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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>
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#include <sys/socket.h>
24
#include <netinet/in.h>
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#ifndef __BEOS__
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# include <arpa/inet.h>
27
#else
28
# include "barpainet.h"
29
#endif
30
#include <netdb.h>
31

    
32
//#define DEBUG
33

    
34

    
35
/* TODO: - add RTCP statistics reporting (should be optional).
36

37
         - add support for h263/mpeg4 packetized output : IDEA: send a
38
         buffer to 'rtp_write_packet' contains all the packets for ONE
39
         frame. Each packet should have a four byte header containing
40
         the length in big endian format (same trick as
41
         'url_open_dyn_packet_buf') 
42
*/
43

    
44
#define RTP_VERSION 2
45

    
46
#define RTP_MAX_SDES 256   /* maximum text length for SDES */
47

    
48
/* RTCP paquets use 0.5 % of the bandwidth */
49
#define RTCP_TX_RATIO_NUM 5
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#define RTCP_TX_RATIO_DEN 1000
51

    
52
typedef enum {
53
  RTCP_SR   = 200,
54
  RTCP_RR   = 201,
55
  RTCP_SDES = 202,
56
  RTCP_BYE  = 203,
57
  RTCP_APP  = 204
58
} rtcp_type_t;
59

    
60
typedef enum {
61
  RTCP_SDES_END    =  0,
62
  RTCP_SDES_CNAME  =  1,
63
  RTCP_SDES_NAME   =  2,
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  RTCP_SDES_EMAIL  =  3,
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  RTCP_SDES_PHONE  =  4,
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  RTCP_SDES_LOC    =  5,
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  RTCP_SDES_TOOL   =  6,
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  RTCP_SDES_NOTE   =  7,
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  RTCP_SDES_PRIV   =  8, 
70
  RTCP_SDES_IMG    =  9,
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  RTCP_SDES_DOOR   = 10,
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  RTCP_SDES_SOURCE = 11
73
} rtcp_sdes_type_t;
74

    
75
enum RTPPayloadType {
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    RTP_PT_ULAW = 0,
77
    RTP_PT_GSM = 3,
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    RTP_PT_G723 = 4,
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    RTP_PT_ALAW = 8,
80
    RTP_PT_S16BE_STEREO = 10,
81
    RTP_PT_S16BE_MONO = 11,
82
    RTP_PT_MPEGAUDIO = 14,
83
    RTP_PT_JPEG = 26,
84
    RTP_PT_H261 = 31,
85
    RTP_PT_MPEGVIDEO = 32,
86
    RTP_PT_MPEG2TS = 33,
87
    RTP_PT_H263 = 34, /* old H263 encapsulation */
88
};
89

    
90
typedef struct RTPContext {
91
    int payload_type;
92
    UINT32 ssrc;
93
    UINT16 seq;
94
    UINT32 timestamp;
95
    UINT32 base_timestamp;
96
    UINT32 cur_timestamp;
97
    int max_payload_size;
98
    /* rtcp sender statistics receive */
99
    INT64 last_rtcp_ntp_time;
100
    UINT32 last_rtcp_timestamp;
101
    /* rtcp sender statistics */
102
    unsigned int packet_count;
103
    unsigned int octet_count;
104
    unsigned int last_octet_count;
105
    int first_packet;
106
    /* buffer for output */
107
    UINT8 buf[RTP_MAX_PACKET_LENGTH];
108
    UINT8 *buf_ptr;
109
} RTPContext;
110

    
111
int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
112
{
113
    switch(payload_type) {
114
    case RTP_PT_ULAW:
115
        codec->codec_id = CODEC_ID_PCM_MULAW;
116
        codec->channels = 1;
117
        codec->sample_rate = 8000;
118
        break;
119
    case RTP_PT_ALAW:
120
        codec->codec_id = CODEC_ID_PCM_ALAW;
121
        codec->channels = 1;
122
        codec->sample_rate = 8000;
123
        break;
124
    case RTP_PT_S16BE_STEREO:
125
        codec->codec_id = CODEC_ID_PCM_S16BE;
126
        codec->channels = 2;
127
        codec->sample_rate = 44100;
128
        break;
129
    case RTP_PT_S16BE_MONO:
130
        codec->codec_id = CODEC_ID_PCM_S16BE;
131
        codec->channels = 1;
132
        codec->sample_rate = 44100;
133
        break;
134
    case RTP_PT_MPEGAUDIO:
135
        codec->codec_id = CODEC_ID_MP2;
136
        break;
137
    case RTP_PT_JPEG:
138
        codec->codec_id = CODEC_ID_MJPEG;
139
        break;
140
    case RTP_PT_MPEGVIDEO:
141
        codec->codec_id = CODEC_ID_MPEG1VIDEO;
142
        break;
143
    default:
144
        return -1;
145
    }
146
    return 0;
147
}
148

    
149
/* return < 0 if unknown payload type */
150
int rtp_get_payload_type(AVCodecContext *codec)
151
{
152
    int payload_type;
153

    
154
    /* compute the payload type */
155
    payload_type = -1;
156
    switch(codec->codec_id) {
157
    case CODEC_ID_PCM_MULAW:
158
        payload_type = RTP_PT_ULAW;
159
        break;
160
    case CODEC_ID_PCM_ALAW:
161
        payload_type = RTP_PT_ALAW;
162
        break;
163
    case CODEC_ID_PCM_S16BE:
164
        if (codec->channels == 1) {
165
            payload_type = RTP_PT_S16BE_MONO;
166
        } else if (codec->channels == 2) {
167
            payload_type = RTP_PT_S16BE_STEREO;
168
        }
169
        break;
170
    case CODEC_ID_MP2:
171
    case CODEC_ID_MP3LAME:
172
        payload_type = RTP_PT_MPEGAUDIO;
173
        break;
174
    case CODEC_ID_MJPEG:
175
        payload_type = RTP_PT_JPEG;
176
        break;
177
    case CODEC_ID_MPEG1VIDEO:
178
        payload_type = RTP_PT_MPEGVIDEO;
179
        break;
180
    default:
181
        break;
182
    }
183
    return payload_type;
184
}
185

    
186
static inline UINT32 decode_be32(const UINT8 *p)
187
{
188
    return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
189
}
190

    
191
static inline UINT32 decode_be64(const UINT8 *p)
192
{
193
    return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
194
}
195

    
196
static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
197
{
198
    RTPContext *s = s1->priv_data;
199

    
200
    if (buf[1] != 200)
201
        return -1;
202
    s->last_rtcp_ntp_time = decode_be64(buf + 8);
203
    s->last_rtcp_timestamp = decode_be32(buf + 16);
204
    return 0;
205
}
206

    
207
/**
208
 * Parse an RTP packet directly sent as raw data. Can only be used if
209
 * 'raw' is given as input file
210
 * @param s1 media file context
211
 * @param pkt returned packet
212
 * @param buf input buffer
213
 * @param len buffer len
214
 * @return zero if no error.
215
 */
216
int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, 
217
                     const unsigned char *buf, int len)
218
{
219
    RTPContext *s = s1->priv_data;
220
    unsigned int ssrc, h;
221
    int payload_type, seq, delta_timestamp;
222
    AVStream *st;
223
    UINT32 timestamp;
224
    
225
    if (len < 12)
226
        return -1;
227

    
228
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
229
        return -1;
230
    if (buf[1] >= 200 && buf[1] <= 204) {
231
        rtcp_parse_packet(s1, buf, len);
232
        return -1;
233
    }
234
    payload_type = buf[1] & 0x7f;
235
    seq  = (buf[2] << 8) | buf[3];
236
    timestamp = decode_be32(buf + 4);
237
    ssrc = decode_be32(buf + 8);
238
    
239
    if (s->payload_type < 0) {
240
        s->payload_type = payload_type;
241
        
242
        if (payload_type == RTP_PT_MPEG2TS) {
243
            /* XXX: special case : not a single codec but a whole stream */
244
            return -1;
245
        } else {
246
            st = av_new_stream(s1, 0);
247
            if (!st)
248
                return -1;
249
            rtp_get_codec_info(&st->codec, payload_type);
250
        }
251
    }
252

    
253
    /* NOTE: we can handle only one payload type */
254
    if (s->payload_type != payload_type)
255
        return -1;
256
#if defined(DEBUG) || 1
257
    if (seq != ((s->seq + 1) & 0xffff)) {
258
        printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
259
               payload_type, seq, ((s->seq + 1) & 0xffff));
260
    }
261
    s->seq = seq;
262
#endif
263
    len -= 12;
264
    buf += 12;
265
    st = s1->streams[0];
266
    switch(st->codec.codec_id) {
267
    case CODEC_ID_MP2:
268
        /* better than nothing: skip mpeg audio RTP header */
269
        if (len <= 4)
270
            return -1;
271
        h = decode_be32(buf);
272
        len -= 4;
273
        buf += 4;
274
        av_new_packet(pkt, len);
275
        memcpy(pkt->data, buf, len);
276
        break;
277
    case CODEC_ID_MPEG1VIDEO:
278
        /* better than nothing: skip mpeg audio RTP header */
279
        if (len <= 4)
280
            return -1;
281
        h = decode_be32(buf);
282
        buf += 4;
283
        len -= 4;
284
        if (h & (1 << 26)) {
285
            /* mpeg2 */
286
            if (len <= 4)
287
                return -1;
288
            buf += 4;
289
            len -= 4;
290
        }
291
        av_new_packet(pkt, len);
292
        memcpy(pkt->data, buf, len);
293
        break;
294
    default:
295
        av_new_packet(pkt, len);
296
        memcpy(pkt->data, buf, len);
297
        break;
298
    }
299

    
300
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
301
        /* compute pts from timestamp with received ntp_time */
302
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
303
        /* XXX: do conversion, but not needed for mpeg at 90 KhZ */
304
        pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
305
    }
306
    return 0;
307
}
308

    
309
static int rtp_read_header(AVFormatContext *s1,
310
                           AVFormatParameters *ap)
311
{
312
    RTPContext *s = s1->priv_data;
313
    s->payload_type = -1;
314
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
315
    return 0;
316
}
317

    
318
static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
319
{
320
    char buf[RTP_MAX_PACKET_LENGTH];
321
    int ret;
322

    
323
    /* XXX: needs a better API for packet handling ? */
324
    for(;;) {
325
        ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
326
        if (ret < 0)
327
            return AVERROR_IO;
328
        if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
329
            break;
330
    }
331
    return 0;
332
}
333

    
334
static int rtp_read_close(AVFormatContext *s1)
335
{
336
    //    RTPContext *s = s1->priv_data;
337
    return 0;
338
}
339

    
340
static int rtp_probe(AVProbeData *p)
341
{
342
    if (strstart(p->filename, "rtp://", NULL))
343
        return AVPROBE_SCORE_MAX;
344
    return 0;
345
}
346

    
347
/* rtp output */
348

    
349
static int rtp_write_header(AVFormatContext *s1)
350
{
351
    RTPContext *s = s1->priv_data;
352
    int payload_type, max_packet_size;
353
    AVStream *st;
354

    
355
    if (s1->nb_streams != 1)
356
        return -1;
357
    st = s1->streams[0];
358

    
359
    payload_type = rtp_get_payload_type(&st->codec);
360
    if (payload_type < 0)
361
        return -1;
362
    s->payload_type = payload_type;
363

    
364
    s->base_timestamp = random();
365
    s->timestamp = s->base_timestamp;
366
    s->ssrc = random();
367
    s->first_packet = 1;
368

    
369
    max_packet_size = url_fget_max_packet_size(&s1->pb);
370
    if (max_packet_size <= 12)
371
        return AVERROR_IO;
372
    s->max_payload_size = max_packet_size - 12;
373

    
374
    switch(st->codec.codec_id) {
375
    case CODEC_ID_MP2:
376
    case CODEC_ID_MP3LAME:
377
        s->buf_ptr = s->buf + 4;
378
        s->cur_timestamp = 0;
379
        break;
380
    case CODEC_ID_MPEG1VIDEO:
381
        s->cur_timestamp = 0;
382
        break;
383
    default:
384
        s->buf_ptr = s->buf;
385
        break;
386
    }
387

    
388
    return 0;
389
}
390

    
391
/* send an rtcp sender report packet */
392
static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
393
{
394
    RTPContext *s = s1->priv_data;
395
#if defined(DEBUG)
396
    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
397
#endif
398
    put_byte(&s1->pb, (RTP_VERSION << 6));
399
    put_byte(&s1->pb, 200);
400
    put_be16(&s1->pb, 6); /* length in words - 1 */
401
    put_be32(&s1->pb, s->ssrc);
402
    put_be64(&s1->pb, ntp_time);
403
    put_be32(&s1->pb, s->timestamp);
404
    put_be32(&s1->pb, s->packet_count);
405
    put_be32(&s1->pb, s->octet_count);
406
    put_flush_packet(&s1->pb);
407
}
408

    
409
/* send an rtp packet. sequence number is incremented, but the caller
410
   must update the timestamp itself */
411
static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
412
{
413
    RTPContext *s = s1->priv_data;
414

    
415
#ifdef DEBUG
416
    printf("rtp_send_data size=%d\n", len);
417
#endif
418

    
419
    /* build the RTP header */
420
    put_byte(&s1->pb, (RTP_VERSION << 6));
421
    put_byte(&s1->pb, s->payload_type & 0x7f);
422
    put_be16(&s1->pb, s->seq);
423
    put_be32(&s1->pb, s->timestamp);
424
    put_be32(&s1->pb, s->ssrc);
425
    
426
    put_buffer(&s1->pb, buf1, len);
427
    put_flush_packet(&s1->pb);
428
    
429
    s->seq++;
430
    s->octet_count += len;
431
    s->packet_count++;
432
}
433

    
434
/* send an integer number of samples and compute time stamp and fill
435
   the rtp send buffer before sending. */
436
static void rtp_send_samples(AVFormatContext *s1,
437
                             UINT8 *buf1, int size, int sample_size)
438
{
439
    RTPContext *s = s1->priv_data;
440
    int len, max_packet_size, n;
441

    
442
    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
443
    /* not needed, but who nows */
444
    if ((size % sample_size) != 0)
445
        av_abort();
446
    while (size > 0) {
447
        len = (max_packet_size - (s->buf_ptr - s->buf));
448
        if (len > size)
449
            len = size;
450

    
451
        /* copy data */
452
        memcpy(s->buf_ptr, buf1, len);
453
        s->buf_ptr += len;
454
        buf1 += len;
455
        size -= len;
456
        n = (s->buf_ptr - s->buf);
457
        /* if buffer full, then send it */
458
        if (n >= max_packet_size) {
459
            rtp_send_data(s1, s->buf, n);
460
            s->buf_ptr = s->buf;
461
            /* update timestamp */
462
            s->timestamp += n / sample_size;
463
        }
464
    }
465
} 
466

    
467
/* NOTE: we suppose that exactly one frame is given as argument here */
468
/* XXX: test it */
469
static void rtp_send_mpegaudio(AVFormatContext *s1,
470
                               UINT8 *buf1, int size)
471
{
472
    RTPContext *s = s1->priv_data;
473
    AVStream *st = s1->streams[0];
474
    int len, count, max_packet_size;
475

    
476
    max_packet_size = s->max_payload_size;
477

    
478
    /* test if we must flush because not enough space */
479
    len = (s->buf_ptr - s->buf);
480
    if ((len + size) > max_packet_size) {
481
        if (len > 4) {
482
            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
483
            s->buf_ptr = s->buf + 4;
484
            /* 90 KHz time stamp */
485
            s->timestamp = s->base_timestamp + 
486
                (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
487
        }
488
    }
489

    
490
    /* add the packet */
491
    if (size > max_packet_size) {
492
        /* big packet: fragment */
493
        count = 0;
494
        while (size > 0) {
495
            len = max_packet_size - 4;
496
            if (len > size)
497
                len = size;
498
            /* build fragmented packet */
499
            s->buf[0] = 0;
500
            s->buf[1] = 0;
501
            s->buf[2] = count >> 8;
502
            s->buf[3] = count;
503
            memcpy(s->buf + 4, buf1, len);
504
            rtp_send_data(s1, s->buf, len + 4);
505
            size -= len;
506
            buf1 += len;
507
            count += len;
508
        }
509
    } else {
510
        if (s->buf_ptr == s->buf + 4) {
511
            /* no fragmentation possible */
512
            s->buf[0] = 0;
513
            s->buf[1] = 0;
514
            s->buf[2] = 0;
515
            s->buf[3] = 0;
516
        }
517
        memcpy(s->buf_ptr, buf1, size);
518
        s->buf_ptr += size;
519
    }
520
    s->cur_timestamp += st->codec.frame_size;
521
}
522

    
523
/* NOTE: a single frame must be passed with sequence header if
524
   needed. XXX: use slices. */
525
static void rtp_send_mpegvideo(AVFormatContext *s1,
526
                               UINT8 *buf1, int size)
527
{
528
    RTPContext *s = s1->priv_data;
529
    AVStream *st = s1->streams[0];
530
    int len, h, max_packet_size;
531
    UINT8 *q;
532

    
533
    max_packet_size = s->max_payload_size;
534

    
535
    while (size > 0) {
536
        /* XXX: more correct headers */
537
        h = 0;
538
        if (st->codec.sub_id == 2)
539
            h |= 1 << 26; /* mpeg 2 indicator */
540
        q = s->buf;
541
        *q++ = h >> 24;
542
        *q++ = h >> 16;
543
        *q++ = h >> 8;
544
        *q++ = h;
545

    
546
        if (st->codec.sub_id == 2) {
547
            h = 0;
548
            *q++ = h >> 24;
549
            *q++ = h >> 16;
550
            *q++ = h >> 8;
551
            *q++ = h;
552
        }
553
        
554
        len = max_packet_size - (q - s->buf);
555
        if (len > size)
556
            len = size;
557

    
558
        memcpy(q, buf1, len);
559
        q += len;
560

    
561
        /* 90 KHz time stamp */
562
        /* XXX: overflow */
563
        s->timestamp = s->base_timestamp + 
564
            (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
565
        rtp_send_data(s1, s->buf, q - s->buf);
566

    
567
        buf1 += len;
568
        size -= len;
569
    }
570
    s->cur_timestamp++;
571
}
572

    
573
/* write an RTP packet. 'buf1' must contain a single specific frame. */
574
static int rtp_write_packet(AVFormatContext *s1, int stream_index,
575
                            UINT8 *buf1, int size, int force_pts)
576
{
577
    RTPContext *s = s1->priv_data;
578
    AVStream *st = s1->streams[0];
579
    int rtcp_bytes;
580
    INT64 ntp_time;
581
    
582
#ifdef DEBUG
583
    printf("%d: write len=%d\n", stream_index, size);
584
#endif
585

    
586
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
587
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / 
588
        RTCP_TX_RATIO_DEN;
589
    if (s->first_packet || rtcp_bytes >= 28) {
590
        /* compute NTP time */
591
        ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
592
        rtcp_send_sr(s1, ntp_time); 
593
        s->last_octet_count = s->octet_count;
594
        s->first_packet = 0;
595
    }
596

    
597
    switch(st->codec.codec_id) {
598
    case CODEC_ID_PCM_MULAW:
599
    case CODEC_ID_PCM_ALAW:
600
    case CODEC_ID_PCM_U8:
601
    case CODEC_ID_PCM_S8:
602
        rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
603
        break;
604
    case CODEC_ID_PCM_U16BE:
605
    case CODEC_ID_PCM_U16LE:
606
    case CODEC_ID_PCM_S16BE:
607
    case CODEC_ID_PCM_S16LE:
608
        rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
609
        break;
610
    case CODEC_ID_MP2:
611
    case CODEC_ID_MP3LAME:
612
        rtp_send_mpegaudio(s1, buf1, size);
613
        break;
614
    case CODEC_ID_MPEG1VIDEO:
615
        rtp_send_mpegvideo(s1, buf1, size);
616
        break;
617
    default:
618
        return AVERROR_IO;
619
    }
620
    return 0;
621
}
622

    
623
static int rtp_write_trailer(AVFormatContext *s1)
624
{
625
    //    RTPContext *s = s1->priv_data;
626
    return 0;
627
}
628

    
629
AVInputFormat rtp_demux = {
630
    "rtp",
631
    "RTP input format",
632
    sizeof(RTPContext),    
633
    rtp_probe,
634
    rtp_read_header,
635
    rtp_read_packet,
636
    rtp_read_close,
637
    .flags = AVFMT_NOHEADER,
638
};
639

    
640
AVOutputFormat rtp_mux = {
641
    "rtp",
642
    "RTP output format",
643
    NULL,
644
    NULL,
645
    sizeof(RTPContext),
646
    CODEC_ID_PCM_MULAW,
647
    CODEC_ID_NONE,
648
    rtp_write_header,
649
    rtp_write_packet,
650
    rtp_write_trailer,
651
};
652

    
653
int rtp_init(void)
654
{
655
    av_register_output_format(&rtp_mux);
656
    av_register_input_format(&rtp_demux);
657
    return 0;
658
}