Revision 83a0d387

View differences:

libavformat/Makefile
121 121
OBJS-$(CONFIG_RM_MUXER)                  += rmenc.o
122 122
OBJS-$(CONFIG_ROQ_DEMUXER)               += idroq.o
123 123
OBJS-$(CONFIG_ROQ_MUXER)                 += raw.o
124
OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtp_mpv.o rtp_aac.o
124
OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtpenc.o rtp_mpv.o rtp_aac.o
125 125
OBJS-$(CONFIG_RTSP_DEMUXER)              += rtsp.o
126
OBJS-$(CONFIG_SDP_DEMUXER)               += rtsp.o rtp.o rtpdec.o rtp_h264.o rtp_mpv.o rtp_aac.o
126
OBJS-$(CONFIG_SDP_DEMUXER)               += rtsp.o rtp.o rtpdec.o rtp_h264.o
127 127
OBJS-$(CONFIG_SEGAFILM_DEMUXER)          += segafilm.o
128 128
OBJS-$(CONFIG_SHORTEN_DEMUXER)           += raw.o
129 129
OBJS-$(CONFIG_SIFF_DEMUXER)              += siff.o
libavformat/rtp.c
19 19
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 20
 */
21 21
#include "avformat.h"
22
#include "mpegts.h"
23 22
#include "bitstream.h"
24 23

  
25 24
#include <unistd.h>
26 25
#include "network.h"
27 26

  
28 27
#include "rtp_internal.h"
29
#include "rtp_mpv.h"
30
#include "rtp_aac.h"
31 28

  
32 29
//#define DEBUG
33 30

  
34
#define RTCP_SR_SIZE 28
35

  
36 31
/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
37 32
AVRtpPayloadType_t AVRtpPayloadTypes[]=
38 33
{
......
225 220

  
226 221
    return CODEC_ID_NONE;
227 222
}
228

  
229
/* rtp output */
230

  
231
static int rtp_write_header(AVFormatContext *s1)
232
{
233
    RTPDemuxContext *s = s1->priv_data;
234
    int payload_type, max_packet_size, n;
235
    AVStream *st;
236

  
237
    if (s1->nb_streams != 1)
238
        return -1;
239
    st = s1->streams[0];
240

  
241
    payload_type = rtp_get_payload_type(st->codec);
242
    if (payload_type < 0)
243
        payload_type = RTP_PT_PRIVATE; /* private payload type */
244
    s->payload_type = payload_type;
245

  
246
// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
247
    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
248
    s->timestamp = s->base_timestamp;
249
    s->cur_timestamp = 0;
250
    s->ssrc = 0; /* FIXME: was random(), what should this be? */
251
    s->first_packet = 1;
252
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
253

  
254
    max_packet_size = url_fget_max_packet_size(s1->pb);
255
    if (max_packet_size <= 12)
256
        return AVERROR(EIO);
257
    s->max_payload_size = max_packet_size - 12;
258

  
259
    s->max_frames_per_packet = 0;
260
    if (s1->max_delay) {
261
        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
262
            if (st->codec->frame_size == 0) {
263
                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
264
            } else {
265
                s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
266
            }
267
        }
268
        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
269
            /* FIXME: We should round down here... */
270
            s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
271
        }
272
    }
273

  
274
    av_set_pts_info(st, 32, 1, 90000);
275
    switch(st->codec->codec_id) {
276
    case CODEC_ID_MP2:
277
    case CODEC_ID_MP3:
278
        s->buf_ptr = s->buf + 4;
279
        break;
280
    case CODEC_ID_MPEG1VIDEO:
281
    case CODEC_ID_MPEG2VIDEO:
282
        break;
283
    case CODEC_ID_MPEG2TS:
284
        n = s->max_payload_size / TS_PACKET_SIZE;
285
        if (n < 1)
286
            n = 1;
287
        s->max_payload_size = n * TS_PACKET_SIZE;
288
        s->buf_ptr = s->buf;
289
        break;
290
    case CODEC_ID_AAC:
291
        s->read_buf_index = 0;
292
    default:
293
        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
294
            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
295
        }
296
        s->buf_ptr = s->buf;
297
        break;
298
    }
299

  
300
    return 0;
301
}
302

  
303
/* send an rtcp sender report packet */
304
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
305
{
306
    RTPDemuxContext *s = s1->priv_data;
307
    uint32_t rtp_ts;
308

  
309
#if defined(DEBUG)
310
    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
311
#endif
312

  
313
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
314
    s->last_rtcp_ntp_time = ntp_time;
315
    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
316
                          s1->streams[0]->time_base) + s->base_timestamp;
317
    put_byte(s1->pb, (RTP_VERSION << 6));
318
    put_byte(s1->pb, 200);
319
    put_be16(s1->pb, 6); /* length in words - 1 */
320
    put_be32(s1->pb, s->ssrc);
321
    put_be32(s1->pb, ntp_time / 1000000);
322
    put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
323
    put_be32(s1->pb, rtp_ts);
324
    put_be32(s1->pb, s->packet_count);
325
    put_be32(s1->pb, s->octet_count);
326
    put_flush_packet(s1->pb);
327
}
328

  
329
/* send an rtp packet. sequence number is incremented, but the caller
330
   must update the timestamp itself */
331
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
332
{
333
    RTPDemuxContext *s = s1->priv_data;
334

  
335
#ifdef DEBUG
336
    printf("rtp_send_data size=%d\n", len);
337
#endif
338

  
339
    /* build the RTP header */
340
    put_byte(s1->pb, (RTP_VERSION << 6));
341
    put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
342
    put_be16(s1->pb, s->seq);
343
    put_be32(s1->pb, s->timestamp);
344
    put_be32(s1->pb, s->ssrc);
345

  
346
    put_buffer(s1->pb, buf1, len);
347
    put_flush_packet(s1->pb);
348

  
349
    s->seq++;
350
    s->octet_count += len;
351
    s->packet_count++;
352
}
353

  
354
/* send an integer number of samples and compute time stamp and fill
355
   the rtp send buffer before sending. */
356
static void rtp_send_samples(AVFormatContext *s1,
357
                             const uint8_t *buf1, int size, int sample_size)
358
{
359
    RTPDemuxContext *s = s1->priv_data;
360
    int len, max_packet_size, n;
361

  
362
    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
363
    /* not needed, but who nows */
364
    if ((size % sample_size) != 0)
365
        av_abort();
366
    n = 0;
367
    while (size > 0) {
368
        s->buf_ptr = s->buf;
369
        len = FFMIN(max_packet_size, size);
370

  
371
        /* copy data */
372
        memcpy(s->buf_ptr, buf1, len);
373
        s->buf_ptr += len;
374
        buf1 += len;
375
        size -= len;
376
        s->timestamp = s->cur_timestamp + n / sample_size;
377
        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
378
        n += (s->buf_ptr - s->buf);
379
    }
380
}
381

  
382
/* NOTE: we suppose that exactly one frame is given as argument here */
383
/* XXX: test it */
384
static void rtp_send_mpegaudio(AVFormatContext *s1,
385
                               const uint8_t *buf1, int size)
386
{
387
    RTPDemuxContext *s = s1->priv_data;
388
    int len, count, max_packet_size;
389

  
390
    max_packet_size = s->max_payload_size;
391

  
392
    /* test if we must flush because not enough space */
393
    len = (s->buf_ptr - s->buf);
394
    if ((len + size) > max_packet_size) {
395
        if (len > 4) {
396
            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
397
            s->buf_ptr = s->buf + 4;
398
        }
399
    }
400
    if (s->buf_ptr == s->buf + 4) {
401
        s->timestamp = s->cur_timestamp;
402
    }
403

  
404
    /* add the packet */
405
    if (size > max_packet_size) {
406
        /* big packet: fragment */
407
        count = 0;
408
        while (size > 0) {
409
            len = max_packet_size - 4;
410
            if (len > size)
411
                len = size;
412
            /* build fragmented packet */
413
            s->buf[0] = 0;
414
            s->buf[1] = 0;
415
            s->buf[2] = count >> 8;
416
            s->buf[3] = count;
417
            memcpy(s->buf + 4, buf1, len);
418
            ff_rtp_send_data(s1, s->buf, len + 4, 0);
419
            size -= len;
420
            buf1 += len;
421
            count += len;
422
        }
423
    } else {
424
        if (s->buf_ptr == s->buf + 4) {
425
            /* no fragmentation possible */
426
            s->buf[0] = 0;
427
            s->buf[1] = 0;
428
            s->buf[2] = 0;
429
            s->buf[3] = 0;
430
        }
431
        memcpy(s->buf_ptr, buf1, size);
432
        s->buf_ptr += size;
433
    }
434
}
435

  
436
static void rtp_send_raw(AVFormatContext *s1,
437
                         const uint8_t *buf1, int size)
438
{
439
    RTPDemuxContext *s = s1->priv_data;
440
    int len, max_packet_size;
441

  
442
    max_packet_size = s->max_payload_size;
443

  
444
    while (size > 0) {
445
        len = max_packet_size;
446
        if (len > size)
447
            len = size;
448

  
449
        s->timestamp = s->cur_timestamp;
450
        ff_rtp_send_data(s1, buf1, len, (len == size));
451

  
452
        buf1 += len;
453
        size -= len;
454
    }
455
}
456

  
457
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
458
static void rtp_send_mpegts_raw(AVFormatContext *s1,
459
                                const uint8_t *buf1, int size)
460
{
461
    RTPDemuxContext *s = s1->priv_data;
462
    int len, out_len;
463

  
464
    while (size >= TS_PACKET_SIZE) {
465
        len = s->max_payload_size - (s->buf_ptr - s->buf);
466
        if (len > size)
467
            len = size;
468
        memcpy(s->buf_ptr, buf1, len);
469
        buf1 += len;
470
        size -= len;
471
        s->buf_ptr += len;
472

  
473
        out_len = s->buf_ptr - s->buf;
474
        if (out_len >= s->max_payload_size) {
475
            ff_rtp_send_data(s1, s->buf, out_len, 0);
476
            s->buf_ptr = s->buf;
477
        }
478
    }
479
}
480

  
481
/* write an RTP packet. 'buf1' must contain a single specific frame. */
482
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
483
{
484
    RTPDemuxContext *s = s1->priv_data;
485
    AVStream *st = s1->streams[0];
486
    int rtcp_bytes;
487
    int size= pkt->size;
488
    uint8_t *buf1= pkt->data;
489

  
490
#ifdef DEBUG
491
    printf("%d: write len=%d\n", pkt->stream_index, size);
492
#endif
493

  
494
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
495
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
496
        RTCP_TX_RATIO_DEN;
497
    if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
498
                           (av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
499
        rtcp_send_sr(s1, av_gettime());
500
        s->last_octet_count = s->octet_count;
501
        s->first_packet = 0;
502
    }
503
    s->cur_timestamp = s->base_timestamp + pkt->pts;
504

  
505
    switch(st->codec->codec_id) {
506
    case CODEC_ID_PCM_MULAW:
507
    case CODEC_ID_PCM_ALAW:
508
    case CODEC_ID_PCM_U8:
509
    case CODEC_ID_PCM_S8:
510
        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
511
        break;
512
    case CODEC_ID_PCM_U16BE:
513
    case CODEC_ID_PCM_U16LE:
514
    case CODEC_ID_PCM_S16BE:
515
    case CODEC_ID_PCM_S16LE:
516
        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
517
        break;
518
    case CODEC_ID_MP2:
519
    case CODEC_ID_MP3:
520
        rtp_send_mpegaudio(s1, buf1, size);
521
        break;
522
    case CODEC_ID_MPEG1VIDEO:
523
    case CODEC_ID_MPEG2VIDEO:
524
        ff_rtp_send_mpegvideo(s1, buf1, size);
525
        break;
526
    case CODEC_ID_AAC:
527
        ff_rtp_send_aac(s1, buf1, size);
528
        break;
529
    case CODEC_ID_MPEG2TS:
530
        rtp_send_mpegts_raw(s1, buf1, size);
531
        break;
532
    default:
533
        /* better than nothing : send the codec raw data */
534
        rtp_send_raw(s1, buf1, size);
535
        break;
536
    }
537
    return 0;
538
}
539

  
540
AVOutputFormat rtp_muxer = {
541
    "rtp",
542
    "RTP output format",
543
    NULL,
544
    NULL,
545
    sizeof(RTPDemuxContext),
546
    CODEC_ID_PCM_MULAW,
547
    CODEC_ID_NONE,
548
    rtp_write_header,
549
    rtp_write_packet,
550
};
libavformat/rtpenc.c
1
/*
2
 * RTP output format
3
 * Copyright (c) 2002 Fabrice Bellard.
4
 *
5
 * This file is part of FFmpeg.
6
 *
7
 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
9
 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
11
 *
12
 * 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.
16
 *
17
 * You should have received a copy of the GNU Lesser General Public
18
 * License along with FFmpeg; if not, write to the Free Software
19
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20
 */
21
#include "avformat.h"
22
#include "mpegts.h"
23
#include "bitstream.h"
24

  
25
#include <unistd.h>
26
#include "network.h"
27

  
28
#include "rtp_internal.h"
29
#include "rtp_mpv.h"
30
#include "rtp_aac.h"
31

  
32
//#define DEBUG
33

  
34
#define RTCP_SR_SIZE 28
35

  
36
static int rtp_write_header(AVFormatContext *s1)
37
{
38
    RTPDemuxContext *s = s1->priv_data;
39
    int payload_type, max_packet_size, n;
40
    AVStream *st;
41

  
42
    if (s1->nb_streams != 1)
43
        return -1;
44
    st = s1->streams[0];
45

  
46
    payload_type = rtp_get_payload_type(st->codec);
47
    if (payload_type < 0)
48
        payload_type = RTP_PT_PRIVATE; /* private payload type */
49
    s->payload_type = payload_type;
50

  
51
// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
52
    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
53
    s->timestamp = s->base_timestamp;
54
    s->cur_timestamp = 0;
55
    s->ssrc = 0; /* FIXME: was random(), what should this be? */
56
    s->first_packet = 1;
57
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
58

  
59
    max_packet_size = url_fget_max_packet_size(s1->pb);
60
    if (max_packet_size <= 12)
61
        return AVERROR(EIO);
62
    s->max_payload_size = max_packet_size - 12;
63

  
64
    s->max_frames_per_packet = 0;
65
    if (s1->max_delay) {
66
        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
67
            if (st->codec->frame_size == 0) {
68
                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
69
            } else {
70
                s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
71
            }
72
        }
73
        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
74
            /* FIXME: We should round down here... */
75
            s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
76
        }
77
    }
78

  
79
    av_set_pts_info(st, 32, 1, 90000);
80
    switch(st->codec->codec_id) {
81
    case CODEC_ID_MP2:
82
    case CODEC_ID_MP3:
83
        s->buf_ptr = s->buf + 4;
84
        break;
85
    case CODEC_ID_MPEG1VIDEO:
86
    case CODEC_ID_MPEG2VIDEO:
87
        break;
88
    case CODEC_ID_MPEG2TS:
89
        n = s->max_payload_size / TS_PACKET_SIZE;
90
        if (n < 1)
91
            n = 1;
92
        s->max_payload_size = n * TS_PACKET_SIZE;
93
        s->buf_ptr = s->buf;
94
        break;
95
    case CODEC_ID_AAC:
96
        s->read_buf_index = 0;
97
    default:
98
        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
99
            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
100
        }
101
        s->buf_ptr = s->buf;
102
        break;
103
    }
104

  
105
    return 0;
106
}
107

  
108
/* send an rtcp sender report packet */
109
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
110
{
111
    RTPDemuxContext *s = s1->priv_data;
112
    uint32_t rtp_ts;
113

  
114
#if defined(DEBUG)
115
    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
116
#endif
117

  
118
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
119
    s->last_rtcp_ntp_time = ntp_time;
120
    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
121
                          s1->streams[0]->time_base) + s->base_timestamp;
122
    put_byte(s1->pb, (RTP_VERSION << 6));
123
    put_byte(s1->pb, 200);
124
    put_be16(s1->pb, 6); /* length in words - 1 */
125
    put_be32(s1->pb, s->ssrc);
126
    put_be32(s1->pb, ntp_time / 1000000);
127
    put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
128
    put_be32(s1->pb, rtp_ts);
129
    put_be32(s1->pb, s->packet_count);
130
    put_be32(s1->pb, s->octet_count);
131
    put_flush_packet(s1->pb);
132
}
133

  
134
/* send an rtp packet. sequence number is incremented, but the caller
135
   must update the timestamp itself */
136
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
137
{
138
    RTPDemuxContext *s = s1->priv_data;
139

  
140
#ifdef DEBUG
141
    printf("rtp_send_data size=%d\n", len);
142
#endif
143

  
144
    /* build the RTP header */
145
    put_byte(s1->pb, (RTP_VERSION << 6));
146
    put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
147
    put_be16(s1->pb, s->seq);
148
    put_be32(s1->pb, s->timestamp);
149
    put_be32(s1->pb, s->ssrc);
150

  
151
    put_buffer(s1->pb, buf1, len);
152
    put_flush_packet(s1->pb);
153

  
154
    s->seq++;
155
    s->octet_count += len;
156
    s->packet_count++;
157
}
158

  
159
/* send an integer number of samples and compute time stamp and fill
160
   the rtp send buffer before sending. */
161
static void rtp_send_samples(AVFormatContext *s1,
162
                             const uint8_t *buf1, int size, int sample_size)
163
{
164
    RTPDemuxContext *s = s1->priv_data;
165
    int len, max_packet_size, n;
166

  
167
    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
168
    /* not needed, but who nows */
169
    if ((size % sample_size) != 0)
170
        av_abort();
171
    n = 0;
172
    while (size > 0) {
173
        s->buf_ptr = s->buf;
174
        len = FFMIN(max_packet_size, size);
175

  
176
        /* copy data */
177
        memcpy(s->buf_ptr, buf1, len);
178
        s->buf_ptr += len;
179
        buf1 += len;
180
        size -= len;
181
        s->timestamp = s->cur_timestamp + n / sample_size;
182
        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
183
        n += (s->buf_ptr - s->buf);
184
    }
185
}
186

  
187
/* NOTE: we suppose that exactly one frame is given as argument here */
188
/* XXX: test it */
189
static void rtp_send_mpegaudio(AVFormatContext *s1,
190
                               const uint8_t *buf1, int size)
191
{
192
    RTPDemuxContext *s = s1->priv_data;
193
    int len, count, max_packet_size;
194

  
195
    max_packet_size = s->max_payload_size;
196

  
197
    /* test if we must flush because not enough space */
198
    len = (s->buf_ptr - s->buf);
199
    if ((len + size) > max_packet_size) {
200
        if (len > 4) {
201
            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
202
            s->buf_ptr = s->buf + 4;
203
        }
204
    }
205
    if (s->buf_ptr == s->buf + 4) {
206
        s->timestamp = s->cur_timestamp;
207
    }
208

  
209
    /* add the packet */
210
    if (size > max_packet_size) {
211
        /* big packet: fragment */
212
        count = 0;
213
        while (size > 0) {
214
            len = max_packet_size - 4;
215
            if (len > size)
216
                len = size;
217
            /* build fragmented packet */
218
            s->buf[0] = 0;
219
            s->buf[1] = 0;
220
            s->buf[2] = count >> 8;
221
            s->buf[3] = count;
222
            memcpy(s->buf + 4, buf1, len);
223
            ff_rtp_send_data(s1, s->buf, len + 4, 0);
224
            size -= len;
225
            buf1 += len;
226
            count += len;
227
        }
228
    } else {
229
        if (s->buf_ptr == s->buf + 4) {
230
            /* no fragmentation possible */
231
            s->buf[0] = 0;
232
            s->buf[1] = 0;
233
            s->buf[2] = 0;
234
            s->buf[3] = 0;
235
        }
236
        memcpy(s->buf_ptr, buf1, size);
237
        s->buf_ptr += size;
238
    }
239
}
240

  
241
static void rtp_send_raw(AVFormatContext *s1,
242
                         const uint8_t *buf1, int size)
243
{
244
    RTPDemuxContext *s = s1->priv_data;
245
    int len, max_packet_size;
246

  
247
    max_packet_size = s->max_payload_size;
248

  
249
    while (size > 0) {
250
        len = max_packet_size;
251
        if (len > size)
252
            len = size;
253

  
254
        s->timestamp = s->cur_timestamp;
255
        ff_rtp_send_data(s1, buf1, len, (len == size));
256

  
257
        buf1 += len;
258
        size -= len;
259
    }
260
}
261

  
262
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
263
static void rtp_send_mpegts_raw(AVFormatContext *s1,
264
                                const uint8_t *buf1, int size)
265
{
266
    RTPDemuxContext *s = s1->priv_data;
267
    int len, out_len;
268

  
269
    while (size >= TS_PACKET_SIZE) {
270
        len = s->max_payload_size - (s->buf_ptr - s->buf);
271
        if (len > size)
272
            len = size;
273
        memcpy(s->buf_ptr, buf1, len);
274
        buf1 += len;
275
        size -= len;
276
        s->buf_ptr += len;
277

  
278
        out_len = s->buf_ptr - s->buf;
279
        if (out_len >= s->max_payload_size) {
280
            ff_rtp_send_data(s1, s->buf, out_len, 0);
281
            s->buf_ptr = s->buf;
282
        }
283
    }
284
}
285

  
286
/* write an RTP packet. 'buf1' must contain a single specific frame. */
287
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
288
{
289
    RTPDemuxContext *s = s1->priv_data;
290
    AVStream *st = s1->streams[0];
291
    int rtcp_bytes;
292
    int size= pkt->size;
293
    uint8_t *buf1= pkt->data;
294

  
295
#ifdef DEBUG
296
    printf("%d: write len=%d\n", pkt->stream_index, size);
297
#endif
298

  
299
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
300
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
301
        RTCP_TX_RATIO_DEN;
302
    if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
303
                           (av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
304
        rtcp_send_sr(s1, av_gettime());
305
        s->last_octet_count = s->octet_count;
306
        s->first_packet = 0;
307
    }
308
    s->cur_timestamp = s->base_timestamp + pkt->pts;
309

  
310
    switch(st->codec->codec_id) {
311
    case CODEC_ID_PCM_MULAW:
312
    case CODEC_ID_PCM_ALAW:
313
    case CODEC_ID_PCM_U8:
314
    case CODEC_ID_PCM_S8:
315
        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
316
        break;
317
    case CODEC_ID_PCM_U16BE:
318
    case CODEC_ID_PCM_U16LE:
319
    case CODEC_ID_PCM_S16BE:
320
    case CODEC_ID_PCM_S16LE:
321
        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
322
        break;
323
    case CODEC_ID_MP2:
324
    case CODEC_ID_MP3:
325
        rtp_send_mpegaudio(s1, buf1, size);
326
        break;
327
    case CODEC_ID_MPEG1VIDEO:
328
    case CODEC_ID_MPEG2VIDEO:
329
        ff_rtp_send_mpegvideo(s1, buf1, size);
330
        break;
331
    case CODEC_ID_AAC:
332
        ff_rtp_send_aac(s1, buf1, size);
333
        break;
334
    case CODEC_ID_MPEG2TS:
335
        rtp_send_mpegts_raw(s1, buf1, size);
336
        break;
337
    default:
338
        /* better than nothing : send the codec raw data */
339
        rtp_send_raw(s1, buf1, size);
340
        break;
341
    }
342
    return 0;
343
}
344

  
345
AVOutputFormat rtp_muxer = {
346
    "rtp",
347
    "RTP output format",
348
    NULL,
349
    NULL,
350
    sizeof(RTPDemuxContext),
351
    CODEC_ID_PCM_MULAW,
352
    CODEC_ID_NONE,
353
    rtp_write_header,
354
    rtp_write_packet,
355
};

Also available in: Unified diff