make AVFMT_NOHEADER flag dynamic - added av_open_input_stream()
[libav.git] / libavformat / rtp.c
CommitLineData
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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
17 * 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>
b8a78f41 23#include <sys/socket.h>
e309128f 24#include <netinet/in.h>
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25#ifndef __BEOS__
26# include <arpa/inet.h>
27#else
28# include "barpainet.h"
29#endif
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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
50#define RTCP_TX_RATIO_DEN 1000
51
52typedef 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
60typedef enum {
61 RTCP_SDES_END = 0,
62 RTCP_SDES_CNAME = 1,
63 RTCP_SDES_NAME = 2,
64 RTCP_SDES_EMAIL = 3,
65 RTCP_SDES_PHONE = 4,
66 RTCP_SDES_LOC = 5,
67 RTCP_SDES_TOOL = 6,
68 RTCP_SDES_NOTE = 7,
69 RTCP_SDES_PRIV = 8,
70 RTCP_SDES_IMG = 9,
71 RTCP_SDES_DOOR = 10,
72 RTCP_SDES_SOURCE = 11
73} rtcp_sdes_type_t;
74
75enum RTPPayloadType {
76 RTP_PT_ULAW = 0,
77 RTP_PT_GSM = 3,
78 RTP_PT_G723 = 4,
79 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 */
65e70450 88 RTP_PT_PRIVATE = 96,
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89};
90
91typedef struct RTPContext {
92 int payload_type;
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93 uint32_t ssrc;
94 uint16_t seq;
95 uint32_t timestamp;
96 uint32_t base_timestamp;
97 uint32_t cur_timestamp;
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98 int max_payload_size;
99 /* rtcp sender statistics receive */
0c1a9eda 100 int64_t last_rtcp_ntp_time;
e5526b2c 101 int64_t first_rtcp_ntp_time;
0c1a9eda 102 uint32_t last_rtcp_timestamp;
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103 /* rtcp sender statistics */
104 unsigned int packet_count;
105 unsigned int octet_count;
106 unsigned int last_octet_count;
107 int first_packet;
108 /* buffer for output */
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109 uint8_t buf[RTP_MAX_PACKET_LENGTH];
110 uint8_t *buf_ptr;
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111} RTPContext;
112
113int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
114{
115 switch(payload_type) {
116 case RTP_PT_ULAW:
117 codec->codec_id = CODEC_ID_PCM_MULAW;
118 codec->channels = 1;
119 codec->sample_rate = 8000;
120 break;
121 case RTP_PT_ALAW:
122 codec->codec_id = CODEC_ID_PCM_ALAW;
123 codec->channels = 1;
124 codec->sample_rate = 8000;
125 break;
126 case RTP_PT_S16BE_STEREO:
127 codec->codec_id = CODEC_ID_PCM_S16BE;
128 codec->channels = 2;
129 codec->sample_rate = 44100;
130 break;
131 case RTP_PT_S16BE_MONO:
132 codec->codec_id = CODEC_ID_PCM_S16BE;
133 codec->channels = 1;
134 codec->sample_rate = 44100;
135 break;
136 case RTP_PT_MPEGAUDIO:
137 codec->codec_id = CODEC_ID_MP2;
138 break;
139 case RTP_PT_JPEG:
140 codec->codec_id = CODEC_ID_MJPEG;
141 break;
142 case RTP_PT_MPEGVIDEO:
143 codec->codec_id = CODEC_ID_MPEG1VIDEO;
144 break;
145 default:
146 return -1;
147 }
148 return 0;
149}
150
151/* return < 0 if unknown payload type */
152int rtp_get_payload_type(AVCodecContext *codec)
153{
154 int payload_type;
155
156 /* compute the payload type */
157 payload_type = -1;
158 switch(codec->codec_id) {
159 case CODEC_ID_PCM_MULAW:
160 payload_type = RTP_PT_ULAW;
161 break;
162 case CODEC_ID_PCM_ALAW:
163 payload_type = RTP_PT_ALAW;
164 break;
165 case CODEC_ID_PCM_S16BE:
166 if (codec->channels == 1) {
167 payload_type = RTP_PT_S16BE_MONO;
168 } else if (codec->channels == 2) {
169 payload_type = RTP_PT_S16BE_STEREO;
170 }
171 break;
172 case CODEC_ID_MP2:
80783dc2 173 case CODEC_ID_MP3:
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174 payload_type = RTP_PT_MPEGAUDIO;
175 break;
176 case CODEC_ID_MJPEG:
177 payload_type = RTP_PT_JPEG;
178 break;
179 case CODEC_ID_MPEG1VIDEO:
180 payload_type = RTP_PT_MPEGVIDEO;
181 break;
182 default:
183 break;
184 }
185 return payload_type;
186}
187
0c1a9eda 188static inline uint32_t decode_be32(const uint8_t *p)
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189{
190 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
191}
192
e5526b2c 193static inline uint64_t decode_be64(const uint8_t *p)
e309128f 194{
0c1a9eda 195 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
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196}
197
198static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
199{
200 RTPContext *s = s1->priv_data;
201
202 if (buf[1] != 200)
203 return -1;
204 s->last_rtcp_ntp_time = decode_be64(buf + 8);
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205 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
206 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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207 s->last_rtcp_timestamp = decode_be32(buf + 16);
208 return 0;
209}
210
211/**
212 * Parse an RTP packet directly sent as raw data. Can only be used if
213 * 'raw' is given as input file
214 * @param s1 media file context
215 * @param pkt returned packet
216 * @param buf input buffer
217 * @param len buffer len
218 * @return zero if no error.
219 */
220int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
221 const unsigned char *buf, int len)
222{
223 RTPContext *s = s1->priv_data;
224 unsigned int ssrc, h;
225 int payload_type, seq, delta_timestamp;
226 AVStream *st;
0c1a9eda 227 uint32_t timestamp;
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228
229 if (len < 12)
230 return -1;
231
232 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
233 return -1;
234 if (buf[1] >= 200 && buf[1] <= 204) {
235 rtcp_parse_packet(s1, buf, len);
236 return -1;
237 }
238 payload_type = buf[1] & 0x7f;
239 seq = (buf[2] << 8) | buf[3];
240 timestamp = decode_be32(buf + 4);
241 ssrc = decode_be32(buf + 8);
242
243 if (s->payload_type < 0) {
244 s->payload_type = payload_type;
245
246 if (payload_type == RTP_PT_MPEG2TS) {
247 /* XXX: special case : not a single codec but a whole stream */
248 return -1;
249 } else {
250 st = av_new_stream(s1, 0);
251 if (!st)
252 return -1;
253 rtp_get_codec_info(&st->codec, payload_type);
254 }
255 }
256
257 /* NOTE: we can handle only one payload type */
258 if (s->payload_type != payload_type)
259 return -1;
260#if defined(DEBUG) || 1
261 if (seq != ((s->seq + 1) & 0xffff)) {
262 printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
263 payload_type, seq, ((s->seq + 1) & 0xffff));
264 }
265 s->seq = seq;
266#endif
267 len -= 12;
268 buf += 12;
269 st = s1->streams[0];
270 switch(st->codec.codec_id) {
271 case CODEC_ID_MP2:
272 /* better than nothing: skip mpeg audio RTP header */
273 if (len <= 4)
274 return -1;
275 h = decode_be32(buf);
276 len -= 4;
277 buf += 4;
278 av_new_packet(pkt, len);
279 memcpy(pkt->data, buf, len);
280 break;
281 case CODEC_ID_MPEG1VIDEO:
282 /* better than nothing: skip mpeg audio RTP header */
283 if (len <= 4)
284 return -1;
285 h = decode_be32(buf);
286 buf += 4;
287 len -= 4;
288 if (h & (1 << 26)) {
289 /* mpeg2 */
290 if (len <= 4)
291 return -1;
292 buf += 4;
293 len -= 4;
294 }
295 av_new_packet(pkt, len);
296 memcpy(pkt->data, buf, len);
297 break;
298 default:
299 av_new_packet(pkt, len);
300 memcpy(pkt->data, buf, len);
301 break;
302 }
303
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304 switch(st->codec.codec_id) {
305 case CODEC_ID_MP2:
306 case CODEC_ID_MPEG1VIDEO:
307 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
308 int64_t addend;
309 /* XXX: is it really necessary to unify the timestamp base ? */
310 /* compute pts from timestamp with received ntp_time */
311 delta_timestamp = timestamp - s->last_rtcp_timestamp;
312 /* convert to 90 kHz without overflow */
313 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
314 addend = (addend * 5625) >> 14;
315 pkt->pts = addend + delta_timestamp;
316 }
317 break;
318 default:
319 /* no timestamp info yet */
320 break;
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321 }
322 return 0;
323}
324
325static int rtp_read_header(AVFormatContext *s1,
326 AVFormatParameters *ap)
327{
328 RTPContext *s = s1->priv_data;
329 s->payload_type = -1;
330 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
e5526b2c 331 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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332 return 0;
333}
334
335static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
336{
337 char buf[RTP_MAX_PACKET_LENGTH];
338 int ret;
339
340 /* XXX: needs a better API for packet handling ? */
341 for(;;) {
342 ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
343 if (ret < 0)
344 return AVERROR_IO;
345 if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
346 break;
347 }
348 return 0;
349}
350
351static int rtp_read_close(AVFormatContext *s1)
352{
353 // RTPContext *s = s1->priv_data;
354 return 0;
355}
356
357static int rtp_probe(AVProbeData *p)
358{
359 if (strstart(p->filename, "rtp://", NULL))
360 return AVPROBE_SCORE_MAX;
361 return 0;
362}
363
364/* rtp output */
365
366static int rtp_write_header(AVFormatContext *s1)
367{
368 RTPContext *s = s1->priv_data;
369 int payload_type, max_packet_size;
370 AVStream *st;
371
372 if (s1->nb_streams != 1)
373 return -1;
374 st = s1->streams[0];
375
376 payload_type = rtp_get_payload_type(&st->codec);
377 if (payload_type < 0)
65e70450 378 payload_type = RTP_PT_PRIVATE; /* private payload type */
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379 s->payload_type = payload_type;
380
381 s->base_timestamp = random();
382 s->timestamp = s->base_timestamp;
383 s->ssrc = random();
384 s->first_packet = 1;
385
386 max_packet_size = url_fget_max_packet_size(&s1->pb);
387 if (max_packet_size <= 12)
388 return AVERROR_IO;
389 s->max_payload_size = max_packet_size - 12;
390
391 switch(st->codec.codec_id) {
392 case CODEC_ID_MP2:
80783dc2 393 case CODEC_ID_MP3:
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394 s->buf_ptr = s->buf + 4;
395 s->cur_timestamp = 0;
396 break;
397 case CODEC_ID_MPEG1VIDEO:
398 s->cur_timestamp = 0;
399 break;
400 default:
401 s->buf_ptr = s->buf;
402 break;
403 }
404
405 return 0;
406}
407
408/* send an rtcp sender report packet */
0c1a9eda 409static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
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410{
411 RTPContext *s = s1->priv_data;
412#if defined(DEBUG)
413 printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
414#endif
415 put_byte(&s1->pb, (RTP_VERSION << 6));
416 put_byte(&s1->pb, 200);
417 put_be16(&s1->pb, 6); /* length in words - 1 */
418 put_be32(&s1->pb, s->ssrc);
419 put_be64(&s1->pb, ntp_time);
420 put_be32(&s1->pb, s->timestamp);
421 put_be32(&s1->pb, s->packet_count);
422 put_be32(&s1->pb, s->octet_count);
423 put_flush_packet(&s1->pb);
424}
425
426/* send an rtp packet. sequence number is incremented, but the caller
427 must update the timestamp itself */
49057904 428static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
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429{
430 RTPContext *s = s1->priv_data;
431
432#ifdef DEBUG
433 printf("rtp_send_data size=%d\n", len);
434#endif
435
436 /* build the RTP header */
437 put_byte(&s1->pb, (RTP_VERSION << 6));
438 put_byte(&s1->pb, s->payload_type & 0x7f);
439 put_be16(&s1->pb, s->seq);
440 put_be32(&s1->pb, s->timestamp);
441 put_be32(&s1->pb, s->ssrc);
442
443 put_buffer(&s1->pb, buf1, len);
444 put_flush_packet(&s1->pb);
445
446 s->seq++;
447 s->octet_count += len;
448 s->packet_count++;
449}
450
451/* send an integer number of samples and compute time stamp and fill
452 the rtp send buffer before sending. */
453static void rtp_send_samples(AVFormatContext *s1,
49057904 454 const uint8_t *buf1, int size, int sample_size)
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455{
456 RTPContext *s = s1->priv_data;
457 int len, max_packet_size, n;
458
459 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
460 /* not needed, but who nows */
461 if ((size % sample_size) != 0)
462 av_abort();
463 while (size > 0) {
464 len = (max_packet_size - (s->buf_ptr - s->buf));
465 if (len > size)
466 len = size;
467
468 /* copy data */
469 memcpy(s->buf_ptr, buf1, len);
470 s->buf_ptr += len;
471 buf1 += len;
472 size -= len;
473 n = (s->buf_ptr - s->buf);
474 /* if buffer full, then send it */
475 if (n >= max_packet_size) {
476 rtp_send_data(s1, s->buf, n);
477 s->buf_ptr = s->buf;
478 /* update timestamp */
479 s->timestamp += n / sample_size;
480 }
481 }
482}
483
484/* NOTE: we suppose that exactly one frame is given as argument here */
485/* XXX: test it */
486static void rtp_send_mpegaudio(AVFormatContext *s1,
49057904 487 const uint8_t *buf1, int size)
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488{
489 RTPContext *s = s1->priv_data;
490 AVStream *st = s1->streams[0];
491 int len, count, max_packet_size;
492
493 max_packet_size = s->max_payload_size;
494
495 /* test if we must flush because not enough space */
496 len = (s->buf_ptr - s->buf);
497 if ((len + size) > max_packet_size) {
498 if (len > 4) {
499 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
500 s->buf_ptr = s->buf + 4;
501 /* 90 KHz time stamp */
502 s->timestamp = s->base_timestamp +
503 (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
504 }
505 }
506
507 /* add the packet */
508 if (size > max_packet_size) {
509 /* big packet: fragment */
510 count = 0;
511 while (size > 0) {
512 len = max_packet_size - 4;
513 if (len > size)
514 len = size;
515 /* build fragmented packet */
516 s->buf[0] = 0;
517 s->buf[1] = 0;
518 s->buf[2] = count >> 8;
519 s->buf[3] = count;
520 memcpy(s->buf + 4, buf1, len);
521 rtp_send_data(s1, s->buf, len + 4);
522 size -= len;
523 buf1 += len;
524 count += len;
525 }
526 } else {
527 if (s->buf_ptr == s->buf + 4) {
528 /* no fragmentation possible */
529 s->buf[0] = 0;
530 s->buf[1] = 0;
531 s->buf[2] = 0;
532 s->buf[3] = 0;
533 }
534 memcpy(s->buf_ptr, buf1, size);
535 s->buf_ptr += size;
536 }
537 s->cur_timestamp += st->codec.frame_size;
538}
539
540/* NOTE: a single frame must be passed with sequence header if
541 needed. XXX: use slices. */
542static void rtp_send_mpegvideo(AVFormatContext *s1,
49057904 543 const uint8_t *buf1, int size)
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544{
545 RTPContext *s = s1->priv_data;
546 AVStream *st = s1->streams[0];
547 int len, h, max_packet_size;
0c1a9eda 548 uint8_t *q;
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549
550 max_packet_size = s->max_payload_size;
551
552 while (size > 0) {
553 /* XXX: more correct headers */
554 h = 0;
555 if (st->codec.sub_id == 2)
556 h |= 1 << 26; /* mpeg 2 indicator */
557 q = s->buf;
558 *q++ = h >> 24;
559 *q++ = h >> 16;
560 *q++ = h >> 8;
561 *q++ = h;
562
563 if (st->codec.sub_id == 2) {
564 h = 0;
565 *q++ = h >> 24;
566 *q++ = h >> 16;
567 *q++ = h >> 8;
568 *q++ = h;
569 }
570
571 len = max_packet_size - (q - s->buf);
572 if (len > size)
573 len = size;
574
575 memcpy(q, buf1, len);
576 q += len;
577
578 /* 90 KHz time stamp */
e309128f 579 s->timestamp = s->base_timestamp +
14bea432 580 av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
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581 rtp_send_data(s1, s->buf, q - s->buf);
582
583 buf1 += len;
584 size -= len;
585 }
586 s->cur_timestamp++;
587}
588
65e70450 589static void rtp_send_raw(AVFormatContext *s1,
49057904 590 const uint8_t *buf1, int size)
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591{
592 RTPContext *s = s1->priv_data;
593 AVStream *st = s1->streams[0];
594 int len, max_packet_size;
595
596 max_packet_size = s->max_payload_size;
597
598 while (size > 0) {
599 len = max_packet_size;
600 if (len > size)
601 len = size;
602
603 /* 90 KHz time stamp */
65e70450 604 s->timestamp = s->base_timestamp +
14bea432 605 av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
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606 rtp_send_data(s1, buf1, len);
607
608 buf1 += len;
609 size -= len;
610 }
611 s->cur_timestamp++;
612}
613
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614/* write an RTP packet. 'buf1' must contain a single specific frame. */
615static int rtp_write_packet(AVFormatContext *s1, int stream_index,
49057904 616 const uint8_t *buf1, int size, int64_t pts)
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617{
618 RTPContext *s = s1->priv_data;
619 AVStream *st = s1->streams[0];
620 int rtcp_bytes;
0c1a9eda 621 int64_t ntp_time;
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622
623#ifdef DEBUG
624 printf("%d: write len=%d\n", stream_index, size);
625#endif
626
627 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
628 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
629 RTCP_TX_RATIO_DEN;
630 if (s->first_packet || rtcp_bytes >= 28) {
631 /* compute NTP time */
e5526b2c 632 /* XXX: 90 kHz timestamp hardcoded */
49057904 633 ntp_time = (pts << 28) / 5625;
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634 rtcp_send_sr(s1, ntp_time);
635 s->last_octet_count = s->octet_count;
636 s->first_packet = 0;
637 }
638
639 switch(st->codec.codec_id) {
640 case CODEC_ID_PCM_MULAW:
641 case CODEC_ID_PCM_ALAW:
642 case CODEC_ID_PCM_U8:
643 case CODEC_ID_PCM_S8:
644 rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
645 break;
646 case CODEC_ID_PCM_U16BE:
647 case CODEC_ID_PCM_U16LE:
648 case CODEC_ID_PCM_S16BE:
649 case CODEC_ID_PCM_S16LE:
650 rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
651 break;
652 case CODEC_ID_MP2:
80783dc2 653 case CODEC_ID_MP3:
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654 rtp_send_mpegaudio(s1, buf1, size);
655 break;
656 case CODEC_ID_MPEG1VIDEO:
657 rtp_send_mpegvideo(s1, buf1, size);
658 break;
659 default:
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660 /* better than nothing : send the codec raw data */
661 rtp_send_raw(s1, buf1, size);
662 break;
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663 }
664 return 0;
665}
666
667static int rtp_write_trailer(AVFormatContext *s1)
668{
669 // RTPContext *s = s1->priv_data;
670 return 0;
671}
672
673AVInputFormat rtp_demux = {
674 "rtp",
675 "RTP input format",
676 sizeof(RTPContext),
677 rtp_probe,
678 rtp_read_header,
679 rtp_read_packet,
680 rtp_read_close,
bb76a117 681 .flags = AVFMT_NOHEADER,
e309128f
FB
682};
683
684AVOutputFormat rtp_mux = {
685 "rtp",
686 "RTP output format",
687 NULL,
688 NULL,
689 sizeof(RTPContext),
690 CODEC_ID_PCM_MULAW,
691 CODEC_ID_NONE,
692 rtp_write_header,
693 rtp_write_packet,
694 rtp_write_trailer,
695};
696
697int rtp_init(void)
698{
699 av_register_output_format(&rtp_mux);
700 av_register_input_format(&rtp_demux);
701 return 0;
702}