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[libav.git] / libavformat / rtpdec.c
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1/*
2 * RTP input format
406792e7 3 * Copyright (c) 2002 Fabrice Bellard
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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 */
245976da 21
7246177d 22/* needed for gethostname() */
d0feff2a 23#define _XOPEN_SOURCE 600
7246177d 24
245976da 25#include "libavcodec/bitstream.h"
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26#include "avformat.h"
27#include "mpegts.h"
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28
29#include <unistd.h>
30#include "network.h"
31
302879cb 32#include "rtpdec.h"
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33#include "rtp_h264.h"
34
35//#define DEBUG
36
37/* TODO: - add RTCP statistics reporting (should be optional).
38
39 - add support for h263/mpeg4 packetized output : IDEA: send a
40 buffer to 'rtp_write_packet' contains all the packets for ONE
41 frame. Each packet should have a four byte header containing
42 the length in big endian format (same trick as
43 'url_open_dyn_packet_buf')
44*/
45
46/* statistics functions */
47RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
48
49static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
50static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
51
0369d2b0 52void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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53{
54 handler->next= RTPFirstDynamicPayloadHandler;
55 RTPFirstDynamicPayloadHandler= handler;
56}
57
58void av_register_rtp_dynamic_payload_handlers(void)
59{
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60 ff_register_dynamic_payload_handler(&mp4v_es_handler);
61 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
62 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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63}
64
65static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
66{
67 if (buf[1] != 200)
68 return -1;
69 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
70 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
71 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
72 s->last_rtcp_timestamp = AV_RB32(buf + 16);
73 return 0;
74}
75
76#define RTP_SEQ_MOD (1<<16)
77
78/**
79* called on parse open packet
80*/
81static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
82{
83 memset(s, 0, sizeof(RTPStatistics));
84 s->max_seq= base_sequence;
85 s->probation= 1;
86}
87
88/**
89* called whenever there is a large jump in sequence numbers, or when they get out of probation...
90*/
91static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
92{
93 s->max_seq= seq;
94 s->cycles= 0;
95 s->base_seq= seq -1;
96 s->bad_seq= RTP_SEQ_MOD + 1;
97 s->received= 0;
98 s->expected_prior= 0;
99 s->received_prior= 0;
100 s->jitter= 0;
101 s->transit= 0;
102}
103
104/**
105* returns 1 if we should handle this packet.
106*/
107static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
108{
109 uint16_t udelta= seq - s->max_seq;
110 const int MAX_DROPOUT= 3000;
111 const int MAX_MISORDER = 100;
112 const int MIN_SEQUENTIAL = 2;
113
114 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
115 if(s->probation)
116 {
117 if(seq==s->max_seq + 1) {
118 s->probation--;
119 s->max_seq= seq;
120 if(s->probation==0) {
121 rtp_init_sequence(s, seq);
122 s->received++;
123 return 1;
124 }
125 } else {
126 s->probation= MIN_SEQUENTIAL - 1;
127 s->max_seq = seq;
128 }
129 } else if (udelta < MAX_DROPOUT) {
130 // in order, with permissible gap
131 if(seq < s->max_seq) {
132 //sequence number wrapped; count antother 64k cycles
133 s->cycles += RTP_SEQ_MOD;
134 }
135 s->max_seq= seq;
136 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
137 // sequence made a large jump...
138 if(seq==s->bad_seq) {
139 // two sequential packets-- assume that the other side restarted without telling us; just resync.
140 rtp_init_sequence(s, seq);
141 } else {
142 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
143 return 0;
144 }
145 } else {
146 // duplicate or reordered packet...
147 }
148 s->received++;
149 return 1;
150}
151
152#if 0
153/**
154* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
155* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
156* never change. I left this in in case someone else can see a way. (rdm)
157*/
158static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
159{
160 uint32_t transit= arrival_timestamp - sent_timestamp;
161 int d;
162 s->transit= transit;
163 d= FFABS(transit - s->transit);
164 s->jitter += d - ((s->jitter + 8)>>4);
165}
166#endif
167
168int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
169{
170 ByteIOContext *pb;
171 uint8_t *buf;
172 int len;
173 int rtcp_bytes;
174 RTPStatistics *stats= &s->statistics;
175 uint32_t lost;
176 uint32_t extended_max;
177 uint32_t expected_interval;
178 uint32_t received_interval;
179 uint32_t lost_interval;
180 uint32_t expected;
181 uint32_t fraction;
182 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
183
184 if (!s->rtp_ctx || (count < 1))
185 return -1;
186
187 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
188 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
189 s->octet_count += count;
190 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
191 RTCP_TX_RATIO_DEN;
192 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
193 if (rtcp_bytes < 28)
194 return -1;
195 s->last_octet_count = s->octet_count;
196
197 if (url_open_dyn_buf(&pb) < 0)
198 return -1;
199
200 // Receiver Report
201 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
202 put_byte(pb, 201);
203 put_be16(pb, 7); /* length in words - 1 */
204 put_be32(pb, s->ssrc); // our own SSRC
205 put_be32(pb, s->ssrc); // XXX: should be the server's here!
206 // some placeholders we should really fill...
207 // RFC 1889/p64
208 extended_max= stats->cycles + stats->max_seq;
209 expected= extended_max - stats->base_seq + 1;
210 lost= expected - stats->received;
211 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
212 expected_interval= expected - stats->expected_prior;
213 stats->expected_prior= expected;
214 received_interval= stats->received - stats->received_prior;
215 stats->received_prior= stats->received;
216 lost_interval= expected_interval - received_interval;
217 if (expected_interval==0 || lost_interval<=0) fraction= 0;
218 else fraction = (lost_interval<<8)/expected_interval;
219
220 fraction= (fraction<<24) | lost;
221
222 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
223 put_be32(pb, extended_max); /* max sequence received */
224 put_be32(pb, stats->jitter>>4); /* jitter */
225
226 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
227 {
228 put_be32(pb, 0); /* last SR timestamp */
229 put_be32(pb, 0); /* delay since last SR */
230 } else {
231 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
232 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
233
234 put_be32(pb, middle_32_bits); /* last SR timestamp */
235 put_be32(pb, delay_since_last); /* delay since last SR */
236 }
237
238 // CNAME
239 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
240 put_byte(pb, 202);
241 len = strlen(s->hostname);
242 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
243 put_be32(pb, s->ssrc);
244 put_byte(pb, 0x01);
245 put_byte(pb, len);
246 put_buffer(pb, s->hostname, len);
247 // padding
248 for (len = (6 + len) % 4; len % 4; len++) {
249 put_byte(pb, 0);
250 }
251
252 put_flush_packet(pb);
253 len = url_close_dyn_buf(pb, &buf);
254 if ((len > 0) && buf) {
255 int result;
e8420626 256 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 257 result= url_write(s->rtp_ctx, buf, len);
e8420626 258 dprintf(s->ic, "result from url_write: %d\n", result);
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259 av_free(buf);
260 }
261 return 0;
262}
263
264/**
265 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
266 * MPEG2TS streams to indicate that they should be demuxed inside the
267 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
268 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
269 */
be73a544 270RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
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271{
272 RTPDemuxContext *s;
273
274 s = av_mallocz(sizeof(RTPDemuxContext));
275 if (!s)
276 return NULL;
277 s->payload_type = payload_type;
278 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
279 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
280 s->ic = s1;
281 s->st = st;
282 s->rtp_payload_data = rtp_payload_data;
283 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
284 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
285 s->ts = mpegts_parse_open(s->ic);
286 if (s->ts == NULL) {
287 av_free(s);
288 return NULL;
289 }
290 } else {
26efefc5 291 av_set_pts_info(st, 32, 1, 90000);
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292 switch(st->codec->codec_id) {
293 case CODEC_ID_MPEG1VIDEO:
294 case CODEC_ID_MPEG2VIDEO:
295 case CODEC_ID_MP2:
296 case CODEC_ID_MP3:
297 case CODEC_ID_MPEG4:
298 case CODEC_ID_H264:
299 st->need_parsing = AVSTREAM_PARSE_FULL;
300 break;
301 default:
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302 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
303 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
304 }
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305 break;
306 }
307 }
308 // needed to send back RTCP RR in RTSP sessions
309 s->rtp_ctx = rtpc;
310 gethostname(s->hostname, sizeof(s->hostname));
311 return s;
312}
313
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314void
315rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
316 RTPDynamicProtocolHandler *handler)
317{
318 s->dynamic_protocol_context = ctx;
319 s->parse_packet = handler->parse_packet;
320}
321
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322static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
323{
324 int au_headers_length, au_header_size, i;
325 GetBitContext getbitcontext;
be73a544 326 RTPPayloadData *infos;
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327
328 infos = s->rtp_payload_data;
329
330 if (infos == NULL)
331 return -1;
332
bd107136 333 /* decode the first 2 bytes where the AUHeader sections are stored
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334 length in bits */
335 au_headers_length = AV_RB16(buf);
336
337 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
338 return -1;
339
340 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
341
342 /* skip AU headers length section (2 bytes) */
343 buf += 2;
344
345 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
346
347 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
348 au_header_size = infos->sizelength + infos->indexlength;
349 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
350 return -1;
351
352 infos->nb_au_headers = au_headers_length / au_header_size;
353 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
354
355 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
356 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
357 but does when sending the whole as one big packet... */
358 infos->au_headers[0].size = 0;
359 infos->au_headers[0].index = 0;
360 for (i = 0; i < infos->nb_au_headers; ++i) {
361 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
362 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
363 }
364
365 infos->nb_au_headers = 1;
366
367 return 0;
368}
369
370/**
371 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
372 */
373static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
374{
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375 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
376 int64_t addend;
377 int delta_timestamp;
378
379 /* compute pts from timestamp with received ntp_time */
380 delta_timestamp = timestamp - s->last_rtcp_timestamp;
381 /* convert to the PTS timebase */
382 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
383 pkt->pts = addend + delta_timestamp;
384 }
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385 pkt->stream_index = s->st->index;
386}
387
388/**
389 * Parse an RTP or RTCP packet directly sent as a buffer.
390 * @param s RTP parse context.
391 * @param pkt returned packet
392 * @param buf input buffer or NULL to read the next packets
393 * @param len buffer len
394 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
395 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
396 */
397int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
398 const uint8_t *buf, int len)
399{
400 unsigned int ssrc, h;
f841a0fc 401 int payload_type, seq, ret, flags = 0;
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402 AVStream *st;
403 uint32_t timestamp;
404 int rv= 0;
405
406 if (!buf) {
407 /* return the next packets, if any */
408 if(s->st && s->parse_packet) {
409 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
1a45a9f4 410 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 411 s->st, pkt, &timestamp, NULL, 0, flags);
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412 finalize_packet(s, pkt, timestamp);
413 return rv;
414 } else {
415 // TODO: Move to a dynamic packet handler (like above)
416 if (s->read_buf_index >= s->read_buf_size)
417 return -1;
418 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
419 s->read_buf_size - s->read_buf_index);
420 if (ret < 0)
421 return -1;
422 s->read_buf_index += ret;
423 if (s->read_buf_index < s->read_buf_size)
424 return 1;
425 else
426 return 0;
427 }
428 }
429
430 if (len < 12)
431 return -1;
432
433 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
434 return -1;
435 if (buf[1] >= 200 && buf[1] <= 204) {
436 rtcp_parse_packet(s, buf, len);
437 return -1;
438 }
439 payload_type = buf[1] & 0x7f;
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440 if (buf[1] & 0x80)
441 flags |= RTP_FLAG_MARKER;
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442 seq = AV_RB16(buf + 2);
443 timestamp = AV_RB32(buf + 4);
444 ssrc = AV_RB32(buf + 8);
445 /* store the ssrc in the RTPDemuxContext */
446 s->ssrc = ssrc;
447
448 /* NOTE: we can handle only one payload type */
449 if (s->payload_type != payload_type)
450 return -1;
451
452 st = s->st;
453 // only do something with this if all the rtp checks pass...
454 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
455 {
456 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
457 payload_type, seq, ((s->seq + 1) & 0xffff));
458 return -1;
459 }
460
461 s->seq = seq;
462 len -= 12;
463 buf += 12;
464
465 if (!st) {
466 /* specific MPEG2TS demux support */
467 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
468 if (ret < 0)
469 return -1;
470 if (ret < len) {
471 s->read_buf_size = len - ret;
472 memcpy(s->buf, buf + ret, s->read_buf_size);
473 s->read_buf_index = 0;
474 return 1;
475 }
b4e3330c 476 } else if (s->parse_packet) {
1a45a9f4 477 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 478 s->st, pkt, &timestamp, buf, len, flags);
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479 } else {
480 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
481 switch(st->codec->codec_id) {
482 case CODEC_ID_MP2:
483 /* better than nothing: skip mpeg audio RTP header */
484 if (len <= 4)
485 return -1;
486 h = AV_RB32(buf);
487 len -= 4;
488 buf += 4;
489 av_new_packet(pkt, len);
490 memcpy(pkt->data, buf, len);
491 break;
492 case CODEC_ID_MPEG1VIDEO:
493 case CODEC_ID_MPEG2VIDEO:
494 /* better than nothing: skip mpeg video RTP header */
495 if (len <= 4)
496 return -1;
497 h = AV_RB32(buf);
498 buf += 4;
499 len -= 4;
500 if (h & (1 << 26)) {
501 /* mpeg2 */
502 if (len <= 4)
503 return -1;
504 buf += 4;
505 len -= 4;
506 }
507 av_new_packet(pkt, len);
508 memcpy(pkt->data, buf, len);
509 break;
510 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
511 // timestamps.
512 // TODO: Put this into a dynamic packet handler...
513 case CODEC_ID_AAC:
514 if (rtp_parse_mp4_au(s, buf))
515 return -1;
516 {
be73a544 517 RTPPayloadData *infos = s->rtp_payload_data;
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518 if (infos == NULL)
519 return -1;
520 buf += infos->au_headers_length_bytes + 2;
521 len -= infos->au_headers_length_bytes + 2;
522
523 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
524 one au_header */
525 av_new_packet(pkt, infos->au_headers[0].size);
526 memcpy(pkt->data, buf, infos->au_headers[0].size);
527 buf += infos->au_headers[0].size;
528 len -= infos->au_headers[0].size;
529 }
530 s->read_buf_size = len;
531 rv= 0;
532 break;
533 default:
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534 av_new_packet(pkt, len);
535 memcpy(pkt->data, buf, len);
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536 break;
537 }
538
539 // now perform timestamp things....
540 finalize_packet(s, pkt, timestamp);
541 }
542 return rv;
543}
544
545void rtp_parse_close(RTPDemuxContext *s)
546{
547 // TODO: fold this into the protocol specific data fields.
548 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
549 mpegts_parse_close(s->ts);
550 }
551 av_free(s);
552}