT_ALAW;
158
        break;
159
    case CODEC_ID_PCM_S16BE:
160
        if (codec->channels == 1) {
161
            payload_type = RTP_PT_S16BE_MONO;
162
        } else if (codec->channels == 2) {
163
            payload_type = RTP_PT_S16BE_STEREO;
164
        }
165
        break;
166
    case CODEC_ID_MP2:
167
    case CODEC_ID_MP3LAME:
168
        payload_type = RTP_PT_MPEGAUDIO;
169
        break;
170
    case CODEC_ID_MJPEG:
171
        payload_type = RTP_PT_JPEG;
172
        break;
173
    case CODEC_ID_MPEG1VIDEO:
174
        payload_type = RTP_PT_MPEGVIDEO;
175
        break;
176
    default:
177
        break;
178
    }
179
    return payload_type;
180
}
181

    
182
static inline UINT32 decode_be32(const UINT8 *p)
183
{
184
    return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
185
}
186

    
187
static inline UINT32 decode_be64(const UINT8 *p)
188
{
189
    return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
190
}
191

    
192
static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
193
{
194
    RTPContext *s = s1->priv_data;
195

    
196
    if (buf[1] != 200)
197
        return -1;
198
    s->last_rtcp_ntp_time = decode_be64(buf + 8);
199
    s->last_rtcp_timestamp = decode_be32(buf + 16);
200
    return 0;
201
}
202

    
203
/**
204
 * Parse an RTP packet directly sent as raw data. Can only be used if
205
 * 'raw' is given as input file
206
 * @param s1 media file context
207
 * @param pkt returned packet
208
 * @param buf input buffer
209
 * @param len buffer len
210
 * @return zero if no error.
211
 */
212
int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, 
213
                     const unsigned char *buf, int len)
214
{
215
    RTPContext *s = s1->priv_data;
216
    unsigned int ssrc, h;
217
    int payload_type, seq, delta_timestamp;
218
    AVStream *st;
219
    UINT32 timestamp;
220
    
221
    if (len < 12)
222
        return -1;
223

    
224
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
225
        return -1;
226
    if (buf[1] >= 200 && buf[1] <= 204) {
227
        rtcp_parse_packet(s1, buf, len);
228
        return -1;
229
    }
230
    payload_type = buf[1] & 0x7f;
231
    seq  = (buf[2] << 8) | buf[3];
232
    timestamp = decode_be32(buf + 4);
233
    ssrc = decode_be32(buf + 8);
234
    
235
    if (s->payload_type < 0) {
236
        s->payload_type = payload_type;
237
        
238
        if (payload_type == RTP_PT_MPEG2TS) {
239
            /* XXX: special case : not a single codec but a whole stream */
240
            return -1;
241
        } else {
242
            st = av_new_stream(s1, 0);
243
            if (!st)
244
                return -1;
245
            rtp_get_codec_info(&st->codec, payload_type);
246
        }
247
    }
248

    
249
    /* NOTE: we can handle only one payload type */
250
    if (s->payload_type != payload_type)
251
        return -1;
252
#if defined(DEBUG) || 1
253
    if (seq != ((s->seq + 1) & 0xffff)) {
254
        printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
255
               payload_type, seq, ((s->seq + 1) & 0xffff));
256
    }
257
    s->seq = seq;
258
#endif
259
    len -= 12;
260
    buf += 12;
261
    st = s1->streams[0];
262
    switch(st->codec.codec_id) {
263
    case CODEC_ID_MP2:
264
        /* better than nothing: skip mpeg audio RTP header */
265
        if (len <= 4)
266
            return -1;
267
        h = decode_be32(buf);
268
        len -= 4;
269
        buf += 4;
270
        av_new_packet(pkt, len);
271
        memcpy(pkt->data, buf, len);
272
        break;
273
    case CODEC_ID_MPEG1VIDEO:
274
        /* better than nothing: skip mpeg audio RTP header */
275
        if (len <= 4)
276
            return -1;
277
        h = decode_be32(buf);
278
        buf += 4;
279
        len -= 4;
280
        if (h & (1 << 26)) {
281
            /* mpeg2 */
282
            if (len <= 4)
283
                return -1;
284
            buf += 4;
285
            len -= 4;
286
        }
287
        av_new_packet(pkt, len);
288
        memcpy(pkt->data, buf, len);
289
        break;
290
    default:
291
        av_new_packet(pkt, len);
292
        memcpy(pkt->data, buf, len);
293
        break;
294
    }
295

    
296
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
297
        /* compute pts from timestamp with received ntp_time */
298
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
299
        /* XXX: do conversion, but not needed for mpeg at 90 KhZ */
300
        pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
301
    }
302
    return 0;
303
}
304

    
305
static int rtp_read_header(AVFormatContext *s1,
306
                           AVFormatParameters *ap)
307
{
308
    RTPContext *s = s1->priv_data;
309
    s->payload_type = -1;
310
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
311
    return 0;
312
}
313

    
314
static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
315
{
316
    char buf[RTP_MAX_PACKET_LENGTH];
317
    int ret;
318

    
319
    /* XXX: needs a better API for packet handling ? */
320
    for(;;) {
321
        ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
322
        if (ret < 0)
323
            return AVERROR_IO;
324
        if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
325
            break;
326
    }
327
    return 0;
328
}
329

    
330
static int rtp_read_close(AVFormatContext *s1)
331
{
332
    //    RTPContext *s = s1->priv_data;
333
    return 0;
334
}
335

    
336
static int rtp_probe(AVProbeData *p)
337
{
338
    if (strstart(p->filename, "rtp://", NULL))
339
        return AVPROBE_SCORE_MAX;
340
    return 0;
341
}
342

    
343
/* rtp output */
344

    
345
static int rtp_write_header(AVFormatContext *s1)
346
{
347
    RTPContext *s = s1->priv_data;
348
    int payload_type, max_packet_size;
349
    AVStream *st;
350

    
351
    if (s1->nb_streams != 1)
352
        return -1;
353
    st = s1->streams[0];
354

    
355
    payload_type = rtp_get_payload_type(&st->codec);
356
    if (payload_type < 0)
357
        return -1;
358
    s->payload_type = payload_type;
359

    
360
    s->base_timestamp = random();
361
    s->timestamp = s->base_timestamp;
362
    s->ssrc = random();
363
    s->first_packet = 1;
364

    
365
    max_packet_size = url_fget_max_packet_size(&s1->pb);
366
    if (max_packet_size <= 12)
367
        return AVERROR_IO;
368
    s->max_payload_size = max_packet_size - 12;
369

    
370
    switch(st->codec.codec_id) {
371
    case CODEC_ID_MP2:
372
    case CODEC_ID_MP3LAME:
373
        s->buf_ptr = s->buf + 4;
374
        s->cur_timestamp = 0;
375
        break;
376
    case CODEC_ID_MPEG1VIDEO:
377
        s->cur_timestamp = 0;
378
        break;
379
    default:
380
        s->buf_ptr = s->buf;
381
        break;
382
    }
383

    
384
    return 0;
385
}
386

    
387
/* send an rtcp sender report packet */
388
static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
389
{
390
    RTPContext *s = s1->priv_data;
391
#if defined(DEBUG)
392
    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
393
#endif
394
    put_byte(&s1->pb, (RTP_VERSION << 6));
395
    put_byte(&s1->pb, 200);
396
    put_be16(&s1->pb, 6); /* length in words - 1 */
397
    put_be32(&s1->pb, s->ssrc);
398
    put_be64(&s1->pb, ntp_time);
399
    put_be32(&s1->pb, s->timestamp);
400
    put_be32(&s1->pb, s->packet_count);
401
    put_be32(&s1->pb, s->octet_count);
402
    put_flush_packet(&s1->pb);
403
}
404

    
405
/* send an rtp packet. sequence number is incremented, but the caller
406
   must update the timestamp itself */
407
static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
408
{
409
    RTPContext *s = s1->priv_data;
410

    
411
#ifdef DEBUG
412
    printf("rtp_send_data size=%d\n", len);
413
#endif
414

    
415
    /* build the RTP header */
416
    put_byte(&s1->pb, (RTP_VERSION << 6));
417
    put_byte(&s1->pb, s->payload_type & 0x7f);
418
    put_be16(&s1->pb, s->seq);
419
    put_be32(&s1->pb, s->timestamp);
420
    put_be32(&s1->pb, s->ssrc);
421
    
422
    put_buffer(&s1->pb, buf1, len);
423
    put_flush_packet(&s1->pb);
424
    
425
    s->seq++;
426
    s->octet_count += len;
427
    s->packet_count++;
428
}
429

    
430
/* send an integer number of samples and compute time stamp and fill
431
   the rtp send buffer before sending. */
432
static void rtp_send_samples(AVFormatContext *s1,
433
                             UINT8 *buf1, int size, int sample_size)
434
{
435
    RTPContext *s = s1->priv_data;
436
    int len, max_packet_size, n;
437

    
438
    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
439
    /* not needed, but who nows */
440
    if ((size % sample_size) != 0)
441
        av_abort();
442
    while (size > 0) {
443
        len = (max_packet_size - (s->buf_ptr - s->buf));
444
        if (len > size)
445
            len = size;
446

    
447
        /* copy data */
448
        memcpy(s->buf_ptr, buf1, len);
449
        s->buf_ptr += len;
450
        buf1 += len;
451
        size -= len;
452
        n = (s->buf_ptr - s->buf);
453
        /* if buffer full, then send it */
454
        if (n >= max_packet_size) {
455
            rtp_send_data(s1, s->buf, n);
456
            s->buf_ptr = s->buf;
457
            /* update timestamp */
458
            s->timestamp += n / sample_size;
459
        }
460
    }
461
} 
462

    
463
/* NOTE: we suppose that exactly one frame is given as argument here */
464
/* XXX: test it */
465
static void rtp_send_mpegaudio(AVFormatContext *s1,
466
                               UINT8 *buf1, int size)
467
{
468
    RTPContext *s = s1->priv_data;
469
    AVStream *st = s1->streams[0];
470
    int len, count, max_packet_size;
471

    
472
    max_packet_size = s->max_payload_size;
473

    
474
    /* test if we must flush because not enough space */
475
    len = (s->buf_ptr - s->buf);
476
    if ((len + size) > max_packet_size) {
477
        if (len > 4) {
478
            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
479
            s->buf_ptr = s->buf + 4;
480
            /* 90 KHz time stamp */
481
            s->timestamp = s->base_timestamp + 
482
                (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
483
        }
484
    }
485

    
486
    /* add the packet */
487
    if (size > max_packet_size) {
488
        /* big packet: fragment */
489
        count = 0;
490
        while (size > 0) {
491
            len = max_packet_size - 4;
492
            if (len > size)
493
                len = size;
494
            /* build fragmented packet */
495
            s->buf[0] = 0;
496
            s->buf[1] = 0;
497
            s->buf[2] = count >> 8;
498
            s->buf[3] = count;
499
            memcpy(s->buf + 4, buf1, len);
500
            rtp_send_data(s1, s->buf, len + 4);
501
            size -= len;
502
            buf1 += len;
503
            count += len;
504
        }
505
    } else {
506
        if (s->buf_ptr == s->buf + 4) {
507
            /* no fragmentation possible */
508
            s->buf[0] = 0;
509
            s->buf[1] = 0;
510
            s->buf[2] = 0;
511
            s->buf[3] = 0;
512
        }
513
        memcpy(s->buf_ptr, buf1, size);
514
        s->buf_ptr += size;
515
    }
516
    s->cur_timestamp += st->codec.frame_size;
517
}
518

    
519
/* NOTE: a single frame must be passed with sequence header if
520
   needed. XXX: use slices. */
521
static void rtp_send_mpegvideo(AVFormatContext *s1,
522
                               UINT8 *buf1, int size)
523
{
524
    RTPContext *s = s1->priv_data;
525
    AVStream *st = s1->streams[0];
526
    int len, h, max_packet_size;
527
    UINT8 *q;
528

    
529
    max_packet_size = s->max_payload_size;
530

    
531
    while (size > 0) {
532
        /* XXX: more correct headers */
533
        h = 0;
534
        if (st->codec.sub_id == 2)
535
            h |= 1 << 26; /* mpeg 2 indicator */
536
        q = s->buf;
537
        *q++ = h >> 24;
538
        *q++ = h >> 16;
539
        *q++ = h >> 8;
540
        *q++ = h;
541

    
542
        if (st->codec.sub_id == 2) {
543
            h = 0;
544
            *q++ = h >> 24;
545
            *q++ = h >> 16;
546
            *q++ = h >> 8;
547
            *q++ = h;
548
        }
549
        
550
        len = max_packet_size - (q - s->buf);
551
        if (len > size)
552
            len = size;
553

    
554
        memcpy(q, buf1, len);
555
        q += len;
556

    
557
        /* 90 KHz time stamp */
558
        /* XXX: overflow */
559
        s->timestamp = s->base_timestamp + 
560
            (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
561
        rtp_send_data(s1, s->buf, q - s->buf);
562

    
563
        buf1 += len;
564
        size -= len;
565
    }
566
    s->cur_timestamp++;
567
}
568

    
569
/* write an RTP packet. 'buf1' must contain a single specific frame. */
570
static int rtp_write_packet(AVFormatContext *s1, int stream_index,
571
                            UINT8 *buf1, int size, int force_pts)
572
{
573
    RTPContext *s = s1->priv_data;
574
    AVStream *st = s1->streams[0];
575
    int rtcp_bytes;
576
    INT64 ntp_time;
577
    
578
#ifdef DEBUG
579
    printf("%d: write len=%d\n", stream_index, size);
580
#endif
581

    
582
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
583
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / 
584
        RTCP_TX_RATIO_DEN;
585
    if (s->first_packet || rtcp_bytes >= 28) {
586
        /* compute NTP time */
587
        ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
588
        rtcp_send_sr(s1, ntp_time); 
589
        s->last_octet_count = s->octet_count;
590
        s->first_packet = 0;
591
    }
592

    
593
    switch(st->codec.codec_id) {
594
    case CODEC_ID_PCM_MULAW:
595
    case CODEC_ID_PCM_ALAW:
596
    case CODEC_ID_PCM_U8:
597
    case CODEC_ID_PCM_S8:
598
        rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
599
        break;
600
    case CODEC_ID_PCM_U16BE:
601
    case CODEC_ID_PCM_U16LE:
602
    case CODEC_ID_PCM_S16BE:
603
    case CODEC_ID_PCM_S16LE:
604
        rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
605
        break;
606
    case CODEC_ID_MP2:
607
    case CODEC_ID_MP3LAME:
608
        rtp_send_mpegaudio(s1, buf1, size);
609
        break;
610
    case CODEC_ID_MPEG1VIDEO:
611
        rtp_send_mpegvideo(s1, buf1, size);
612
        break;
613
    default:
614
        return AVERROR_IO;
615
    }
616
    return 0;
617
}
618

    
619
static int rtp_write_trailer(AVFormatContext *s1)
620
{
621
    //    RTPContext *s = s1->priv_data;
622
    return 0;
623
}
624

    
625
AVInputFormat rtp_demux = {
626
    "rtp",
627
    "RTP input format",
628
    sizeof(RTPContext),    
629
    rtp_probe,
630
    rtp_read_header,
631
    rtp_read_packet,
632
    rtp_read_close,
633
    .flags = AVFMT_NOHEADER,
634
};
635

    
636
AVOutputFormat rtp_mux = {
637
    "rtp",
638
    "RTP output format",
639
    NULL,
640
    NULL,
641
    sizeof(RTPContext),
642
    CODEC_ID_PCM_MULAW,
643
    CODEC_ID_NONE,
644
    rtp_write_header,
645
    rtp_write_packet,
646
    rtp_write_trailer,
647
};
648

    
649
int rtp_init(void)
650
{
651
    av_register_output_format(&rtp_mux);
652
    av_register_input_format(&rtp_demux);
653
    return 0;
654
}