Statistics
| Branch: | Revision:

ffmpeg / libavformat / rtp.c @ 7be806f3

History | View | Annotate | Download (20.6 KB)

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
17
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
18
 */
19
#include "avformat.h"
20
#include "mpegts.h"
21

    
22
#include <unistd.h>
23
#include <sys/types.h>
24
#include <sys/socket.h>
25
#include <netinet/in.h>
26
#ifndef __BEOS__
27
# include <arpa/inet.h>
28
#else
29
# include "barpainet.h"
30
#endif
31
#include <netdb.h>
32

    
33
//#define DEBUG
34

    
35

    
36
/* TODO: - add RTCP statistics reporting (should be optional).
37

38
         - add support for h263/mpeg4 packetized output : IDEA: send a
39
         buffer to 'rtp_write_packet' contains all the packets for ONE
40
         frame. Each packet should have a four byte header containing
41
         the length in big endian format (same trick as
42
         'url_open_dyn_packet_buf') 
43
*/
44

    
45
#define RTP_VERSION 2
46

    
47
#define RTP_MAX_SDES 256   /* maximum text length for SDES */
48

    
49
/* RTCP paquets use 0.5 % of the bandwidth */
50
#define RTCP_TX_RATIO_NUM 5
51
#define RTCP_TX_RATIO_DEN 1000
52

    
53
typedef enum {
54
  RTCP_SR   = 200,
55
  RTCP_RR   = 201,
56
  RTCP_SDES = 202,
57
  RTCP_BYE  = 203,
58
  RTCP_APP  = 204
59
} rtcp_type_t;
60

    
61
typedef enum {
62
  RTCP_SDES_END    =  0,
63
  RTCP_SDES_CNAME  =  1,
64
  RTCP_SDES_NAME   =  2,
65
  RTCP_SDES_EMAIL  =  3,
66
  RTCP_SDES_PHONE  =  4,
67
  RTCP_SDES_LOC    =  5,
68
  RTCP_SDES_TOOL   =  6,
69
  RTCP_SDES_NOTE   =  7,
70
  RTCP_SDES_PRIV   =  8, 
71
  RTCP_SDES_IMG    =  9,
72
  RTCP_SDES_DOOR   = 10,
73
  RTCP_SDES_SOURCE = 11
74
} rtcp_sdes_type_t;
75

    
76
struct RTPDemuxContext {
77
    AVFormatContext *ic;
78
    AVStream *st;
79
    int payload_type;
80
    uint32_t ssrc;
81
    uint16_t seq;
82
    uint32_t timestamp;
83
    uint32_t base_timestamp;
84
    uint32_t cur_timestamp;
85
    int max_payload_size;
86
    MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
87
    int read_buf_index;
88
    int read_buf_size;
89
    
90
    /* rtcp sender statistics receive */
91
    int64_t last_rtcp_ntp_time;
92
    int64_t first_rtcp_ntp_time;
93
    uint32_t last_rtcp_timestamp;
94
    /* rtcp sender statistics */
95
    unsigned int packet_count;
96
    unsigned int octet_count;
97
    unsigned int last_octet_count;
98
    int first_packet;
99
    /* buffer for output */
100
    uint8_t buf[RTP_MAX_PACKET_LENGTH];
101
    uint8_t *buf_ptr;
102
};
103

    
104
int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
105
{
106
    switch(payload_type) {
107
    case RTP_PT_ULAW:
108
        codec->codec_type = CODEC_TYPE_AUDIO;
109
        codec->codec_id = CODEC_ID_PCM_MULAW;
110
        codec->channels = 1;
111
        codec->sample_rate = 8000;
112
        break;
113
    case RTP_PT_ALAW:
114
        codec->codec_type = CODEC_TYPE_AUDIO;
115
        codec->codec_id = CODEC_ID_PCM_ALAW;
116
        codec->channels = 1;
117
        codec->sample_rate = 8000;
118
        break;
119
    case RTP_PT_S16BE_STEREO:
120
        codec->codec_type = CODEC_TYPE_AUDIO;
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_type = CODEC_TYPE_AUDIO;
127
        codec->codec_id = CODEC_ID_PCM_S16BE;
128
        codec->channels = 1;
129
        codec->sample_rate = 44100;
130
        break;
131
    case RTP_PT_MPEGAUDIO:
132
        codec->codec_type = CODEC_TYPE_AUDIO;
133
        codec->codec_id = CODEC_ID_MP2;
134
        break;
135
    case RTP_PT_JPEG:
136
        codec->codec_type = CODEC_TYPE_VIDEO;
137
        codec->codec_id = CODEC_ID_MJPEG;
138
        break;
139
    case RTP_PT_MPEGVIDEO:
140
        codec->codec_type = CODEC_TYPE_VIDEO;
141
        codec->codec_id = CODEC_ID_MPEG1VIDEO;
142
        break;
143
    case RTP_PT_MPEG2TS:
144
        codec->codec_type = CODEC_TYPE_DATA;
145
        codec->codec_id = CODEC_ID_MPEG2TS;
146
        break;
147
    default:
148
        return -1;
149
    }
150
    return 0;
151
}
152

    
153
/* return < 0 if unknown payload type */
154
int rtp_get_payload_type(AVCodecContext *codec)
155
{
156
    int payload_type;
157

    
158
    /* compute the payload type */
159
    payload_type = -1;
160
    switch(codec->codec_id) {
161
    case CODEC_ID_PCM_MULAW:
162
        payload_type = RTP_PT_ULAW;
163
        break;
164
    case CODEC_ID_PCM_ALAW:
165
        payload_type = RTP_PT_ALAW;
166
        break;
167
    case CODEC_ID_PCM_S16BE:
168
        if (codec->channels == 1) {
169
            payload_type = RTP_PT_S16BE_MONO;
170
        } else if (codec->channels == 2) {
171
            payload_type = RTP_PT_S16BE_STEREO;
172
        }
173
        break;
174
    case CODEC_ID_MP2:
175
    case CODEC_ID_MP3:
176
        payload_type = RTP_PT_MPEGAUDIO;
177
        break;
178
    case CODEC_ID_MJPEG:
179
        payload_type = RTP_PT_JPEG;
180
        break;
181
    case CODEC_ID_MPEG1VIDEO:
182
        payload_type = RTP_PT_MPEGVIDEO;
183
        break;
184
    case CODEC_ID_MPEG2TS:
185
        payload_type = RTP_PT_MPEG2TS;
186
        break;
187
    default:
188
        break;
189
    }
190
    return payload_type;
191
}
192

    
193
static inline uint32_t decode_be32(const uint8_t *p)
194
{
195
    return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
196
}
197

    
198
static inline uint64_t decode_be64(const uint8_t *p)
199
{
200
    return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
201
}
202

    
203
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
204
{
205
    if (buf[1] != 200)
206
        return -1;
207
    s->last_rtcp_ntp_time = decode_be64(buf + 8);
208
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
209
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
210
    s->last_rtcp_timestamp = decode_be32(buf + 16);
211
    return 0;
212
}
213

    
214
/**
215
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
216
 * MPEG2TS streams to indicate that they should be demuxed inside the
217
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) 
218
 */
219
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
220
{
221
    RTPDemuxContext *s;
222

    
223
    s = av_mallocz(sizeof(RTPDemuxContext));
224
    if (!s)
225
        return NULL;
226
    s->payload_type = payload_type;
227
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
228
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
229
    s->ic = s1;
230
    s->st = st;
231
    if (payload_type == RTP_PT_MPEG2TS) {
232
        s->ts = mpegts_parse_open(s->ic);
233
        if (s->ts == NULL) {
234
            av_free(s);
235
            return NULL;
236
        }
237
    } else {
238
        switch(st->codec.codec_id) {
239
        case CODEC_ID_MPEG1VIDEO:
240
        case CODEC_ID_MPEG2VIDEO:
241
        case CODEC_ID_MP2:
242
        case CODEC_ID_MP3:
243
        case CODEC_ID_MPEG4:
244
            st->need_parsing = 1;
245
            break;
246
        default:
247
            break;
248
        }
249
    }
250
    return s;
251
}
252

    
253
/**
254
 * Parse an RTP or RTCP packet directly sent as a buffer. 
255
 * @param s RTP parse context.
256
 * @param pkt returned packet
257
 * @param buf input buffer or NULL to read the next packets
258
 * @param len buffer len
259
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow 
260
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
261
 */
262
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
263
                     const uint8_t *buf, int len)
264
{
265
    unsigned int ssrc, h;
266
    int payload_type, seq, delta_timestamp, ret;
267
    AVStream *st;
268
    uint32_t timestamp;
269
    
270
    if (!buf) {
271
        /* return the next packets, if any */
272
        if (s->read_buf_index >= s->read_buf_size)
273
            return -1;
274
        ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, 
275
                                  s->read_buf_size - s->read_buf_index);
276
        if (ret < 0)
277
            return -1;
278
        s->read_buf_index += ret;
279
        if (s->read_buf_index < s->read_buf_size)
280
            return 1;
281
        else
282
            return 0;
283
    }
284

    
285
    if (len < 12)
286
        return -1;
287

    
288
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
289
        return -1;
290
    if (buf[1] >= 200 && buf[1] <= 204) {
291
        rtcp_parse_packet(s, buf, len);
292
        return -1;
293
    }
294
    payload_type = buf[1] & 0x7f;
295
    seq  = (buf[2] << 8) | buf[3];
296
    timestamp = decode_be32(buf + 4);
297
    ssrc = decode_be32(buf + 8);
298
    
299
    /* NOTE: we can handle only one payload type */
300
    if (s->payload_type != payload_type)
301
        return -1;
302
#if defined(DEBUG) || 1
303
    if (seq != ((s->seq + 1) & 0xffff)) {
304
        av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
305
               payload_type, seq, ((s->seq + 1) & 0xffff));
306
    }
307
    s->seq = seq;
308
#endif
309
    len -= 12;
310
    buf += 12;
311

    
312
    st = s->st;
313
    if (!st) {
314
        /* specific MPEG2TS demux support */
315
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
316
        if (ret < 0)
317
            return -1;
318
        if (ret < len) {
319
            s->read_buf_size = len - ret;
320
            memcpy(s->buf, buf + ret, s->read_buf_size);
321
            s->read_buf_index = 0;
322
            return 1;
323
        }
324
    } else {
325
        switch(st->codec.codec_id) {
326
        case CODEC_ID_MP2:
327
            /* better than nothing: skip mpeg audio RTP header */
328
            if (len <= 4)
329
                return -1;
330
            h = decode_be32(buf);
331
            len -= 4;
332
            buf += 4;
333
            av_new_packet(pkt, len);
334
            memcpy(pkt->data, buf, len);
335
            break;
336
        case CODEC_ID_MPEG1VIDEO:
337
            /* better than nothing: skip mpeg video RTP header */
338
            if (len <= 4)
339
                return -1;
340
            h = decode_be32(buf);
341
            buf += 4;
342
            len -= 4;
343
            if (h & (1 << 26)) {
344
                /* mpeg2 */
345
                if (len <= 4)
346
                    return -1;
347
                buf += 4;
348
                len -= 4;
349
            }
350
            av_new_packet(pkt, len);
351
            memcpy(pkt->data, buf, len);
352
            break;
353
        default:
354
            av_new_packet(pkt, len);
355
            memcpy(pkt->data, buf, len);
356
            break;
357
        }
358
        
359
        switch(st->codec.codec_id) {
360
        case CODEC_ID_MP2:
361
        case CODEC_ID_MPEG1VIDEO:
362
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
363
                int64_t addend;
364
                /* XXX: is it really necessary to unify the timestamp base ? */
365
                /* compute pts from timestamp with received ntp_time */
366
                delta_timestamp = timestamp - s->last_rtcp_timestamp;
367
                /* convert to 90 kHz without overflow */
368
                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
369
                addend = (addend * 5625) >> 14;
370
                pkt->pts = addend + delta_timestamp;
371
            }
372
            break;
373
        default:
374
            /* no timestamp info yet */
375
            break;
376
        }
377
        pkt->stream_index = s->st->index;
378
    }
379
    return 0;
380
}
381

    
382
void rtp_parse_close(RTPDemuxContext *s)
383
{
384
    if (s->payload_type == RTP_PT_MPEG2TS) {
385
        mpegts_parse_close(s->ts);
386
    }
387
    av_free(s);
388
}
389

    
390
/* rtp output */
391

    
392
static int rtp_write_header(AVFormatContext *s1)
393
{
394
    RTPDemuxContext *s = s1->priv_data;
395
    int payload_type, max_packet_size, n;
396
    AVStream *st;
397

    
398
    if (s1->nb_streams != 1)
399
        return -1;
400
    st = s1->streams[0];
401

    
402
    payload_type = rtp_get_payload_type(&st->codec);
403
    if (payload_type < 0)
404
        payload_type = RTP_PT_PRIVATE; /* private payload type */
405
    s->payload_type = payload_type;
406

    
407
    s->base_timestamp = random();
408
    s->timestamp = s->base_timestamp;
409
    s->ssrc = random();
410
    s->first_packet = 1;
411

    
412
    max_packet_size = url_fget_max_packet_size(&s1->pb);
413
    if (max_packet_size <= 12)
414
        return AVERROR_IO;
415
    s->max_payload_size = max_packet_size - 12;
416

    
417
    switch(st->codec.codec_id) {
418
    case CODEC_ID_MP2:
419
    case CODEC_ID_MP3:
420
        s->buf_ptr = s->buf + 4;
421
        s->cur_timestamp = 0;
422
        break;
423
    case CODEC_ID_MPEG1VIDEO:
424
        s->cur_timestamp = 0;
425
        break;
426
    case CODEC_ID_MPEG2TS:
427
        n = s->max_payload_size / TS_PACKET_SIZE;
428
        if (n < 1)
429
            n = 1;
430
        s->max_payload_size = n * TS_PACKET_SIZE;
431
        s->buf_ptr = s->buf;
432
        break;
433
    default:
434
        s->buf_ptr = s->buf;
435
        break;
436
    }
437

    
438
    return 0;
439
}
440

    
441
/* send an rtcp sender report packet */
442
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
443
{
444
    RTPDemuxContext *s = s1->priv_data;
445
#if defined(DEBUG)
446
    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
447
#endif
448
    put_byte(&s1->pb, (RTP_VERSION << 6));
449
    put_byte(&s1->pb, 200);
450
    put_be16(&s1->pb, 6); /* length in words - 1 */
451
    put_be32(&s1->pb, s->ssrc);
452
    put_be64(&s1->pb, ntp_time);
453
    put_be32(&s1->pb, s->timestamp);
454
    put_be32(&s1->pb, s->packet_count);
455
    put_be32(&s1->pb, s->octet_count);
456
    put_flush_packet(&s1->pb);
457
}
458

    
459
/* send an rtp packet. sequence number is incremented, but the caller
460
   must update the timestamp itself */
461
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
462
{
463
    RTPDemuxContext *s = s1->priv_data;
464

    
465
#ifdef DEBUG
466
    printf("rtp_send_data size=%d\n", len);
467
#endif
468

    
469
    /* build the RTP header */
470
    put_byte(&s1->pb, (RTP_VERSION << 6));
471
    put_byte(&s1->pb, s->payload_type & 0x7f);
472
    put_be16(&s1->pb, s->seq);
473
    put_be32(&s1->pb, s->timestamp);
474
    put_be32(&s1->pb, s->ssrc);
475
    
476
    put_buffer(&s1->pb, buf1, len);
477
    put_flush_packet(&s1->pb);
478
    
479
    s->seq++;
480
    s->octet_count += len;
481
    s->packet_count++;
482
}
483

    
484
/* send an integer number of samples and compute time stamp and fill
485
   the rtp send buffer before sending. */
486
static void rtp_send_samples(AVFormatContext *s1,
487
                             const uint8_t *buf1, int size, int sample_size)
488
{
489
    RTPDemuxContext *s = s1->priv_data;
490
    int len, max_packet_size, n;
491

    
492
    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
493
    /* not needed, but who nows */
494
    if ((size % sample_size) != 0)
495
        av_abort();
496
    while (size > 0) {
497
        len = (max_packet_size - (s->buf_ptr - s->buf));
498
        if (len > size)
499
            len = size;
500

    
501
        /* copy data */
502
        memcpy(s->buf_ptr, buf1, len);
503
        s->buf_ptr += len;
504
        buf1 += len;
505
        size -= len;
506
        n = (s->buf_ptr - s->buf);
507
        /* if buffer full, then send it */
508
        if (n >= max_packet_size) {
509
            rtp_send_data(s1, s->buf, n);
510
            s->buf_ptr = s->buf;
511
            /* update timestamp */
512
            s->timestamp += n / sample_size;
513
        }
514
    }
515
} 
516

    
517
/* NOTE: we suppose that exactly one frame is given as argument here */
518
/* XXX: test it */
519
static void rtp_send_mpegaudio(AVFormatContext *s1,
520
                               const uint8_t *buf1, int size)
521
{
522
    RTPDemuxContext *s = s1->priv_data;
523
    AVStream *st = s1->streams[0];
524
    int len, count, max_packet_size;
525

    
526
    max_packet_size = s->max_payload_size;
527

    
528
    /* test if we must flush because not enough space */
529
    len = (s->buf_ptr - s->buf);
530
    if ((len + size) > max_packet_size) {
531
        if (len > 4) {
532
            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
533
            s->buf_ptr = s->buf + 4;
534
            /* 90 KHz time stamp */
535
            s->timestamp = s->base_timestamp + 
536
                (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
537
        }
538
    }
539

    
540
    /* add the packet */
541
    if (size > max_packet_size) {
542
        /* big packet: fragment */
543
        count = 0;
544
        while (size > 0) {
545
            len = max_packet_size - 4;
546
            if (len > size)
547
                len = size;
548
            /* build fragmented packet */
549
            s->buf[0] = 0;
550
            s->buf[1] = 0;
551
            s->buf[2] = count >> 8;
552
            s->buf[3] = count;
553
            memcpy(s->buf + 4, buf1, len);
554
            rtp_send_data(s1, s->buf, len + 4);
555
            size -= len;
556
            buf1 += len;
557
            count += len;
558
        }
559
    } else {
560
        if (s->buf_ptr == s->buf + 4) {
561
            /* no fragmentation possible */
562
            s->buf[0] = 0;
563
            s->buf[1] = 0;
564
            s->buf[2] = 0;
565
            s->buf[3] = 0;
566
        }
567
        memcpy(s->buf_ptr, buf1, size);
568
        s->buf_ptr += size;
569
    }
570
    s->cur_timestamp += st->codec.frame_size;
571
}
572

    
573
/* NOTE: a single frame must be passed with sequence header if
574
   needed. XXX: use slices. */
575
static void rtp_send_mpegvideo(AVFormatContext *s1,
576
                               const uint8_t *buf1, int size)
577
{
578
    RTPDemuxContext *s = s1->priv_data;
579
    AVStream *st = s1->streams[0];
580
    int len, h, max_packet_size;
581
    uint8_t *q;
582

    
583
    max_packet_size = s->max_payload_size;
584

    
585
    while (size > 0) {
586
        /* XXX: more correct headers */
587
        h = 0;
588
        if (st->codec.sub_id == 2)
589
            h |= 1 << 26; /* mpeg 2 indicator */
590
        q = s->buf;
591
        *q++ = h >> 24;
592
        *q++ = h >> 16;
593
        *q++ = h >> 8;
594
        *q++ = h;
595

    
596
        if (st->codec.sub_id == 2) {
597
            h = 0;
598
            *q++ = h >> 24;
599
            *q++ = h >> 16;
600
            *q++ = h >> 8;
601
            *q++ = h;
602
        }
603
        
604
        len = max_packet_size - (q - s->buf);
605
        if (len > size)
606
            len = size;
607

    
608
        memcpy(q, buf1, len);
609
        q += len;
610

    
611
        /* 90 KHz time stamp */
612
        s->timestamp = s->base_timestamp + 
613
            av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
614
        rtp_send_data(s1, s->buf, q - s->buf);
615

    
616
        buf1 += len;
617
        size -= len;
618
    }
619
    s->cur_timestamp++;
620
}
621

    
622
static void rtp_send_raw(AVFormatContext *s1,
623
                         const uint8_t *buf1, int size)
624
{
625
    RTPDemuxContext *s = s1->priv_data;
626
    AVStream *st = s1->streams[0];
627
    int len, max_packet_size;
628

    
629
    max_packet_size = s->max_payload_size;
630

    
631
    while (size > 0) {
632
        len = max_packet_size;
633
        if (len > size)
634
            len = size;
635

    
636
        /* 90 KHz time stamp */
637
        s->timestamp = s->base_timestamp + 
638
            av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
639
        rtp_send_data(s1, buf1, len);
640

    
641
        buf1 += len;
642
        size -= len;
643
    }
644
    s->cur_timestamp++;
645
}
646

    
647
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
648
static void rtp_send_mpegts_raw(AVFormatContext *s1,
649
                                const uint8_t *buf1, int size)
650
{
651
    RTPDemuxContext *s = s1->priv_data;
652
    int len, out_len;
653

    
654
    while (size >= TS_PACKET_SIZE) {
655
        len = s->max_payload_size - (s->buf_ptr - s->buf);
656
        if (len > size)
657
            len = size;
658
        memcpy(s->buf_ptr, buf1, len);
659
        buf1 += len;
660
        size -= len;
661
        s->buf_ptr += len;
662
        
663
        out_len = s->buf_ptr - s->buf;
664
        if (out_len >= s->max_payload_size) {
665
            rtp_send_data(s1, s->buf, out_len);
666
            s->buf_ptr = s->buf;
667
        }
668
    }
669
}
670

    
671
/* write an RTP packet. 'buf1' must contain a single specific frame. */
672
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
673
{
674
    RTPDemuxContext *s = s1->priv_data;
675
    AVStream *st = s1->streams[0];
676
    int rtcp_bytes;
677
    int64_t ntp_time;
678
    int size= pkt->size;
679
    uint8_t *buf1= pkt->data;
680
    
681
#ifdef DEBUG
682
    printf("%d: write len=%d\n", pkt->stream_index, size);
683
#endif
684

    
685
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
686
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / 
687
        RTCP_TX_RATIO_DEN;
688
    if (s->first_packet || rtcp_bytes >= 28) {
689
        /* compute NTP time */
690
        /* XXX: 90 kHz timestamp hardcoded */
691
        ntp_time = (pkt->pts << 28) / 5625;
692
        rtcp_send_sr(s1, ntp_time); 
693
        s->last_octet_count = s->octet_count;
694
        s->first_packet = 0;
695
    }
696

    
697
    switch(st->codec.codec_id) {
698
    case CODEC_ID_PCM_MULAW:
699
    case CODEC_ID_PCM_ALAW:
700
    case CODEC_ID_PCM_U8:
701
    case CODEC_ID_PCM_S8:
702
        rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
703
        break;
704
    case CODEC_ID_PCM_U16BE:
705
    case CODEC_ID_PCM_U16LE:
706
    case CODEC_ID_PCM_S16BE:
707
    case CODEC_ID_PCM_S16LE:
708
        rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
709
        break;
710
    case CODEC_ID_MP2:
711
    case CODEC_ID_MP3:
712
        rtp_send_mpegaudio(s1, buf1, size);
713
        break;
714
    case CODEC_ID_MPEG1VIDEO:
715
        rtp_send_mpegvideo(s1, buf1, size);
716
        break;
717
    case CODEC_ID_MPEG2TS:
718
        rtp_send_mpegts_raw(s1, buf1, size);
719
        break;
720
    default:
721
        /* better than nothing : send the codec raw data */
722
        rtp_send_raw(s1, buf1, size);
723
        break;
724
    }
725
    return 0;
726
}
727

    
728
static int rtp_write_trailer(AVFormatContext *s1)
729
{
730
    //    RTPDemuxContext *s = s1->priv_data;
731
    return 0;
732
}
733

    
734
AVOutputFormat rtp_mux = {
735
    "rtp",
736
    "RTP output format",
737
    NULL,
738
    NULL,
739
    sizeof(RTPDemuxContext),
740
    CODEC_ID_PCM_MULAW,
741
    CODEC_ID_NONE,
742
    rtp_write_header,
743
    rtp_write_packet,
744
    rtp_write_trailer,
745
};
746

    
747
int rtp_init(void)
748
{
749
    av_register_output_format(&rtp_mux);
750
    return 0;
751
}