configure: Set _DARWIN_C_SOURCE while testing for struct ipv6_mreq
[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"
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{
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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
75static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
76{
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77 int payload_len;
78 while (len >= 2) {
79 switch (buf[1]) {
80 case RTCP_SR:
81 if (len < 16) {
82 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
83 return AVERROR_INVALIDDATA;
84 }
85 payload_len = (AV_RB16(buf + 2) + 1) * 4;
86
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87 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
88 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
89 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
90 s->last_rtcp_timestamp = AV_RB32(buf + 16);
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91
92 buf += payload_len;
93 len -= payload_len;
94 break;
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95 case RTCP_BYE:
96 return -RTCP_BYE;
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97 default:
98 return -1;
99 }
100 }
b20359f5 101 return -1;
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102}
103
104#define RTP_SEQ_MOD (1<<16)
105
106/**
107* called on parse open packet
108*/
109static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
110{
111 memset(s, 0, sizeof(RTPStatistics));
112 s->max_seq= base_sequence;
113 s->probation= 1;
114}
115
116/**
117* called whenever there is a large jump in sequence numbers, or when they get out of probation...
118*/
119static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
120{
121 s->max_seq= seq;
122 s->cycles= 0;
123 s->base_seq= seq -1;
124 s->bad_seq= RTP_SEQ_MOD + 1;
125 s->received= 0;
126 s->expected_prior= 0;
127 s->received_prior= 0;
128 s->jitter= 0;
129 s->transit= 0;
130}
131
132/**
133* returns 1 if we should handle this packet.
134*/
135static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
136{
137 uint16_t udelta= seq - s->max_seq;
138 const int MAX_DROPOUT= 3000;
139 const int MAX_MISORDER = 100;
140 const int MIN_SEQUENTIAL = 2;
141
142 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
143 if(s->probation)
144 {
145 if(seq==s->max_seq + 1) {
146 s->probation--;
147 s->max_seq= seq;
148 if(s->probation==0) {
149 rtp_init_sequence(s, seq);
150 s->received++;
151 return 1;
152 }
153 } else {
154 s->probation= MIN_SEQUENTIAL - 1;
155 s->max_seq = seq;
156 }
157 } else if (udelta < MAX_DROPOUT) {
158 // in order, with permissible gap
159 if(seq < s->max_seq) {
160 //sequence number wrapped; count antother 64k cycles
161 s->cycles += RTP_SEQ_MOD;
162 }
163 s->max_seq= seq;
164 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
165 // sequence made a large jump...
166 if(seq==s->bad_seq) {
167 // two sequential packets-- assume that the other side restarted without telling us; just resync.
168 rtp_init_sequence(s, seq);
169 } else {
170 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
171 return 0;
172 }
173 } else {
174 // duplicate or reordered packet...
175 }
176 s->received++;
177 return 1;
178}
179
180#if 0
181/**
182* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
183* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
184* never change. I left this in in case someone else can see a way. (rdm)
185*/
186static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
187{
188 uint32_t transit= arrival_timestamp - sent_timestamp;
189 int d;
190 s->transit= transit;
191 d= FFABS(transit - s->transit);
192 s->jitter += d - ((s->jitter + 8)>>4);
193}
194#endif
195
196int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
197{
198 ByteIOContext *pb;
199 uint8_t *buf;
200 int len;
201 int rtcp_bytes;
202 RTPStatistics *stats= &s->statistics;
203 uint32_t lost;
204 uint32_t extended_max;
205 uint32_t expected_interval;
206 uint32_t received_interval;
207 uint32_t lost_interval;
208 uint32_t expected;
209 uint32_t fraction;
210 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
211
212 if (!s->rtp_ctx || (count < 1))
213 return -1;
214
215 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
216 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
217 s->octet_count += count;
218 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
219 RTCP_TX_RATIO_DEN;
220 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
221 if (rtcp_bytes < 28)
222 return -1;
223 s->last_octet_count = s->octet_count;
224
225 if (url_open_dyn_buf(&pb) < 0)
226 return -1;
227
228 // Receiver Report
229 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 230 put_byte(pb, RTCP_RR);
8eb793c4 231 put_be16(pb, 7); /* length in words - 1 */
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232 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
233 put_be32(pb, s->ssrc + 1);
234 put_be32(pb, s->ssrc); // server SSRC
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235 // some placeholders we should really fill...
236 // RFC 1889/p64
237 extended_max= stats->cycles + stats->max_seq;
238 expected= extended_max - stats->base_seq + 1;
239 lost= expected - stats->received;
240 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
241 expected_interval= expected - stats->expected_prior;
242 stats->expected_prior= expected;
243 received_interval= stats->received - stats->received_prior;
244 stats->received_prior= stats->received;
245 lost_interval= expected_interval - received_interval;
246 if (expected_interval==0 || lost_interval<=0) fraction= 0;
247 else fraction = (lost_interval<<8)/expected_interval;
248
249 fraction= (fraction<<24) | lost;
250
251 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
252 put_be32(pb, extended_max); /* max sequence received */
253 put_be32(pb, stats->jitter>>4); /* jitter */
254
255 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
256 {
257 put_be32(pb, 0); /* last SR timestamp */
258 put_be32(pb, 0); /* delay since last SR */
259 } else {
260 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
261 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
262
263 put_be32(pb, middle_32_bits); /* last SR timestamp */
264 put_be32(pb, delay_since_last); /* delay since last SR */
265 }
266
267 // CNAME
268 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 269 put_byte(pb, RTCP_SDES);
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270 len = strlen(s->hostname);
271 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
272 put_be32(pb, s->ssrc);
273 put_byte(pb, 0x01);
274 put_byte(pb, len);
275 put_buffer(pb, s->hostname, len);
276 // padding
277 for (len = (6 + len) % 4; len % 4; len++) {
278 put_byte(pb, 0);
279 }
280
281 put_flush_packet(pb);
282 len = url_close_dyn_buf(pb, &buf);
283 if ((len > 0) && buf) {
284 int result;
e8420626 285 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 286 result= url_write(s->rtp_ctx, buf, len);
e8420626 287 dprintf(s->ic, "result from url_write: %d\n", result);
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288 av_free(buf);
289 }
290 return 0;
291}
292
9c8fa20d
MS
293void rtp_send_punch_packets(URLContext* rtp_handle)
294{
295 ByteIOContext *pb;
296 uint8_t *buf;
297 int len;
298
299 /* Send a small RTP packet */
300 if (url_open_dyn_buf(&pb) < 0)
301 return;
302
303 put_byte(pb, (RTP_VERSION << 6));
304 put_byte(pb, 0); /* Payload type */
305 put_be16(pb, 0); /* Seq */
306 put_be32(pb, 0); /* Timestamp */
307 put_be32(pb, 0); /* SSRC */
308
309 put_flush_packet(pb);
310 len = url_close_dyn_buf(pb, &buf);
311 if ((len > 0) && buf)
312 url_write(rtp_handle, buf, len);
313 av_free(buf);
314
315 /* Send a minimal RTCP RR */
316 if (url_open_dyn_buf(&pb) < 0)
317 return;
318
319 put_byte(pb, (RTP_VERSION << 6));
7f3468d3 320 put_byte(pb, RTCP_RR); /* receiver report */
9c8fa20d
MS
321 put_be16(pb, 1); /* length in words - 1 */
322 put_be32(pb, 0); /* our own SSRC */
323
324 put_flush_packet(pb);
325 len = url_close_dyn_buf(pb, &buf);
326 if ((len > 0) && buf)
327 url_write(rtp_handle, buf, len);
328 av_free(buf);
329}
330
331
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332/**
333 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
334 * MPEG2TS streams to indicate that they should be demuxed inside the
335 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 336 */
58ee0991 337RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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338{
339 RTPDemuxContext *s;
340
341 s = av_mallocz(sizeof(RTPDemuxContext));
342 if (!s)
343 return NULL;
344 s->payload_type = payload_type;
345 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 346 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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347 s->ic = s1;
348 s->st = st;
58ee0991 349 s->queue_size = queue_size;
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350 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
351 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 352 s->ts = ff_mpegts_parse_open(s->ic);
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353 if (s->ts == NULL) {
354 av_free(s);
355 return NULL;
356 }
357 } else {
26efefc5 358 av_set_pts_info(st, 32, 1, 90000);
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359 switch(st->codec->codec_id) {
360 case CODEC_ID_MPEG1VIDEO:
361 case CODEC_ID_MPEG2VIDEO:
362 case CODEC_ID_MP2:
363 case CODEC_ID_MP3:
364 case CODEC_ID_MPEG4:
45aa9080 365 case CODEC_ID_H263:
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366 case CODEC_ID_H264:
367 st->need_parsing = AVSTREAM_PARSE_FULL;
368 break;
0048a2a8
MS
369 case CODEC_ID_ADPCM_G722:
370 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
371 /* According to RFC 3551, the stream clock rate is 8000
372 * even if the sample rate is 16000. */
373 if (st->codec->sample_rate == 8000)
374 st->codec->sample_rate = 16000;
375 break;
8eb793c4 376 default:
72415b2a 377 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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LA
378 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
379 }
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380 break;
381 }
382 }
383 // needed to send back RTCP RR in RTSP sessions
384 s->rtp_ctx = rtpc;
385 gethostname(s->hostname, sizeof(s->hostname));
386 return s;
387}
388
99a1d191
RB
389void
390rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
391 RTPDynamicProtocolHandler *handler)
392{
393 s->dynamic_protocol_context = ctx;
394 s->parse_packet = handler->parse_packet;
395}
396
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397/**
398 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
399 */
400static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
401{
d74c6145 402 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
fba7815d
LA
403 int64_t addend;
404 int delta_timestamp;
405
406 /* compute pts from timestamp with received ntp_time */
407 delta_timestamp = timestamp - s->last_rtcp_timestamp;
408 /* convert to the PTS timebase */
2cab6b48 409 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 410 pkt->pts = s->range_start_offset + addend + delta_timestamp;
fba7815d 411 }
8eb793c4
LA
412}
413
02607418
MS
414static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
415 const uint8_t *buf, int len)
8eb793c4
LA
416{
417 unsigned int ssrc, h;
f841a0fc 418 int payload_type, seq, ret, flags = 0;
8eb793c4
LA
419 AVStream *st;
420 uint32_t timestamp;
421 int rv= 0;
422
8eb793c4 423 payload_type = buf[1] & 0x7f;
144ae29d
RB
424 if (buf[1] & 0x80)
425 flags |= RTP_FLAG_MARKER;
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LA
426 seq = AV_RB16(buf + 2);
427 timestamp = AV_RB32(buf + 4);
428 ssrc = AV_RB32(buf + 8);
429 /* store the ssrc in the RTPDemuxContext */
430 s->ssrc = ssrc;
431
432 /* NOTE: we can handle only one payload type */
433 if (s->payload_type != payload_type)
434 return -1;
435
436 st = s->st;
437 // only do something with this if all the rtp checks pass...
438 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
439 {
440 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
441 payload_type, seq, ((s->seq + 1) & 0xffff));
442 return -1;
443 }
444
445 s->seq = seq;
446 len -= 12;
447 buf += 12;
448
449 if (!st) {
450 /* specific MPEG2TS demux support */
9125806e 451 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
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452 if (ret < 0)
453 return -1;
454 if (ret < len) {
455 s->read_buf_size = len - ret;
456 memcpy(s->buf, buf + ret, s->read_buf_size);
457 s->read_buf_index = 0;
458 return 1;
459 }
f3e71942 460 return 0;
b4e3330c 461 } else if (s->parse_packet) {
1a45a9f4 462 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 463 s->st, pkt, &timestamp, buf, len, flags);
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464 } else {
465 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
466 switch(st->codec->codec_id) {
467 case CODEC_ID_MP2:
76faff6e 468 case CODEC_ID_MP3:
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LA
469 /* better than nothing: skip mpeg audio RTP header */
470 if (len <= 4)
471 return -1;
472 h = AV_RB32(buf);
473 len -= 4;
474 buf += 4;
475 av_new_packet(pkt, len);
476 memcpy(pkt->data, buf, len);
477 break;
478 case CODEC_ID_MPEG1VIDEO:
479 case CODEC_ID_MPEG2VIDEO:
480 /* better than nothing: skip mpeg video RTP header */
481 if (len <= 4)
482 return -1;
483 h = AV_RB32(buf);
484 buf += 4;
485 len -= 4;
486 if (h & (1 << 26)) {
487 /* mpeg2 */
488 if (len <= 4)
489 return -1;
490 buf += 4;
491 len -= 4;
492 }
493 av_new_packet(pkt, len);
494 memcpy(pkt->data, buf, len);
495 break;
8eb793c4 496 default:
f739b36d
RB
497 av_new_packet(pkt, len);
498 memcpy(pkt->data, buf, len);
8eb793c4
LA
499 break;
500 }
eafb17d1
RB
501
502 pkt->stream_index = st->index;
f3e71942 503 }
8eb793c4 504
95f03cf3
RB
505 // now perform timestamp things....
506 finalize_packet(s, pkt, timestamp);
f3e71942 507
58ee0991 508 s->prev_ret = rv;
8eb793c4
LA
509 return rv;
510}
511
58ee0991
MS
512void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
513{
514 while (s->queue) {
515 RTPPacket *next = s->queue->next;
516 av_free(s->queue->buf);
517 av_free(s->queue);
518 s->queue = next;
519 }
520 s->seq = 0;
521 s->queue_len = 0;
522 s->prev_ret = 0;
523}
524
525static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
526{
527 uint16_t seq = AV_RB16(buf + 2);
528 RTPPacket *cur = s->queue, *prev = NULL, *packet;
529
530 /* Find the correct place in the queue to insert the packet */
531 while (cur) {
532 int16_t diff = seq - cur->seq;
533 if (diff < 0)
534 break;
535 prev = cur;
536 cur = cur->next;
537 }
538
539 packet = av_mallocz(sizeof(*packet));
540 if (!packet)
541 return;
542 packet->recvtime = av_gettime();
543 packet->seq = seq;
544 packet->len = len;
545 packet->buf = buf;
546 packet->next = cur;
547 if (prev)
548 prev->next = packet;
549 else
550 s->queue = packet;
551 s->queue_len++;
552}
553
554static int has_next_packet(RTPDemuxContext *s)
555{
556 return s->queue && s->queue->seq == s->seq + 1;
557}
558
559int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
560{
561 return s->queue ? s->queue->recvtime : 0;
562}
563
564static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
565{
566 int rv;
567 RTPPacket *next;
568
569 if (s->queue_len <= 0)
570 return -1;
571
572 if (!has_next_packet(s))
573 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
574 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
575
576 /* Parse the first packet in the queue, and dequeue it */
577 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
578 next = s->queue->next;
579 av_free(s->queue->buf);
580 av_free(s->queue);
581 s->queue = next;
582 s->queue_len--;
583 return rv ? rv : has_next_packet(s);
584}
585
02607418
MS
586/**
587 * Parse an RTP or RTCP packet directly sent as a buffer.
588 * @param s RTP parse context.
589 * @param pkt returned packet
590 * @param bufptr pointer to the input buffer or NULL to read the next packets
591 * @param len buffer len
592 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
593 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
594 */
595int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
596 uint8_t **bufptr, int len)
597{
598 uint8_t* buf = bufptr ? *bufptr : NULL;
599 int ret, flags = 0;
600 uint32_t timestamp;
601 int rv= 0;
602
603 if (!buf) {
58ee0991
MS
604 /* If parsing of the previous packet actually returned 0, there's
605 * nothing more to be parsed from that packet, but we may have
606 * indicated that we can return the next enqueued packet. */
607 if (!s->prev_ret)
608 return rtp_parse_queued_packet(s, pkt);
02607418
MS
609 /* return the next packets, if any */
610 if(s->st && s->parse_packet) {
611 /* timestamp should be overwritten by parse_packet, if not,
612 * the packet is left with pts == AV_NOPTS_VALUE */
613 timestamp = RTP_NOTS_VALUE;
614 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
615 s->st, pkt, &timestamp, NULL, 0, flags);
616 finalize_packet(s, pkt, timestamp);
58ee0991
MS
617 s->prev_ret = rv;
618 return rv ? rv : has_next_packet(s);
02607418
MS
619 } else {
620 // TODO: Move to a dynamic packet handler (like above)
621 if (s->read_buf_index >= s->read_buf_size)
622 return -1;
623 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
624 s->read_buf_size - s->read_buf_index);
625 if (ret < 0)
626 return -1;
627 s->read_buf_index += ret;
628 if (s->read_buf_index < s->read_buf_size)
629 return 1;
58ee0991
MS
630 else {
631 s->prev_ret = 0;
632 return has_next_packet(s);
633 }
02607418
MS
634 }
635 }
636
637 if (len < 12)
638 return -1;
639
640 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
641 return -1;
642 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
643 return rtcp_parse_packet(s, buf, len);
644 }
645
58ee0991
MS
646 if (s->seq == 0 || s->queue_size <= 1) {
647 /* First packet, or no reordering */
648 return rtp_parse_packet_internal(s, pkt, buf, len);
649 } else {
650 uint16_t seq = AV_RB16(buf + 2);
651 int16_t diff = seq - s->seq;
652 if (diff < 0) {
653 /* Packet older than the previously emitted one, drop */
654 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
655 "RTP: dropping old packet received too late\n");
656 return -1;
657 } else if (diff <= 1) {
658 /* Correct packet */
659 rv = rtp_parse_packet_internal(s, pkt, buf, len);
660 return rv ? rv : has_next_packet(s);
661 } else {
662 /* Still missing some packet, enqueue this one. */
663 enqueue_packet(s, buf, len);
664 *bufptr = NULL;
665 /* Return the first enqueued packet if the queue is full,
666 * even if we're missing something */
667 if (s->queue_len >= s->queue_size)
668 return rtp_parse_queued_packet(s, pkt);
669 return -1;
670 }
671 }
02607418
MS
672}
673
8eb793c4
LA
674void rtp_parse_close(RTPDemuxContext *s)
675{
58ee0991 676 ff_rtp_reset_packet_queue(s);
8eb793c4 677 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 678 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
679 }
680 av_free(s);
681}
016bc031
JA
682
683int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
684 int (*parse_fmtp)(AVStream *stream,
685 PayloadContext *data,
686 char *attr, char *value))
687{
688 char attr[256];
824535e3 689 char *value;
016bc031 690 int res;
824535e3
JA
691 int value_size = strlen(p) + 1;
692
693 if (!(value = av_malloc(value_size))) {
694 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
695 return AVERROR(ENOMEM);
696 }
016bc031
JA
697
698 // remove protocol identifier
699 while (*p && *p == ' ') p++; // strip spaces
700 while (*p && *p != ' ') p++; // eat protocol identifier
701 while (*p && *p == ' ') p++; // strip trailing spaces
702
703 while (ff_rtsp_next_attr_and_value(&p,
704 attr, sizeof(attr),
824535e3 705 value, value_size)) {
016bc031
JA
706
707 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
708 if (res < 0 && res != AVERROR_PATCHWELCOME) {
709 av_free(value);
016bc031 710 return res;
824535e3 711 }
016bc031 712 }
824535e3 713 av_free(value);
016bc031
JA
714 return 0;
715}