ffserver: use a local MAX_STREAMS limit
[libav.git] / libavformat / rtpdec.c
CommitLineData
8eb793c4
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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"
965a3ddb 33#include "rtpdec_formats.h"
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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
0369d2b0 49void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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50{
51 handler->next= RTPFirstDynamicPayloadHandler;
52 RTPFirstDynamicPayloadHandler= handler;
53}
54
55void av_register_rtp_dynamic_payload_handlers(void)
56{
9b3788ef
JA
57 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
58 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
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59 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
60 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
45aa9080
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61 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
62 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 63 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 64 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
887af2aa 65 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
a59096e4 66 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
4449df6b 67 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
1ddc176e 68 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
51291e60 69 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
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70
71 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
72 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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73
74 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
75 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
76 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
77 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
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78}
79
80static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
81{
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82 int payload_len;
83 while (len >= 2) {
84 switch (buf[1]) {
85 case RTCP_SR:
86 if (len < 16) {
87 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
88 return AVERROR_INVALIDDATA;
89 }
90 payload_len = (AV_RB16(buf + 2) + 1) * 4;
91
682d28a9
JA
92 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
93 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
94 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
95 s->last_rtcp_timestamp = AV_RB32(buf + 16);
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96
97 buf += payload_len;
98 len -= payload_len;
99 break;
b20359f5
JA
100 case RTCP_BYE:
101 return -RTCP_BYE;
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102 default:
103 return -1;
104 }
105 }
b20359f5 106 return -1;
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107}
108
109#define RTP_SEQ_MOD (1<<16)
110
111/**
112* called on parse open packet
113*/
114static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
115{
116 memset(s, 0, sizeof(RTPStatistics));
117 s->max_seq= base_sequence;
118 s->probation= 1;
119}
120
121/**
122* called whenever there is a large jump in sequence numbers, or when they get out of probation...
123*/
124static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
125{
126 s->max_seq= seq;
127 s->cycles= 0;
128 s->base_seq= seq -1;
129 s->bad_seq= RTP_SEQ_MOD + 1;
130 s->received= 0;
131 s->expected_prior= 0;
132 s->received_prior= 0;
133 s->jitter= 0;
134 s->transit= 0;
135}
136
137/**
138* returns 1 if we should handle this packet.
139*/
140static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
141{
142 uint16_t udelta= seq - s->max_seq;
143 const int MAX_DROPOUT= 3000;
144 const int MAX_MISORDER = 100;
145 const int MIN_SEQUENTIAL = 2;
146
147 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
148 if(s->probation)
149 {
150 if(seq==s->max_seq + 1) {
151 s->probation--;
152 s->max_seq= seq;
153 if(s->probation==0) {
154 rtp_init_sequence(s, seq);
155 s->received++;
156 return 1;
157 }
158 } else {
159 s->probation= MIN_SEQUENTIAL - 1;
160 s->max_seq = seq;
161 }
162 } else if (udelta < MAX_DROPOUT) {
163 // in order, with permissible gap
164 if(seq < s->max_seq) {
165 //sequence number wrapped; count antother 64k cycles
166 s->cycles += RTP_SEQ_MOD;
167 }
168 s->max_seq= seq;
169 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
170 // sequence made a large jump...
171 if(seq==s->bad_seq) {
172 // two sequential packets-- assume that the other side restarted without telling us; just resync.
173 rtp_init_sequence(s, seq);
174 } else {
175 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
176 return 0;
177 }
178 } else {
179 // duplicate or reordered packet...
180 }
181 s->received++;
182 return 1;
183}
184
185#if 0
186/**
187* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
188* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
189* never change. I left this in in case someone else can see a way. (rdm)
190*/
191static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
192{
193 uint32_t transit= arrival_timestamp - sent_timestamp;
194 int d;
195 s->transit= transit;
196 d= FFABS(transit - s->transit);
197 s->jitter += d - ((s->jitter + 8)>>4);
198}
199#endif
200
201int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
202{
203 ByteIOContext *pb;
204 uint8_t *buf;
205 int len;
206 int rtcp_bytes;
207 RTPStatistics *stats= &s->statistics;
208 uint32_t lost;
209 uint32_t extended_max;
210 uint32_t expected_interval;
211 uint32_t received_interval;
212 uint32_t lost_interval;
213 uint32_t expected;
214 uint32_t fraction;
215 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
216
217 if (!s->rtp_ctx || (count < 1))
218 return -1;
219
220 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
221 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
222 s->octet_count += count;
223 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
224 RTCP_TX_RATIO_DEN;
225 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
226 if (rtcp_bytes < 28)
227 return -1;
228 s->last_octet_count = s->octet_count;
229
230 if (url_open_dyn_buf(&pb) < 0)
231 return -1;
232
233 // Receiver Report
234 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 235 put_byte(pb, RTCP_RR);
8eb793c4 236 put_be16(pb, 7); /* length in words - 1 */
952139a3
LA
237 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
238 put_be32(pb, s->ssrc + 1);
239 put_be32(pb, s->ssrc); // server SSRC
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240 // some placeholders we should really fill...
241 // RFC 1889/p64
242 extended_max= stats->cycles + stats->max_seq;
243 expected= extended_max - stats->base_seq + 1;
244 lost= expected - stats->received;
245 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
246 expected_interval= expected - stats->expected_prior;
247 stats->expected_prior= expected;
248 received_interval= stats->received - stats->received_prior;
249 stats->received_prior= stats->received;
250 lost_interval= expected_interval - received_interval;
251 if (expected_interval==0 || lost_interval<=0) fraction= 0;
252 else fraction = (lost_interval<<8)/expected_interval;
253
254 fraction= (fraction<<24) | lost;
255
256 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
257 put_be32(pb, extended_max); /* max sequence received */
258 put_be32(pb, stats->jitter>>4); /* jitter */
259
260 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
261 {
262 put_be32(pb, 0); /* last SR timestamp */
263 put_be32(pb, 0); /* delay since last SR */
264 } else {
265 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
266 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
267
268 put_be32(pb, middle_32_bits); /* last SR timestamp */
269 put_be32(pb, delay_since_last); /* delay since last SR */
270 }
271
272 // CNAME
273 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 274 put_byte(pb, RTCP_SDES);
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275 len = strlen(s->hostname);
276 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
277 put_be32(pb, s->ssrc);
278 put_byte(pb, 0x01);
279 put_byte(pb, len);
280 put_buffer(pb, s->hostname, len);
281 // padding
282 for (len = (6 + len) % 4; len % 4; len++) {
283 put_byte(pb, 0);
284 }
285
286 put_flush_packet(pb);
287 len = url_close_dyn_buf(pb, &buf);
288 if ((len > 0) && buf) {
289 int result;
e8420626 290 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 291 result= url_write(s->rtp_ctx, buf, len);
e8420626 292 dprintf(s->ic, "result from url_write: %d\n", result);
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293 av_free(buf);
294 }
295 return 0;
296}
297
9c8fa20d
MS
298void rtp_send_punch_packets(URLContext* rtp_handle)
299{
300 ByteIOContext *pb;
301 uint8_t *buf;
302 int len;
303
304 /* Send a small RTP packet */
305 if (url_open_dyn_buf(&pb) < 0)
306 return;
307
308 put_byte(pb, (RTP_VERSION << 6));
309 put_byte(pb, 0); /* Payload type */
310 put_be16(pb, 0); /* Seq */
311 put_be32(pb, 0); /* Timestamp */
312 put_be32(pb, 0); /* SSRC */
313
314 put_flush_packet(pb);
315 len = url_close_dyn_buf(pb, &buf);
316 if ((len > 0) && buf)
317 url_write(rtp_handle, buf, len);
318 av_free(buf);
319
320 /* Send a minimal RTCP RR */
321 if (url_open_dyn_buf(&pb) < 0)
322 return;
323
324 put_byte(pb, (RTP_VERSION << 6));
7f3468d3 325 put_byte(pb, RTCP_RR); /* receiver report */
9c8fa20d
MS
326 put_be16(pb, 1); /* length in words - 1 */
327 put_be32(pb, 0); /* our own SSRC */
328
329 put_flush_packet(pb);
330 len = url_close_dyn_buf(pb, &buf);
331 if ((len > 0) && buf)
332 url_write(rtp_handle, buf, len);
333 av_free(buf);
334}
335
336
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337/**
338 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
339 * MPEG2TS streams to indicate that they should be demuxed inside the
340 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 341 */
58ee0991 342RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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LA
343{
344 RTPDemuxContext *s;
345
346 s = av_mallocz(sizeof(RTPDemuxContext));
347 if (!s)
348 return NULL;
349 s->payload_type = payload_type;
350 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 351 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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352 s->ic = s1;
353 s->st = st;
58ee0991 354 s->queue_size = queue_size;
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355 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
356 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 357 s->ts = ff_mpegts_parse_open(s->ic);
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LA
358 if (s->ts == NULL) {
359 av_free(s);
360 return NULL;
361 }
362 } else {
26efefc5 363 av_set_pts_info(st, 32, 1, 90000);
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LA
364 switch(st->codec->codec_id) {
365 case CODEC_ID_MPEG1VIDEO:
366 case CODEC_ID_MPEG2VIDEO:
367 case CODEC_ID_MP2:
368 case CODEC_ID_MP3:
369 case CODEC_ID_MPEG4:
45aa9080 370 case CODEC_ID_H263:
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LA
371 case CODEC_ID_H264:
372 st->need_parsing = AVSTREAM_PARSE_FULL;
373 break;
0048a2a8
MS
374 case CODEC_ID_ADPCM_G722:
375 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
376 /* According to RFC 3551, the stream clock rate is 8000
377 * even if the sample rate is 16000. */
378 if (st->codec->sample_rate == 8000)
379 st->codec->sample_rate = 16000;
380 break;
8eb793c4 381 default:
72415b2a 382 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
d6b9e57a
LA
383 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
384 }
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385 break;
386 }
387 }
388 // needed to send back RTCP RR in RTSP sessions
389 s->rtp_ctx = rtpc;
390 gethostname(s->hostname, sizeof(s->hostname));
391 return s;
392}
393
99a1d191
RB
394void
395rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
396 RTPDynamicProtocolHandler *handler)
397{
398 s->dynamic_protocol_context = ctx;
399 s->parse_packet = handler->parse_packet;
400}
401
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402/**
403 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
404 */
405static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
406{
d74c6145 407 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
fba7815d
LA
408 int64_t addend;
409 int delta_timestamp;
410
411 /* compute pts from timestamp with received ntp_time */
412 delta_timestamp = timestamp - s->last_rtcp_timestamp;
413 /* convert to the PTS timebase */
2cab6b48 414 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);
5948f822 415 pkt->pts = s->range_start_offset + addend + delta_timestamp;
fba7815d 416 }
8eb793c4
LA
417}
418
02607418
MS
419static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
420 const uint8_t *buf, int len)
8eb793c4
LA
421{
422 unsigned int ssrc, h;
f841a0fc 423 int payload_type, seq, ret, flags = 0;
9446b4bb 424 int ext;
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LA
425 AVStream *st;
426 uint32_t timestamp;
427 int rv= 0;
428
9446b4bb 429 ext = buf[0] & 0x10;
8eb793c4 430 payload_type = buf[1] & 0x7f;
144ae29d
RB
431 if (buf[1] & 0x80)
432 flags |= RTP_FLAG_MARKER;
8eb793c4
LA
433 seq = AV_RB16(buf + 2);
434 timestamp = AV_RB32(buf + 4);
435 ssrc = AV_RB32(buf + 8);
436 /* store the ssrc in the RTPDemuxContext */
437 s->ssrc = ssrc;
438
439 /* NOTE: we can handle only one payload type */
440 if (s->payload_type != payload_type)
441 return -1;
442
443 st = s->st;
444 // only do something with this if all the rtp checks pass...
445 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
446 {
447 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
448 payload_type, seq, ((s->seq + 1) & 0xffff));
449 return -1;
450 }
451
452 s->seq = seq;
453 len -= 12;
454 buf += 12;
455
9446b4bb
RS
456 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
457 if (ext) {
458 if (len < 4)
459 return -1;
460 /* calculate the header extension length (stored as number
461 * of 32-bit words) */
462 ext = (AV_RB16(buf + 2) + 1) << 2;
463
464 if (len < ext)
465 return -1;
466 // skip past RTP header extension
467 len -= ext;
468 buf += ext;
469 }
470
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471 if (!st) {
472 /* specific MPEG2TS demux support */
9125806e 473 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
4ffff367 474 if (ret < 0)
91ec7aea 475 return ret;
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LA
476 if (ret < len) {
477 s->read_buf_size = len - ret;
478 memcpy(s->buf, buf + ret, s->read_buf_size);
479 s->read_buf_index = 0;
480 return 1;
481 }
f3e71942 482 return 0;
b4e3330c 483 } else if (s->parse_packet) {
1a45a9f4 484 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 485 s->st, pkt, &timestamp, buf, len, flags);
8eb793c4
LA
486 } else {
487 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
488 switch(st->codec->codec_id) {
489 case CODEC_ID_MP2:
76faff6e 490 case CODEC_ID_MP3:
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LA
491 /* better than nothing: skip mpeg audio RTP header */
492 if (len <= 4)
493 return -1;
494 h = AV_RB32(buf);
495 len -= 4;
496 buf += 4;
497 av_new_packet(pkt, len);
498 memcpy(pkt->data, buf, len);
499 break;
500 case CODEC_ID_MPEG1VIDEO:
501 case CODEC_ID_MPEG2VIDEO:
502 /* better than nothing: skip mpeg video RTP header */
503 if (len <= 4)
504 return -1;
505 h = AV_RB32(buf);
506 buf += 4;
507 len -= 4;
508 if (h & (1 << 26)) {
509 /* mpeg2 */
510 if (len <= 4)
511 return -1;
512 buf += 4;
513 len -= 4;
514 }
515 av_new_packet(pkt, len);
516 memcpy(pkt->data, buf, len);
517 break;
8eb793c4 518 default:
f739b36d
RB
519 av_new_packet(pkt, len);
520 memcpy(pkt->data, buf, len);
8eb793c4
LA
521 break;
522 }
eafb17d1
RB
523
524 pkt->stream_index = st->index;
f3e71942 525 }
8eb793c4 526
95f03cf3
RB
527 // now perform timestamp things....
528 finalize_packet(s, pkt, timestamp);
f3e71942 529
8eb793c4
LA
530 return rv;
531}
532
58ee0991
MS
533void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
534{
535 while (s->queue) {
536 RTPPacket *next = s->queue->next;
537 av_free(s->queue->buf);
538 av_free(s->queue);
539 s->queue = next;
540 }
541 s->seq = 0;
542 s->queue_len = 0;
543 s->prev_ret = 0;
544}
545
546static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
547{
548 uint16_t seq = AV_RB16(buf + 2);
549 RTPPacket *cur = s->queue, *prev = NULL, *packet;
550
551 /* Find the correct place in the queue to insert the packet */
552 while (cur) {
553 int16_t diff = seq - cur->seq;
554 if (diff < 0)
555 break;
556 prev = cur;
557 cur = cur->next;
558 }
559
560 packet = av_mallocz(sizeof(*packet));
561 if (!packet)
562 return;
563 packet->recvtime = av_gettime();
564 packet->seq = seq;
565 packet->len = len;
566 packet->buf = buf;
567 packet->next = cur;
568 if (prev)
569 prev->next = packet;
570 else
571 s->queue = packet;
572 s->queue_len++;
573}
574
575static int has_next_packet(RTPDemuxContext *s)
576{
ddcf8411 577 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
58ee0991
MS
578}
579
580int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
581{
582 return s->queue ? s->queue->recvtime : 0;
583}
584
585static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
586{
587 int rv;
588 RTPPacket *next;
589
590 if (s->queue_len <= 0)
591 return -1;
592
593 if (!has_next_packet(s))
594 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
595 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
596
597 /* Parse the first packet in the queue, and dequeue it */
598 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
599 next = s->queue->next;
600 av_free(s->queue->buf);
601 av_free(s->queue);
602 s->queue = next;
603 s->queue_len--;
4ffff367 604 return rv;
58ee0991
MS
605}
606
4ffff367 607static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
02607418
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608 uint8_t **bufptr, int len)
609{
610 uint8_t* buf = bufptr ? *bufptr : NULL;
611 int ret, flags = 0;
612 uint32_t timestamp;
613 int rv= 0;
614
615 if (!buf) {
f6e138b4
MS
616 /* If parsing of the previous packet actually returned 0 or an error,
617 * there's nothing more to be parsed from that packet, but we may have
58ee0991 618 * indicated that we can return the next enqueued packet. */
f6e138b4 619 if (s->prev_ret <= 0)
58ee0991 620 return rtp_parse_queued_packet(s, pkt);
02607418
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621 /* return the next packets, if any */
622 if(s->st && s->parse_packet) {
623 /* timestamp should be overwritten by parse_packet, if not,
624 * the packet is left with pts == AV_NOPTS_VALUE */
625 timestamp = RTP_NOTS_VALUE;
626 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
627 s->st, pkt, &timestamp, NULL, 0, flags);
628 finalize_packet(s, pkt, timestamp);
4ffff367 629 return rv;
02607418
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630 } else {
631 // TODO: Move to a dynamic packet handler (like above)
4ffff367 632 if (s->read_buf_index >= s->read_buf_size)
91ec7aea 633 return AVERROR(EAGAIN);
02607418
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634 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
635 s->read_buf_size - s->read_buf_index);
4ffff367 636 if (ret < 0)
91ec7aea 637 return ret;
02607418
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638 s->read_buf_index += ret;
639 if (s->read_buf_index < s->read_buf_size)
640 return 1;
4ffff367
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641 else
642 return 0;
02607418
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643 }
644 }
645
646 if (len < 12)
647 return -1;
648
649 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
650 return -1;
651 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
652 return rtcp_parse_packet(s, buf, len);
653 }
654
65cdee9c 655 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
58ee0991
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656 /* First packet, or no reordering */
657 return rtp_parse_packet_internal(s, pkt, buf, len);
658 } else {
659 uint16_t seq = AV_RB16(buf + 2);
660 int16_t diff = seq - s->seq;
661 if (diff < 0) {
662 /* Packet older than the previously emitted one, drop */
663 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
664 "RTP: dropping old packet received too late\n");
665 return -1;
666 } else if (diff <= 1) {
667 /* Correct packet */
668 rv = rtp_parse_packet_internal(s, pkt, buf, len);
4ffff367 669 return rv;
58ee0991
MS
670 } else {
671 /* Still missing some packet, enqueue this one. */
672 enqueue_packet(s, buf, len);
673 *bufptr = NULL;
674 /* Return the first enqueued packet if the queue is full,
675 * even if we're missing something */
676 if (s->queue_len >= s->queue_size)
677 return rtp_parse_queued_packet(s, pkt);
678 return -1;
679 }
680 }
02607418
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681}
682
4ffff367
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683/**
684 * Parse an RTP or RTCP packet directly sent as a buffer.
685 * @param s RTP parse context.
686 * @param pkt returned packet
687 * @param bufptr pointer to the input buffer or NULL to read the next packets
688 * @param len buffer len
689 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
690 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
691 */
692int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
693 uint8_t **bufptr, int len)
694{
695 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
696 s->prev_ret = rv;
d678a6fd
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697 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
698 rv = rtp_parse_queued_packet(s, pkt);
4ffff367
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699 return rv ? rv : has_next_packet(s);
700}
701
8eb793c4
LA
702void rtp_parse_close(RTPDemuxContext *s)
703{
58ee0991 704 ff_rtp_reset_packet_queue(s);
8eb793c4 705 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 706 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
707 }
708 av_free(s);
709}
016bc031
JA
710
711int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
712 int (*parse_fmtp)(AVStream *stream,
713 PayloadContext *data,
714 char *attr, char *value))
715{
716 char attr[256];
824535e3 717 char *value;
016bc031 718 int res;
824535e3
JA
719 int value_size = strlen(p) + 1;
720
721 if (!(value = av_malloc(value_size))) {
722 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
723 return AVERROR(ENOMEM);
724 }
016bc031
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725
726 // remove protocol identifier
727 while (*p && *p == ' ') p++; // strip spaces
728 while (*p && *p != ' ') p++; // eat protocol identifier
729 while (*p && *p == ' ') p++; // strip trailing spaces
730
731 while (ff_rtsp_next_attr_and_value(&p,
732 attr, sizeof(attr),
824535e3 733 value, value_size)) {
016bc031
JA
734
735 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
736 if (res < 0 && res != AVERROR_PATCHWELCOME) {
737 av_free(value);
016bc031 738 return res;
824535e3 739 }
016bc031 740 }
824535e3 741 av_free(value);
016bc031
JA
742 return 0;
743}