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[libav.git] / libavformat / rtpdec.c
1 /*
2 * RTP input 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
22 #include "libavcodec/bitstream.h"
23 #include "avformat.h"
24 #include "mpegts.h"
25
26 #include <unistd.h>
27 #include "network.h"
28
29 #include "rtp_internal.h"
30 #include "rtp_h264.h"
31
32 //#define DEBUG
33
34 /* TODO: - add RTCP statistics reporting (should be optional).
35
36 - add support for h263/mpeg4 packetized output : IDEA: send a
37 buffer to 'rtp_write_packet' contains all the packets for ONE
38 frame. Each packet should have a four byte header containing
39 the length in big endian format (same trick as
40 'url_open_dyn_packet_buf')
41 */
42
43 /* statistics functions */
44 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
45
46 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
47 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
48
49 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
50 {
51 handler->next= RTPFirstDynamicPayloadHandler;
52 RTPFirstDynamicPayloadHandler= handler;
53 }
54
55 void av_register_rtp_dynamic_payload_handlers(void)
56 {
57 register_dynamic_payload_handler(&mp4v_es_handler);
58 register_dynamic_payload_handler(&mpeg4_generic_handler);
59 register_dynamic_payload_handler(&ff_h264_dynamic_handler);
60 }
61
62 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
63 {
64 if (buf[1] != 200)
65 return -1;
66 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
67 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
68 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
69 s->last_rtcp_timestamp = AV_RB32(buf + 16);
70 return 0;
71 }
72
73 #define RTP_SEQ_MOD (1<<16)
74
75 /**
76 * called on parse open packet
77 */
78 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
79 {
80 memset(s, 0, sizeof(RTPStatistics));
81 s->max_seq= base_sequence;
82 s->probation= 1;
83 }
84
85 /**
86 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
87 */
88 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
89 {
90 s->max_seq= seq;
91 s->cycles= 0;
92 s->base_seq= seq -1;
93 s->bad_seq= RTP_SEQ_MOD + 1;
94 s->received= 0;
95 s->expected_prior= 0;
96 s->received_prior= 0;
97 s->jitter= 0;
98 s->transit= 0;
99 }
100
101 /**
102 * returns 1 if we should handle this packet.
103 */
104 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
105 {
106 uint16_t udelta= seq - s->max_seq;
107 const int MAX_DROPOUT= 3000;
108 const int MAX_MISORDER = 100;
109 const int MIN_SEQUENTIAL = 2;
110
111 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
112 if(s->probation)
113 {
114 if(seq==s->max_seq + 1) {
115 s->probation--;
116 s->max_seq= seq;
117 if(s->probation==0) {
118 rtp_init_sequence(s, seq);
119 s->received++;
120 return 1;
121 }
122 } else {
123 s->probation= MIN_SEQUENTIAL - 1;
124 s->max_seq = seq;
125 }
126 } else if (udelta < MAX_DROPOUT) {
127 // in order, with permissible gap
128 if(seq < s->max_seq) {
129 //sequence number wrapped; count antother 64k cycles
130 s->cycles += RTP_SEQ_MOD;
131 }
132 s->max_seq= seq;
133 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
134 // sequence made a large jump...
135 if(seq==s->bad_seq) {
136 // two sequential packets-- assume that the other side restarted without telling us; just resync.
137 rtp_init_sequence(s, seq);
138 } else {
139 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
140 return 0;
141 }
142 } else {
143 // duplicate or reordered packet...
144 }
145 s->received++;
146 return 1;
147 }
148
149 #if 0
150 /**
151 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
152 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
153 * never change. I left this in in case someone else can see a way. (rdm)
154 */
155 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
156 {
157 uint32_t transit= arrival_timestamp - sent_timestamp;
158 int d;
159 s->transit= transit;
160 d= FFABS(transit - s->transit);
161 s->jitter += d - ((s->jitter + 8)>>4);
162 }
163 #endif
164
165 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
166 {
167 ByteIOContext *pb;
168 uint8_t *buf;
169 int len;
170 int rtcp_bytes;
171 RTPStatistics *stats= &s->statistics;
172 uint32_t lost;
173 uint32_t extended_max;
174 uint32_t expected_interval;
175 uint32_t received_interval;
176 uint32_t lost_interval;
177 uint32_t expected;
178 uint32_t fraction;
179 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
180
181 if (!s->rtp_ctx || (count < 1))
182 return -1;
183
184 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
185 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
186 s->octet_count += count;
187 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
188 RTCP_TX_RATIO_DEN;
189 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
190 if (rtcp_bytes < 28)
191 return -1;
192 s->last_octet_count = s->octet_count;
193
194 if (url_open_dyn_buf(&pb) < 0)
195 return -1;
196
197 // Receiver Report
198 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
199 put_byte(pb, 201);
200 put_be16(pb, 7); /* length in words - 1 */
201 put_be32(pb, s->ssrc); // our own SSRC
202 put_be32(pb, s->ssrc); // XXX: should be the server's here!
203 // some placeholders we should really fill...
204 // RFC 1889/p64
205 extended_max= stats->cycles + stats->max_seq;
206 expected= extended_max - stats->base_seq + 1;
207 lost= expected - stats->received;
208 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
209 expected_interval= expected - stats->expected_prior;
210 stats->expected_prior= expected;
211 received_interval= stats->received - stats->received_prior;
212 stats->received_prior= stats->received;
213 lost_interval= expected_interval - received_interval;
214 if (expected_interval==0 || lost_interval<=0) fraction= 0;
215 else fraction = (lost_interval<<8)/expected_interval;
216
217 fraction= (fraction<<24) | lost;
218
219 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
220 put_be32(pb, extended_max); /* max sequence received */
221 put_be32(pb, stats->jitter>>4); /* jitter */
222
223 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
224 {
225 put_be32(pb, 0); /* last SR timestamp */
226 put_be32(pb, 0); /* delay since last SR */
227 } else {
228 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
229 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
230
231 put_be32(pb, middle_32_bits); /* last SR timestamp */
232 put_be32(pb, delay_since_last); /* delay since last SR */
233 }
234
235 // CNAME
236 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
237 put_byte(pb, 202);
238 len = strlen(s->hostname);
239 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
240 put_be32(pb, s->ssrc);
241 put_byte(pb, 0x01);
242 put_byte(pb, len);
243 put_buffer(pb, s->hostname, len);
244 // padding
245 for (len = (6 + len) % 4; len % 4; len++) {
246 put_byte(pb, 0);
247 }
248
249 put_flush_packet(pb);
250 len = url_close_dyn_buf(pb, &buf);
251 if ((len > 0) && buf) {
252 int result;
253 #if defined(DEBUG)
254 printf("sending %d bytes of RR\n", len);
255 #endif
256 result= url_write(s->rtp_ctx, buf, len);
257 #if defined(DEBUG)
258 printf("result from url_write: %d\n", result);
259 #endif
260 av_free(buf);
261 }
262 return 0;
263 }
264
265 /**
266 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
267 * MPEG2TS streams to indicate that they should be demuxed inside the
268 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
269 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
270 */
271 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
272 {
273 RTPDemuxContext *s;
274
275 s = av_mallocz(sizeof(RTPDemuxContext));
276 if (!s)
277 return NULL;
278 s->payload_type = payload_type;
279 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
280 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
281 s->ic = s1;
282 s->st = st;
283 s->rtp_payload_data = rtp_payload_data;
284 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
285 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
286 s->ts = mpegts_parse_open(s->ic);
287 if (s->ts == NULL) {
288 av_free(s);
289 return NULL;
290 }
291 } else {
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:
302 break;
303 }
304 }
305 // needed to send back RTCP RR in RTSP sessions
306 s->rtp_ctx = rtpc;
307 gethostname(s->hostname, sizeof(s->hostname));
308 return s;
309 }
310
311 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
312 {
313 int au_headers_length, au_header_size, i;
314 GetBitContext getbitcontext;
315 rtp_payload_data_t *infos;
316
317 infos = s->rtp_payload_data;
318
319 if (infos == NULL)
320 return -1;
321
322 /* decode the first 2 bytes where the AUHeader sections are stored
323 length in bits */
324 au_headers_length = AV_RB16(buf);
325
326 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
327 return -1;
328
329 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
330
331 /* skip AU headers length section (2 bytes) */
332 buf += 2;
333
334 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
335
336 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
337 au_header_size = infos->sizelength + infos->indexlength;
338 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
339 return -1;
340
341 infos->nb_au_headers = au_headers_length / au_header_size;
342 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
343
344 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
345 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
346 but does when sending the whole as one big packet... */
347 infos->au_headers[0].size = 0;
348 infos->au_headers[0].index = 0;
349 for (i = 0; i < infos->nb_au_headers; ++i) {
350 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
351 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
352 }
353
354 infos->nb_au_headers = 1;
355
356 return 0;
357 }
358
359 /**
360 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
361 */
362 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
363 {
364 switch(s->st->codec->codec_id) {
365 case CODEC_ID_MP2:
366 case CODEC_ID_MPEG1VIDEO:
367 case CODEC_ID_MPEG2VIDEO:
368 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
369 int64_t addend;
370
371 int delta_timestamp;
372 /* XXX: is it really necessary to unify the timestamp base ? */
373 /* compute pts from timestamp with received ntp_time */
374 delta_timestamp = timestamp - s->last_rtcp_timestamp;
375 /* convert to 90 kHz without overflow */
376 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
377 addend = (addend * 5625) >> 14;
378 pkt->pts = addend + delta_timestamp;
379 }
380 break;
381 case CODEC_ID_AAC:
382 case CODEC_ID_H264:
383 case CODEC_ID_MPEG4:
384 pkt->pts = timestamp;
385 break;
386 default:
387 /* no timestamp info yet */
388 break;
389 }
390 pkt->stream_index = s->st->index;
391 }
392
393 /**
394 * Parse an RTP or RTCP packet directly sent as a buffer.
395 * @param s RTP parse context.
396 * @param pkt returned packet
397 * @param buf input buffer or NULL to read the next packets
398 * @param len buffer len
399 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
400 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
401 */
402 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
403 const uint8_t *buf, int len)
404 {
405 unsigned int ssrc, h;
406 int payload_type, seq, ret, flags = 0;
407 AVStream *st;
408 uint32_t timestamp;
409 int rv= 0;
410
411 if (!buf) {
412 /* return the next packets, if any */
413 if(s->st && s->parse_packet) {
414 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
415 rv= s->parse_packet(s, pkt, &timestamp, NULL, 0, flags);
416 finalize_packet(s, pkt, timestamp);
417 return rv;
418 } else {
419 // TODO: Move to a dynamic packet handler (like above)
420 if (s->read_buf_index >= s->read_buf_size)
421 return -1;
422 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
423 s->read_buf_size - s->read_buf_index);
424 if (ret < 0)
425 return -1;
426 s->read_buf_index += ret;
427 if (s->read_buf_index < s->read_buf_size)
428 return 1;
429 else
430 return 0;
431 }
432 }
433
434 if (len < 12)
435 return -1;
436
437 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
438 return -1;
439 if (buf[1] >= 200 && buf[1] <= 204) {
440 rtcp_parse_packet(s, buf, len);
441 return -1;
442 }
443 payload_type = buf[1] & 0x7f;
444 seq = AV_RB16(buf + 2);
445 timestamp = AV_RB32(buf + 4);
446 ssrc = AV_RB32(buf + 8);
447 /* store the ssrc in the RTPDemuxContext */
448 s->ssrc = ssrc;
449
450 /* NOTE: we can handle only one payload type */
451 if (s->payload_type != payload_type)
452 return -1;
453
454 st = s->st;
455 // only do something with this if all the rtp checks pass...
456 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
457 {
458 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
459 payload_type, seq, ((s->seq + 1) & 0xffff));
460 return -1;
461 }
462
463 s->seq = seq;
464 len -= 12;
465 buf += 12;
466
467 if (!st) {
468 /* specific MPEG2TS demux support */
469 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
470 if (ret < 0)
471 return -1;
472 if (ret < len) {
473 s->read_buf_size = len - ret;
474 memcpy(s->buf, buf + ret, s->read_buf_size);
475 s->read_buf_index = 0;
476 return 1;
477 }
478 } else if (s->parse_packet) {
479 rv = s->parse_packet(s, pkt, &timestamp, buf, len, flags);
480 } else {
481 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
482 switch(st->codec->codec_id) {
483 case CODEC_ID_MP2:
484 /* better than nothing: skip mpeg audio RTP header */
485 if (len <= 4)
486 return -1;
487 h = AV_RB32(buf);
488 len -= 4;
489 buf += 4;
490 av_new_packet(pkt, len);
491 memcpy(pkt->data, buf, len);
492 break;
493 case CODEC_ID_MPEG1VIDEO:
494 case CODEC_ID_MPEG2VIDEO:
495 /* better than nothing: skip mpeg video RTP header */
496 if (len <= 4)
497 return -1;
498 h = AV_RB32(buf);
499 buf += 4;
500 len -= 4;
501 if (h & (1 << 26)) {
502 /* mpeg2 */
503 if (len <= 4)
504 return -1;
505 buf += 4;
506 len -= 4;
507 }
508 av_new_packet(pkt, len);
509 memcpy(pkt->data, buf, len);
510 break;
511 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
512 // timestamps.
513 // TODO: Put this into a dynamic packet handler...
514 case CODEC_ID_AAC:
515 if (rtp_parse_mp4_au(s, buf))
516 return -1;
517 {
518 rtp_payload_data_t *infos = s->rtp_payload_data;
519 if (infos == NULL)
520 return -1;
521 buf += infos->au_headers_length_bytes + 2;
522 len -= infos->au_headers_length_bytes + 2;
523
524 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
525 one au_header */
526 av_new_packet(pkt, infos->au_headers[0].size);
527 memcpy(pkt->data, buf, infos->au_headers[0].size);
528 buf += infos->au_headers[0].size;
529 len -= infos->au_headers[0].size;
530 }
531 s->read_buf_size = len;
532 rv= 0;
533 break;
534 default:
535 av_new_packet(pkt, len);
536 memcpy(pkt->data, buf, len);
537 break;
538 }
539
540 // now perform timestamp things....
541 finalize_packet(s, pkt, timestamp);
542 }
543 return rv;
544 }
545
546 void rtp_parse_close(RTPDemuxContext *s)
547 {
548 // TODO: fold this into the protocol specific data fields.
549 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
550 mpegts_parse_close(s->ts);
551 }
552 av_free(s);
553 }