Recognize FFMPEG_FORCE_NOCOLOR environment variable on Win32
[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>
1e515c42 30#include <strings.h>
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31#include "network.h"
32
302879cb 33#include "rtpdec.h"
965a3ddb 34#include "rtpdec_formats.h"
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35
36//#define DEBUG
37
38/* TODO: - add RTCP statistics reporting (should be optional).
39
40 - add support for h263/mpeg4 packetized output : IDEA: send a
41 buffer to 'rtp_write_packet' contains all the packets for ONE
42 frame. Each packet should have a four byte header containing
43 the length in big endian format (same trick as
44 'url_open_dyn_packet_buf')
45*/
46
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47RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
48 .enc_name = "X-MP3-draft-00",
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = CODEC_ID_MP3ADU,
51};
52
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53/* statistics functions */
54RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
55
0369d2b0 56void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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57{
58 handler->next= RTPFirstDynamicPayloadHandler;
59 RTPFirstDynamicPayloadHandler= handler;
60}
61
62void av_register_rtp_dynamic_payload_handlers(void)
63{
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64 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
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66 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
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68 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 70 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 71 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
887af2aa 72 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
a59096e4 73 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
4449df6b 74 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
1ddc176e 75 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
51291e60 76 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
35014efc 77 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
2eeefe20 78 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
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79
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
81 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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82
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
84 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
86 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
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87}
88
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89RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
90 enum AVMediaType codec_type)
91{
92 RTPDynamicProtocolHandler *handler;
93 for (handler = RTPFirstDynamicPayloadHandler;
94 handler; handler = handler->next)
95 if (!strcasecmp(name, handler->enc_name) &&
96 codec_type == handler->codec_type)
97 return handler;
98 return NULL;
99}
100
101RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
102 enum AVMediaType codec_type)
103{
104 RTPDynamicProtocolHandler *handler;
105 for (handler = RTPFirstDynamicPayloadHandler;
106 handler; handler = handler->next)
107 if (handler->static_payload_id && handler->static_payload_id == id &&
108 codec_type == handler->codec_type)
109 return handler;
110 return NULL;
111}
112
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113static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
114{
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115 int payload_len;
116 while (len >= 2) {
117 switch (buf[1]) {
118 case RTCP_SR:
119 if (len < 16) {
120 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
121 return AVERROR_INVALIDDATA;
122 }
123 payload_len = (AV_RB16(buf + 2) + 1) * 4;
124
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125 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
126 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
127 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
128 s->last_rtcp_timestamp = AV_RB32(buf + 16);
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129
130 buf += payload_len;
131 len -= payload_len;
132 break;
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133 case RTCP_BYE:
134 return -RTCP_BYE;
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135 default:
136 return -1;
137 }
138 }
b20359f5 139 return -1;
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140}
141
142#define RTP_SEQ_MOD (1<<16)
143
144/**
145* called on parse open packet
146*/
147static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
148{
149 memset(s, 0, sizeof(RTPStatistics));
150 s->max_seq= base_sequence;
151 s->probation= 1;
152}
153
154/**
155* called whenever there is a large jump in sequence numbers, or when they get out of probation...
156*/
157static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
158{
159 s->max_seq= seq;
160 s->cycles= 0;
161 s->base_seq= seq -1;
162 s->bad_seq= RTP_SEQ_MOD + 1;
163 s->received= 0;
164 s->expected_prior= 0;
165 s->received_prior= 0;
166 s->jitter= 0;
167 s->transit= 0;
168}
169
170/**
171* returns 1 if we should handle this packet.
172*/
173static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
174{
175 uint16_t udelta= seq - s->max_seq;
176 const int MAX_DROPOUT= 3000;
177 const int MAX_MISORDER = 100;
178 const int MIN_SEQUENTIAL = 2;
179
180 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
181 if(s->probation)
182 {
183 if(seq==s->max_seq + 1) {
184 s->probation--;
185 s->max_seq= seq;
186 if(s->probation==0) {
187 rtp_init_sequence(s, seq);
188 s->received++;
189 return 1;
190 }
191 } else {
192 s->probation= MIN_SEQUENTIAL - 1;
193 s->max_seq = seq;
194 }
195 } else if (udelta < MAX_DROPOUT) {
196 // in order, with permissible gap
197 if(seq < s->max_seq) {
198 //sequence number wrapped; count antother 64k cycles
199 s->cycles += RTP_SEQ_MOD;
200 }
201 s->max_seq= seq;
202 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
203 // sequence made a large jump...
204 if(seq==s->bad_seq) {
205 // two sequential packets-- assume that the other side restarted without telling us; just resync.
206 rtp_init_sequence(s, seq);
207 } else {
208 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
209 return 0;
210 }
211 } else {
212 // duplicate or reordered packet...
213 }
214 s->received++;
215 return 1;
216}
217
218#if 0
219/**
220* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
221* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
222* never change. I left this in in case someone else can see a way. (rdm)
223*/
224static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
225{
226 uint32_t transit= arrival_timestamp - sent_timestamp;
227 int d;
228 s->transit= transit;
229 d= FFABS(transit - s->transit);
230 s->jitter += d - ((s->jitter + 8)>>4);
231}
232#endif
233
234int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
235{
236 ByteIOContext *pb;
237 uint8_t *buf;
238 int len;
239 int rtcp_bytes;
240 RTPStatistics *stats= &s->statistics;
241 uint32_t lost;
242 uint32_t extended_max;
243 uint32_t expected_interval;
244 uint32_t received_interval;
245 uint32_t lost_interval;
246 uint32_t expected;
247 uint32_t fraction;
248 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
249
250 if (!s->rtp_ctx || (count < 1))
251 return -1;
252
253 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
254 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
255 s->octet_count += count;
256 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
257 RTCP_TX_RATIO_DEN;
258 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
259 if (rtcp_bytes < 28)
260 return -1;
261 s->last_octet_count = s->octet_count;
262
263 if (url_open_dyn_buf(&pb) < 0)
264 return -1;
265
266 // Receiver Report
267 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 268 put_byte(pb, RTCP_RR);
8eb793c4 269 put_be16(pb, 7); /* length in words - 1 */
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270 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
271 put_be32(pb, s->ssrc + 1);
272 put_be32(pb, s->ssrc); // server SSRC
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273 // some placeholders we should really fill...
274 // RFC 1889/p64
275 extended_max= stats->cycles + stats->max_seq;
276 expected= extended_max - stats->base_seq + 1;
277 lost= expected - stats->received;
278 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
279 expected_interval= expected - stats->expected_prior;
280 stats->expected_prior= expected;
281 received_interval= stats->received - stats->received_prior;
282 stats->received_prior= stats->received;
283 lost_interval= expected_interval - received_interval;
284 if (expected_interval==0 || lost_interval<=0) fraction= 0;
285 else fraction = (lost_interval<<8)/expected_interval;
286
287 fraction= (fraction<<24) | lost;
288
289 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
290 put_be32(pb, extended_max); /* max sequence received */
291 put_be32(pb, stats->jitter>>4); /* jitter */
292
293 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
294 {
295 put_be32(pb, 0); /* last SR timestamp */
296 put_be32(pb, 0); /* delay since last SR */
297 } else {
298 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
299 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
300
301 put_be32(pb, middle_32_bits); /* last SR timestamp */
302 put_be32(pb, delay_since_last); /* delay since last SR */
303 }
304
305 // CNAME
306 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 307 put_byte(pb, RTCP_SDES);
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308 len = strlen(s->hostname);
309 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
310 put_be32(pb, s->ssrc);
311 put_byte(pb, 0x01);
312 put_byte(pb, len);
313 put_buffer(pb, s->hostname, len);
314 // padding
315 for (len = (6 + len) % 4; len % 4; len++) {
316 put_byte(pb, 0);
317 }
318
319 put_flush_packet(pb);
320 len = url_close_dyn_buf(pb, &buf);
321 if ((len > 0) && buf) {
322 int result;
e8420626 323 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 324 result= url_write(s->rtp_ctx, buf, len);
e8420626 325 dprintf(s->ic, "result from url_write: %d\n", result);
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326 av_free(buf);
327 }
328 return 0;
329}
330
9c8fa20d
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331void rtp_send_punch_packets(URLContext* rtp_handle)
332{
333 ByteIOContext *pb;
334 uint8_t *buf;
335 int len;
336
337 /* Send a small RTP packet */
338 if (url_open_dyn_buf(&pb) < 0)
339 return;
340
341 put_byte(pb, (RTP_VERSION << 6));
342 put_byte(pb, 0); /* Payload type */
343 put_be16(pb, 0); /* Seq */
344 put_be32(pb, 0); /* Timestamp */
345 put_be32(pb, 0); /* SSRC */
346
347 put_flush_packet(pb);
348 len = url_close_dyn_buf(pb, &buf);
349 if ((len > 0) && buf)
350 url_write(rtp_handle, buf, len);
351 av_free(buf);
352
353 /* Send a minimal RTCP RR */
354 if (url_open_dyn_buf(&pb) < 0)
355 return;
356
357 put_byte(pb, (RTP_VERSION << 6));
7f3468d3 358 put_byte(pb, RTCP_RR); /* receiver report */
9c8fa20d
MS
359 put_be16(pb, 1); /* length in words - 1 */
360 put_be32(pb, 0); /* our own SSRC */
361
362 put_flush_packet(pb);
363 len = url_close_dyn_buf(pb, &buf);
364 if ((len > 0) && buf)
365 url_write(rtp_handle, buf, len);
366 av_free(buf);
367}
368
369
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370/**
371 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
372 * MPEG2TS streams to indicate that they should be demuxed inside the
373 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 374 */
58ee0991 375RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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376{
377 RTPDemuxContext *s;
378
379 s = av_mallocz(sizeof(RTPDemuxContext));
380 if (!s)
381 return NULL;
382 s->payload_type = payload_type;
383 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 384 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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385 s->ic = s1;
386 s->st = st;
58ee0991 387 s->queue_size = queue_size;
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388 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
389 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 390 s->ts = ff_mpegts_parse_open(s->ic);
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391 if (s->ts == NULL) {
392 av_free(s);
393 return NULL;
394 }
395 } else {
396 switch(st->codec->codec_id) {
397 case CODEC_ID_MPEG1VIDEO:
398 case CODEC_ID_MPEG2VIDEO:
399 case CODEC_ID_MP2:
400 case CODEC_ID_MP3:
401 case CODEC_ID_MPEG4:
45aa9080 402 case CODEC_ID_H263:
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403 case CODEC_ID_H264:
404 st->need_parsing = AVSTREAM_PARSE_FULL;
405 break;
0048a2a8 406 case CODEC_ID_ADPCM_G722:
0048a2a8
MS
407 /* According to RFC 3551, the stream clock rate is 8000
408 * even if the sample rate is 16000. */
409 if (st->codec->sample_rate == 8000)
410 st->codec->sample_rate = 16000;
411 break;
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412 default:
413 break;
414 }
415 }
416 // needed to send back RTCP RR in RTSP sessions
417 s->rtp_ctx = rtpc;
418 gethostname(s->hostname, sizeof(s->hostname));
419 return s;
420}
421
99a1d191
RB
422void
423rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
424 RTPDynamicProtocolHandler *handler)
425{
426 s->dynamic_protocol_context = ctx;
427 s->parse_packet = handler->parse_packet;
428}
429
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430/**
431 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
432 */
433static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
434{
d74c6145 435 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
fba7815d
LA
436 int64_t addend;
437 int delta_timestamp;
438
439 /* compute pts from timestamp with received ntp_time */
440 delta_timestamp = timestamp - s->last_rtcp_timestamp;
441 /* convert to the PTS timebase */
2cab6b48 442 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 443 pkt->pts = s->range_start_offset + addend + delta_timestamp;
fba7815d 444 }
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445}
446
02607418
MS
447static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
448 const uint8_t *buf, int len)
8eb793c4
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449{
450 unsigned int ssrc, h;
f841a0fc 451 int payload_type, seq, ret, flags = 0;
9446b4bb 452 int ext;
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453 AVStream *st;
454 uint32_t timestamp;
455 int rv= 0;
456
9446b4bb 457 ext = buf[0] & 0x10;
8eb793c4 458 payload_type = buf[1] & 0x7f;
144ae29d
RB
459 if (buf[1] & 0x80)
460 flags |= RTP_FLAG_MARKER;
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461 seq = AV_RB16(buf + 2);
462 timestamp = AV_RB32(buf + 4);
463 ssrc = AV_RB32(buf + 8);
464 /* store the ssrc in the RTPDemuxContext */
465 s->ssrc = ssrc;
466
467 /* NOTE: we can handle only one payload type */
468 if (s->payload_type != payload_type)
469 return -1;
470
471 st = s->st;
472 // only do something with this if all the rtp checks pass...
473 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
474 {
475 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
476 payload_type, seq, ((s->seq + 1) & 0xffff));
477 return -1;
478 }
479
4838cdab
MS
480 if (buf[0] & 0x20) {
481 int padding = buf[len - 1];
482 if (len >= 12 + padding)
483 len -= padding;
484 }
485
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486 s->seq = seq;
487 len -= 12;
488 buf += 12;
489
9446b4bb
RS
490 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
491 if (ext) {
492 if (len < 4)
493 return -1;
494 /* calculate the header extension length (stored as number
495 * of 32-bit words) */
496 ext = (AV_RB16(buf + 2) + 1) << 2;
497
498 if (len < ext)
499 return -1;
500 // skip past RTP header extension
501 len -= ext;
502 buf += ext;
503 }
504
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LA
505 if (!st) {
506 /* specific MPEG2TS demux support */
9125806e 507 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
946df059
MS
508 /* The only error that can be returned from ff_mpegts_parse_packet
509 * is "no more data to return from the provided buffer", so return
510 * AVERROR(EAGAIN) for all errors */
4ffff367 511 if (ret < 0)
946df059 512 return AVERROR(EAGAIN);
8eb793c4
LA
513 if (ret < len) {
514 s->read_buf_size = len - ret;
515 memcpy(s->buf, buf + ret, s->read_buf_size);
516 s->read_buf_index = 0;
517 return 1;
518 }
f3e71942 519 return 0;
b4e3330c 520 } else if (s->parse_packet) {
1a45a9f4 521 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 522 s->st, pkt, &timestamp, buf, len, flags);
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LA
523 } else {
524 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
525 switch(st->codec->codec_id) {
526 case CODEC_ID_MP2:
76faff6e 527 case CODEC_ID_MP3:
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LA
528 /* better than nothing: skip mpeg audio RTP header */
529 if (len <= 4)
530 return -1;
531 h = AV_RB32(buf);
532 len -= 4;
533 buf += 4;
534 av_new_packet(pkt, len);
535 memcpy(pkt->data, buf, len);
536 break;
537 case CODEC_ID_MPEG1VIDEO:
538 case CODEC_ID_MPEG2VIDEO:
539 /* better than nothing: skip mpeg video RTP header */
540 if (len <= 4)
541 return -1;
542 h = AV_RB32(buf);
543 buf += 4;
544 len -= 4;
545 if (h & (1 << 26)) {
546 /* mpeg2 */
547 if (len <= 4)
548 return -1;
549 buf += 4;
550 len -= 4;
551 }
552 av_new_packet(pkt, len);
553 memcpy(pkt->data, buf, len);
554 break;
8eb793c4 555 default:
f739b36d
RB
556 av_new_packet(pkt, len);
557 memcpy(pkt->data, buf, len);
8eb793c4
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558 break;
559 }
eafb17d1
RB
560
561 pkt->stream_index = st->index;
f3e71942 562 }
8eb793c4 563
95f03cf3
RB
564 // now perform timestamp things....
565 finalize_packet(s, pkt, timestamp);
f3e71942 566
8eb793c4
LA
567 return rv;
568}
569
58ee0991
MS
570void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
571{
572 while (s->queue) {
573 RTPPacket *next = s->queue->next;
574 av_free(s->queue->buf);
575 av_free(s->queue);
576 s->queue = next;
577 }
578 s->seq = 0;
579 s->queue_len = 0;
580 s->prev_ret = 0;
581}
582
583static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
584{
585 uint16_t seq = AV_RB16(buf + 2);
586 RTPPacket *cur = s->queue, *prev = NULL, *packet;
587
588 /* Find the correct place in the queue to insert the packet */
589 while (cur) {
590 int16_t diff = seq - cur->seq;
591 if (diff < 0)
592 break;
593 prev = cur;
594 cur = cur->next;
595 }
596
597 packet = av_mallocz(sizeof(*packet));
598 if (!packet)
599 return;
600 packet->recvtime = av_gettime();
601 packet->seq = seq;
602 packet->len = len;
603 packet->buf = buf;
604 packet->next = cur;
605 if (prev)
606 prev->next = packet;
607 else
608 s->queue = packet;
609 s->queue_len++;
610}
611
612static int has_next_packet(RTPDemuxContext *s)
613{
ddcf8411 614 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
58ee0991
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615}
616
617int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
618{
619 return s->queue ? s->queue->recvtime : 0;
620}
621
622static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
623{
624 int rv;
625 RTPPacket *next;
626
627 if (s->queue_len <= 0)
628 return -1;
629
630 if (!has_next_packet(s))
631 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
632 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
633
634 /* Parse the first packet in the queue, and dequeue it */
635 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
636 next = s->queue->next;
637 av_free(s->queue->buf);
638 av_free(s->queue);
639 s->queue = next;
640 s->queue_len--;
4ffff367 641 return rv;
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642}
643
4ffff367 644static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
02607418
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645 uint8_t **bufptr, int len)
646{
647 uint8_t* buf = bufptr ? *bufptr : NULL;
648 int ret, flags = 0;
649 uint32_t timestamp;
650 int rv= 0;
651
652 if (!buf) {
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653 /* If parsing of the previous packet actually returned 0 or an error,
654 * there's nothing more to be parsed from that packet, but we may have
58ee0991 655 * indicated that we can return the next enqueued packet. */
f6e138b4 656 if (s->prev_ret <= 0)
58ee0991 657 return rtp_parse_queued_packet(s, pkt);
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658 /* return the next packets, if any */
659 if(s->st && s->parse_packet) {
660 /* timestamp should be overwritten by parse_packet, if not,
661 * the packet is left with pts == AV_NOPTS_VALUE */
662 timestamp = RTP_NOTS_VALUE;
663 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
664 s->st, pkt, &timestamp, NULL, 0, flags);
665 finalize_packet(s, pkt, timestamp);
4ffff367 666 return rv;
02607418
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667 } else {
668 // TODO: Move to a dynamic packet handler (like above)
4ffff367 669 if (s->read_buf_index >= s->read_buf_size)
91ec7aea 670 return AVERROR(EAGAIN);
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671 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
672 s->read_buf_size - s->read_buf_index);
4ffff367 673 if (ret < 0)
946df059 674 return AVERROR(EAGAIN);
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675 s->read_buf_index += ret;
676 if (s->read_buf_index < s->read_buf_size)
677 return 1;
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678 else
679 return 0;
02607418
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680 }
681 }
682
683 if (len < 12)
684 return -1;
685
686 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
687 return -1;
688 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
689 return rtcp_parse_packet(s, buf, len);
690 }
691
65cdee9c 692 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
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693 /* First packet, or no reordering */
694 return rtp_parse_packet_internal(s, pkt, buf, len);
695 } else {
696 uint16_t seq = AV_RB16(buf + 2);
697 int16_t diff = seq - s->seq;
698 if (diff < 0) {
699 /* Packet older than the previously emitted one, drop */
700 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
701 "RTP: dropping old packet received too late\n");
702 return -1;
703 } else if (diff <= 1) {
704 /* Correct packet */
705 rv = rtp_parse_packet_internal(s, pkt, buf, len);
4ffff367 706 return rv;
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707 } else {
708 /* Still missing some packet, enqueue this one. */
709 enqueue_packet(s, buf, len);
710 *bufptr = NULL;
711 /* Return the first enqueued packet if the queue is full,
712 * even if we're missing something */
713 if (s->queue_len >= s->queue_size)
714 return rtp_parse_queued_packet(s, pkt);
715 return -1;
716 }
717 }
02607418
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718}
719
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720/**
721 * Parse an RTP or RTCP packet directly sent as a buffer.
722 * @param s RTP parse context.
723 * @param pkt returned packet
724 * @param bufptr pointer to the input buffer or NULL to read the next packets
725 * @param len buffer len
726 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
727 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
728 */
729int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
730 uint8_t **bufptr, int len)
731{
732 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
733 s->prev_ret = rv;
d678a6fd
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734 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
735 rv = rtp_parse_queued_packet(s, pkt);
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736 return rv ? rv : has_next_packet(s);
737}
738
8eb793c4
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739void rtp_parse_close(RTPDemuxContext *s)
740{
58ee0991 741 ff_rtp_reset_packet_queue(s);
8eb793c4 742 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 743 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
744 }
745 av_free(s);
746}
016bc031
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747
748int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
749 int (*parse_fmtp)(AVStream *stream,
750 PayloadContext *data,
751 char *attr, char *value))
752{
753 char attr[256];
824535e3 754 char *value;
016bc031 755 int res;
824535e3
JA
756 int value_size = strlen(p) + 1;
757
758 if (!(value = av_malloc(value_size))) {
759 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
760 return AVERROR(ENOMEM);
761 }
016bc031
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762
763 // remove protocol identifier
764 while (*p && *p == ' ') p++; // strip spaces
765 while (*p && *p != ' ') p++; // eat protocol identifier
766 while (*p && *p == ' ') p++; // strip trailing spaces
767
768 while (ff_rtsp_next_attr_and_value(&p,
769 attr, sizeof(attr),
824535e3 770 value, value_size)) {
016bc031
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771
772 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
773 if (res < 0 && res != AVERROR_PATCHWELCOME) {
774 av_free(value);
016bc031 775 return res;
824535e3 776 }
016bc031 777 }
824535e3 778 av_free(value);
016bc031
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779 return 0;
780}