Use parentheses around && within ||, fix the gcc warning:
[libav.git] / libavformat / rtpdec.c
CommitLineData
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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
9106a698 25#include "libavcodec/get_bits.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"
e9fce261 33#include "rtp_asf.h"
8eb793c4 34#include "rtp_h264.h"
e6327fba 35#include "rtp_vorbis.h"
45aa9080 36#include "rtpdec_h263.h"
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37
38//#define DEBUG
39
40/* TODO: - add RTCP statistics reporting (should be optional).
41
42 - add support for h263/mpeg4 packetized output : IDEA: send a
43 buffer to 'rtp_write_packet' contains all the packets for ONE
44 frame. Each packet should have a four byte header containing
45 the length in big endian format (same trick as
46 'url_open_dyn_packet_buf')
47*/
48
49/* statistics functions */
50RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
51
52static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
53static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
54
0369d2b0 55void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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56{
57 handler->next= RTPFirstDynamicPayloadHandler;
58 RTPFirstDynamicPayloadHandler= handler;
59}
60
61void av_register_rtp_dynamic_payload_handlers(void)
62{
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63 ff_register_dynamic_payload_handler(&mp4v_es_handler);
64 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
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65 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 67 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 68 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
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69
70 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
71 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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72}
73
74static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
75{
76 if (buf[1] != 200)
77 return -1;
78 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
79 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
80 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
81 s->last_rtcp_timestamp = AV_RB32(buf + 16);
82 return 0;
83}
84
85#define RTP_SEQ_MOD (1<<16)
86
87/**
88* called on parse open packet
89*/
90static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
91{
92 memset(s, 0, sizeof(RTPStatistics));
93 s->max_seq= base_sequence;
94 s->probation= 1;
95}
96
97/**
98* called whenever there is a large jump in sequence numbers, or when they get out of probation...
99*/
100static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
101{
102 s->max_seq= seq;
103 s->cycles= 0;
104 s->base_seq= seq -1;
105 s->bad_seq= RTP_SEQ_MOD + 1;
106 s->received= 0;
107 s->expected_prior= 0;
108 s->received_prior= 0;
109 s->jitter= 0;
110 s->transit= 0;
111}
112
113/**
114* returns 1 if we should handle this packet.
115*/
116static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
117{
118 uint16_t udelta= seq - s->max_seq;
119 const int MAX_DROPOUT= 3000;
120 const int MAX_MISORDER = 100;
121 const int MIN_SEQUENTIAL = 2;
122
123 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
124 if(s->probation)
125 {
126 if(seq==s->max_seq + 1) {
127 s->probation--;
128 s->max_seq= seq;
129 if(s->probation==0) {
130 rtp_init_sequence(s, seq);
131 s->received++;
132 return 1;
133 }
134 } else {
135 s->probation= MIN_SEQUENTIAL - 1;
136 s->max_seq = seq;
137 }
138 } else if (udelta < MAX_DROPOUT) {
139 // in order, with permissible gap
140 if(seq < s->max_seq) {
141 //sequence number wrapped; count antother 64k cycles
142 s->cycles += RTP_SEQ_MOD;
143 }
144 s->max_seq= seq;
145 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
146 // sequence made a large jump...
147 if(seq==s->bad_seq) {
148 // two sequential packets-- assume that the other side restarted without telling us; just resync.
149 rtp_init_sequence(s, seq);
150 } else {
151 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
152 return 0;
153 }
154 } else {
155 // duplicate or reordered packet...
156 }
157 s->received++;
158 return 1;
159}
160
161#if 0
162/**
163* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
164* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
165* never change. I left this in in case someone else can see a way. (rdm)
166*/
167static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
168{
169 uint32_t transit= arrival_timestamp - sent_timestamp;
170 int d;
171 s->transit= transit;
172 d= FFABS(transit - s->transit);
173 s->jitter += d - ((s->jitter + 8)>>4);
174}
175#endif
176
177int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
178{
179 ByteIOContext *pb;
180 uint8_t *buf;
181 int len;
182 int rtcp_bytes;
183 RTPStatistics *stats= &s->statistics;
184 uint32_t lost;
185 uint32_t extended_max;
186 uint32_t expected_interval;
187 uint32_t received_interval;
188 uint32_t lost_interval;
189 uint32_t expected;
190 uint32_t fraction;
191 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
192
193 if (!s->rtp_ctx || (count < 1))
194 return -1;
195
196 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
197 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
198 s->octet_count += count;
199 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
200 RTCP_TX_RATIO_DEN;
201 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
202 if (rtcp_bytes < 28)
203 return -1;
204 s->last_octet_count = s->octet_count;
205
206 if (url_open_dyn_buf(&pb) < 0)
207 return -1;
208
209 // Receiver Report
210 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
211 put_byte(pb, 201);
212 put_be16(pb, 7); /* length in words - 1 */
213 put_be32(pb, s->ssrc); // our own SSRC
214 put_be32(pb, s->ssrc); // XXX: should be the server's here!
215 // some placeholders we should really fill...
216 // RFC 1889/p64
217 extended_max= stats->cycles + stats->max_seq;
218 expected= extended_max - stats->base_seq + 1;
219 lost= expected - stats->received;
220 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
221 expected_interval= expected - stats->expected_prior;
222 stats->expected_prior= expected;
223 received_interval= stats->received - stats->received_prior;
224 stats->received_prior= stats->received;
225 lost_interval= expected_interval - received_interval;
226 if (expected_interval==0 || lost_interval<=0) fraction= 0;
227 else fraction = (lost_interval<<8)/expected_interval;
228
229 fraction= (fraction<<24) | lost;
230
231 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
232 put_be32(pb, extended_max); /* max sequence received */
233 put_be32(pb, stats->jitter>>4); /* jitter */
234
235 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
236 {
237 put_be32(pb, 0); /* last SR timestamp */
238 put_be32(pb, 0); /* delay since last SR */
239 } else {
240 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
241 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
242
243 put_be32(pb, middle_32_bits); /* last SR timestamp */
244 put_be32(pb, delay_since_last); /* delay since last SR */
245 }
246
247 // CNAME
248 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
249 put_byte(pb, 202);
250 len = strlen(s->hostname);
251 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
252 put_be32(pb, s->ssrc);
253 put_byte(pb, 0x01);
254 put_byte(pb, len);
255 put_buffer(pb, s->hostname, len);
256 // padding
257 for (len = (6 + len) % 4; len % 4; len++) {
258 put_byte(pb, 0);
259 }
260
261 put_flush_packet(pb);
262 len = url_close_dyn_buf(pb, &buf);
263 if ((len > 0) && buf) {
264 int result;
e8420626 265 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 266 result= url_write(s->rtp_ctx, buf, len);
e8420626 267 dprintf(s->ic, "result from url_write: %d\n", result);
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268 av_free(buf);
269 }
270 return 0;
271}
272
273/**
274 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
275 * MPEG2TS streams to indicate that they should be demuxed inside the
276 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
277 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
278 */
be73a544 279RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
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280{
281 RTPDemuxContext *s;
282
283 s = av_mallocz(sizeof(RTPDemuxContext));
284 if (!s)
285 return NULL;
286 s->payload_type = payload_type;
287 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
288 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
289 s->ic = s1;
290 s->st = st;
291 s->rtp_payload_data = rtp_payload_data;
292 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
293 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
294 s->ts = mpegts_parse_open(s->ic);
295 if (s->ts == NULL) {
296 av_free(s);
297 return NULL;
298 }
299 } else {
26efefc5 300 av_set_pts_info(st, 32, 1, 90000);
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301 switch(st->codec->codec_id) {
302 case CODEC_ID_MPEG1VIDEO:
303 case CODEC_ID_MPEG2VIDEO:
304 case CODEC_ID_MP2:
305 case CODEC_ID_MP3:
306 case CODEC_ID_MPEG4:
45aa9080 307 case CODEC_ID_H263:
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308 case CODEC_ID_H264:
309 st->need_parsing = AVSTREAM_PARSE_FULL;
310 break;
311 default:
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312 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
313 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
314 }
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315 break;
316 }
317 }
318 // needed to send back RTCP RR in RTSP sessions
319 s->rtp_ctx = rtpc;
320 gethostname(s->hostname, sizeof(s->hostname));
321 return s;
322}
323
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324void
325rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
326 RTPDynamicProtocolHandler *handler)
327{
328 s->dynamic_protocol_context = ctx;
329 s->parse_packet = handler->parse_packet;
330}
331
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332static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
333{
334 int au_headers_length, au_header_size, i;
335 GetBitContext getbitcontext;
be73a544 336 RTPPayloadData *infos;
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337
338 infos = s->rtp_payload_data;
339
340 if (infos == NULL)
341 return -1;
342
bd107136 343 /* decode the first 2 bytes where the AUHeader sections are stored
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344 length in bits */
345 au_headers_length = AV_RB16(buf);
346
347 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
348 return -1;
349
350 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
351
352 /* skip AU headers length section (2 bytes) */
353 buf += 2;
354
355 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
356
357 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
358 au_header_size = infos->sizelength + infos->indexlength;
359 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
360 return -1;
361
362 infos->nb_au_headers = au_headers_length / au_header_size;
363 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
364
365 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
366 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
367 but does when sending the whole as one big packet... */
368 infos->au_headers[0].size = 0;
369 infos->au_headers[0].index = 0;
370 for (i = 0; i < infos->nb_au_headers; ++i) {
371 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
372 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
373 }
374
375 infos->nb_au_headers = 1;
376
377 return 0;
378}
379
380/**
381 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
382 */
383static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
384{
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385 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
386 int64_t addend;
387 int delta_timestamp;
388
389 /* compute pts from timestamp with received ntp_time */
390 delta_timestamp = timestamp - s->last_rtcp_timestamp;
391 /* convert to the PTS timebase */
392 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);
393 pkt->pts = addend + delta_timestamp;
394 }
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395}
396
397/**
398 * Parse an RTP or RTCP packet directly sent as a buffer.
399 * @param s RTP parse context.
400 * @param pkt returned packet
401 * @param buf input buffer or NULL to read the next packets
402 * @param len buffer len
403 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
404 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
405 */
406int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
407 const uint8_t *buf, int len)
408{
409 unsigned int ssrc, h;
f841a0fc 410 int payload_type, seq, ret, flags = 0;
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411 AVStream *st;
412 uint32_t timestamp;
413 int rv= 0;
414
415 if (!buf) {
416 /* return the next packets, if any */
417 if(s->st && s->parse_packet) {
418 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
1a45a9f4 419 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 420 s->st, pkt, &timestamp, NULL, 0, flags);
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421 finalize_packet(s, pkt, timestamp);
422 return rv;
423 } else {
424 // TODO: Move to a dynamic packet handler (like above)
425 if (s->read_buf_index >= s->read_buf_size)
426 return -1;
427 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
428 s->read_buf_size - s->read_buf_index);
429 if (ret < 0)
430 return -1;
431 s->read_buf_index += ret;
432 if (s->read_buf_index < s->read_buf_size)
433 return 1;
434 else
435 return 0;
436 }
437 }
438
439 if (len < 12)
440 return -1;
441
442 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
443 return -1;
444 if (buf[1] >= 200 && buf[1] <= 204) {
445 rtcp_parse_packet(s, buf, len);
446 return -1;
447 }
448 payload_type = buf[1] & 0x7f;
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449 if (buf[1] & 0x80)
450 flags |= RTP_FLAG_MARKER;
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451 seq = AV_RB16(buf + 2);
452 timestamp = AV_RB32(buf + 4);
453 ssrc = AV_RB32(buf + 8);
454 /* store the ssrc in the RTPDemuxContext */
455 s->ssrc = ssrc;
456
457 /* NOTE: we can handle only one payload type */
458 if (s->payload_type != payload_type)
459 return -1;
460
461 st = s->st;
462 // only do something with this if all the rtp checks pass...
463 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
464 {
465 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
466 payload_type, seq, ((s->seq + 1) & 0xffff));
467 return -1;
468 }
469
470 s->seq = seq;
471 len -= 12;
472 buf += 12;
473
474 if (!st) {
475 /* specific MPEG2TS demux support */
476 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
477 if (ret < 0)
478 return -1;
479 if (ret < len) {
480 s->read_buf_size = len - ret;
481 memcpy(s->buf, buf + ret, s->read_buf_size);
482 s->read_buf_index = 0;
483 return 1;
484 }
f3e71942 485 return 0;
b4e3330c 486 } else if (s->parse_packet) {
1a45a9f4 487 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 488 s->st, pkt, &timestamp, buf, len, flags);
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489 } else {
490 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
491 switch(st->codec->codec_id) {
492 case CODEC_ID_MP2:
76faff6e 493 case CODEC_ID_MP3:
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494 /* better than nothing: skip mpeg audio RTP header */
495 if (len <= 4)
496 return -1;
497 h = AV_RB32(buf);
498 len -= 4;
499 buf += 4;
500 av_new_packet(pkt, len);
501 memcpy(pkt->data, buf, len);
502 break;
503 case CODEC_ID_MPEG1VIDEO:
504 case CODEC_ID_MPEG2VIDEO:
505 /* better than nothing: skip mpeg video RTP header */
506 if (len <= 4)
507 return -1;
508 h = AV_RB32(buf);
509 buf += 4;
510 len -= 4;
511 if (h & (1 << 26)) {
512 /* mpeg2 */
513 if (len <= 4)
514 return -1;
515 buf += 4;
516 len -= 4;
517 }
518 av_new_packet(pkt, len);
519 memcpy(pkt->data, buf, len);
520 break;
521 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
522 // timestamps.
523 // TODO: Put this into a dynamic packet handler...
524 case CODEC_ID_AAC:
525 if (rtp_parse_mp4_au(s, buf))
526 return -1;
527 {
be73a544 528 RTPPayloadData *infos = s->rtp_payload_data;
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529 if (infos == NULL)
530 return -1;
531 buf += infos->au_headers_length_bytes + 2;
532 len -= infos->au_headers_length_bytes + 2;
533
534 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
535 one au_header */
536 av_new_packet(pkt, infos->au_headers[0].size);
537 memcpy(pkt->data, buf, infos->au_headers[0].size);
538 buf += infos->au_headers[0].size;
539 len -= infos->au_headers[0].size;
540 }
541 s->read_buf_size = len;
542 rv= 0;
543 break;
544 default:
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545 av_new_packet(pkt, len);
546 memcpy(pkt->data, buf, len);
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547 break;
548 }
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549
550 pkt->stream_index = st->index;
f3e71942 551 }
8eb793c4 552
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553 // now perform timestamp things....
554 finalize_packet(s, pkt, timestamp);
f3e71942 555
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556 return rv;
557}
558
559void rtp_parse_close(RTPDemuxContext *s)
560{
561 // TODO: fold this into the protocol specific data fields.
562 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
563 mpegts_parse_close(s->ts);
564 }
565 av_free(s);
566}