1 /*
2 * RTP input format
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 #include "libavutil/mathematics.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/time.h"
26
27 #include "avformat.h"
28 #include "network.h"
29 #include "srtp.h"
30 #include "url.h"
31 #include "rtpdec.h"
32 #include "rtpdec_formats.h"
33
34 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
35
36 static RTPDynamicProtocolHandler l24_dynamic_handler = {
37 .enc_name = "L24",
38 .codec_type = AVMEDIA_TYPE_AUDIO,
39 .codec_id = AV_CODEC_ID_PCM_S24BE,
40 };
41
42 static RTPDynamicProtocolHandler gsm_dynamic_handler = {
43 .enc_name = "GSM",
44 .codec_type = AVMEDIA_TYPE_AUDIO,
45 .codec_id = AV_CODEC_ID_GSM,
46 };
47
48 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
49 .enc_name = "X-MP3-draft-00",
50 .codec_type = AVMEDIA_TYPE_AUDIO,
51 .codec_id = AV_CODEC_ID_MP3ADU,
52 };
53
54 static RTPDynamicProtocolHandler speex_dynamic_handler = {
55 .enc_name = "speex",
56 .codec_type = AVMEDIA_TYPE_AUDIO,
57 .codec_id = AV_CODEC_ID_SPEEX,
58 };
59
60 static RTPDynamicProtocolHandler opus_dynamic_handler = {
61 .enc_name = "opus",
62 .codec_type = AVMEDIA_TYPE_AUDIO,
63 .codec_id = AV_CODEC_ID_OPUS,
64 };
65
66 static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
67 .enc_name = "t140",
68 .codec_type = AVMEDIA_TYPE_SUBTITLE,
69 .codec_id = AV_CODEC_ID_TEXT,
70 };
71
72 extern RTPDynamicProtocolHandler ff_rdt_video_handler;
73 extern RTPDynamicProtocolHandler ff_rdt_audio_handler;
74 extern RTPDynamicProtocolHandler ff_rdt_live_video_handler;
75 extern RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
76
77 static const RTPDynamicProtocolHandler *rtp_dynamic_protocol_handler_list[] = {
78 /* rtp */
79 &ff_ac3_dynamic_handler,
80 &ff_amr_nb_dynamic_handler,
81 &ff_amr_wb_dynamic_handler,
82 &ff_dv_dynamic_handler,
83 &ff_g726_16_dynamic_handler,
84 &ff_g726_24_dynamic_handler,
85 &ff_g726_32_dynamic_handler,
86 &ff_g726_40_dynamic_handler,
87 &ff_g726le_16_dynamic_handler,
88 &ff_g726le_24_dynamic_handler,
89 &ff_g726le_32_dynamic_handler,
90 &ff_g726le_40_dynamic_handler,
91 &ff_h261_dynamic_handler,
92 &ff_h263_1998_dynamic_handler,
93 &ff_h263_2000_dynamic_handler,
94 &ff_h263_rfc2190_dynamic_handler,
95 &ff_h264_dynamic_handler,
96 &ff_hevc_dynamic_handler,
97 &ff_ilbc_dynamic_handler,
98 &ff_jpeg_dynamic_handler,
99 &ff_mp4a_latm_dynamic_handler,
100 &ff_mp4v_es_dynamic_handler,
101 &ff_mpeg_audio_dynamic_handler,
102 &ff_mpeg_audio_robust_dynamic_handler,
103 &ff_mpeg_video_dynamic_handler,
104 &ff_mpeg4_generic_dynamic_handler,
105 &ff_mpegts_dynamic_handler,
106 &ff_ms_rtp_asf_pfa_handler,
107 &ff_ms_rtp_asf_pfv_handler,
108 &ff_qcelp_dynamic_handler,
109 &ff_qdm2_dynamic_handler,
110 &ff_qt_rtp_aud_handler,
111 &ff_qt_rtp_vid_handler,
112 &ff_quicktime_rtp_aud_handler,
113 &ff_quicktime_rtp_vid_handler,
114 &ff_rfc4175_rtp_handler,
115 &ff_svq3_dynamic_handler,
116 &ff_theora_dynamic_handler,
117 &ff_vc2hq_dynamic_handler,
118 &ff_vorbis_dynamic_handler,
119 &ff_vp8_dynamic_handler,
120 &ff_vp9_dynamic_handler,
121 &gsm_dynamic_handler,
122 &l24_dynamic_handler,
123 &opus_dynamic_handler,
124 &realmedia_mp3_dynamic_handler,
125 &speex_dynamic_handler,
126 &t140_dynamic_handler,
127 /* rdt */
128 &ff_rdt_video_handler,
129 &ff_rdt_audio_handler,
130 &ff_rdt_live_video_handler,
131 &ff_rdt_live_audio_handler,
132 NULL,
133 };
134
ff_rtp_handler_iterate(void ** opaque)135 const RTPDynamicProtocolHandler *ff_rtp_handler_iterate(void **opaque)
136 {
137 uintptr_t i = (uintptr_t)*opaque;
138 const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i];
139
140 if (r)
141 *opaque = (void*)(i + 1);
142
143 return r;
144 }
145
ff_rtp_handler_find_by_name(const char * name,enum AVMediaType codec_type)146 const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
147 enum AVMediaType codec_type)
148 {
149 void *i = 0;
150 const RTPDynamicProtocolHandler *handler;
151 while (handler = ff_rtp_handler_iterate(&i)) {
152 if (handler->enc_name &&
153 !av_strcasecmp(name, handler->enc_name) &&
154 codec_type == handler->codec_type)
155 return handler;
156 }
157 return NULL;
158 }
159
ff_rtp_handler_find_by_id(int id,enum AVMediaType codec_type)160 const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
161 enum AVMediaType codec_type)
162 {
163 void *i = 0;
164 const RTPDynamicProtocolHandler *handler;
165 while (handler = ff_rtp_handler_iterate(&i)) {
166 if (handler->static_payload_id && handler->static_payload_id == id &&
167 codec_type == handler->codec_type)
168 return handler;
169 }
170 return NULL;
171 }
172
rtcp_parse_packet(RTPDemuxContext * s,const unsigned char * buf,int len)173 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
174 int len)
175 {
176 int payload_len;
177 while (len >= 4) {
178 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
179
180 switch (buf[1]) {
181 case RTCP_SR:
182 if (payload_len < 20) {
183 av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
184 return AVERROR_INVALIDDATA;
185 }
186
187 s->last_rtcp_reception_time = av_gettime_relative();
188 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
189 s->last_rtcp_timestamp = AV_RB32(buf + 16);
190 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
191 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
192 if (!s->base_timestamp)
193 s->base_timestamp = s->last_rtcp_timestamp;
194 s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
195 }
196
197 break;
198 case RTCP_BYE:
199 return -RTCP_BYE;
200 }
201
202 buf += payload_len;
203 len -= payload_len;
204 }
205 return -1;
206 }
207
208 #define RTP_SEQ_MOD (1 << 16)
209
rtp_init_statistics(RTPStatistics * s,uint16_t base_sequence)210 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
211 {
212 memset(s, 0, sizeof(RTPStatistics));
213 s->max_seq = base_sequence;
214 s->probation = 1;
215 }
216
217 /*
218 * Called whenever there is a large jump in sequence numbers,
219 * or when they get out of probation...
220 */
rtp_init_sequence(RTPStatistics * s,uint16_t seq)221 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
222 {
223 s->max_seq = seq;
224 s->cycles = 0;
225 s->base_seq = seq - 1;
226 s->bad_seq = RTP_SEQ_MOD + 1;
227 s->received = 0;
228 s->expected_prior = 0;
229 s->received_prior = 0;
230 s->jitter = 0;
231 s->transit = 0;
232 }
233
234 /* Returns 1 if we should handle this packet. */
rtp_valid_packet_in_sequence(RTPStatistics * s,uint16_t seq)235 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
236 {
237 uint16_t udelta = seq - s->max_seq;
238 const int MAX_DROPOUT = 3000;
239 const int MAX_MISORDER = 100;
240 const int MIN_SEQUENTIAL = 2;
241
242 /* source not valid until MIN_SEQUENTIAL packets with sequence
243 * seq. numbers have been received */
244 if (s->probation) {
245 if (seq == s->max_seq + 1) {
246 s->probation--;
247 s->max_seq = seq;
248 if (s->probation == 0) {
249 rtp_init_sequence(s, seq);
250 s->received++;
251 return 1;
252 }
253 } else {
254 s->probation = MIN_SEQUENTIAL - 1;
255 s->max_seq = seq;
256 }
257 } else if (udelta < MAX_DROPOUT) {
258 // in order, with permissible gap
259 if (seq < s->max_seq) {
260 // sequence number wrapped; count another 64k cycles
261 s->cycles += RTP_SEQ_MOD;
262 }
263 s->max_seq = seq;
264 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
265 // sequence made a large jump...
266 if (seq == s->bad_seq) {
267 /* two sequential packets -- assume that the other side
268 * restarted without telling us; just resync. */
269 rtp_init_sequence(s, seq);
270 } else {
271 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
272 return 0;
273 }
274 } else {
275 // duplicate or reordered packet...
276 }
277 s->received++;
278 return 1;
279 }
280
rtcp_update_jitter(RTPStatistics * s,uint32_t sent_timestamp,uint32_t arrival_timestamp)281 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
282 uint32_t arrival_timestamp)
283 {
284 // Most of this is pretty straight from RFC 3550 appendix A.8
285 uint32_t transit = arrival_timestamp - sent_timestamp;
286 uint32_t prev_transit = s->transit;
287 int32_t d = transit - prev_transit;
288 // Doing the FFABS() call directly on the "transit - prev_transit"
289 // expression doesn't work, since it's an unsigned expression. Doing the
290 // transit calculation in unsigned is desired though, since it most
291 // probably will need to wrap around.
292 d = FFABS(d);
293 s->transit = transit;
294 if (!prev_transit)
295 return;
296 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
297 }
298
ff_rtp_check_and_send_back_rr(RTPDemuxContext * s,URLContext * fd,AVIOContext * avio,int count)299 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
300 AVIOContext *avio, int count)
301 {
302 AVIOContext *pb;
303 uint8_t *buf;
304 int len;
305 int rtcp_bytes;
306 RTPStatistics *stats = &s->statistics;
307 uint32_t lost;
308 uint32_t extended_max;
309 uint32_t expected_interval;
310 uint32_t received_interval;
311 int32_t lost_interval;
312 uint32_t expected;
313 uint32_t fraction;
314
315 if ((!fd && !avio) || (count < 1))
316 return -1;
317
318 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
319 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
320 s->octet_count += count;
321 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
322 RTCP_TX_RATIO_DEN;
323 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
324 if (rtcp_bytes < 28)
325 return -1;
326 s->last_octet_count = s->octet_count;
327
328 if (!fd)
329 pb = avio;
330 else if (avio_open_dyn_buf(&pb) < 0)
331 return -1;
332
333 // Receiver Report
334 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
335 avio_w8(pb, RTCP_RR);
336 avio_wb16(pb, 7); /* length in words - 1 */
337 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
338 avio_wb32(pb, s->ssrc + 1);
339 avio_wb32(pb, s->ssrc); // server SSRC
340 // some placeholders we should really fill...
341 // RFC 1889/p64
342 extended_max = stats->cycles + stats->max_seq;
343 expected = extended_max - stats->base_seq;
344 lost = expected - stats->received;
345 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
346 expected_interval = expected - stats->expected_prior;
347 stats->expected_prior = expected;
348 received_interval = stats->received - stats->received_prior;
349 stats->received_prior = stats->received;
350 lost_interval = expected_interval - received_interval;
351 if (expected_interval == 0 || lost_interval <= 0)
352 fraction = 0;
353 else
354 fraction = (lost_interval << 8) / expected_interval;
355
356 fraction = (fraction << 24) | lost;
357
358 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
359 avio_wb32(pb, extended_max); /* max sequence received */
360 avio_wb32(pb, stats->jitter >> 4); /* jitter */
361
362 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
363 avio_wb32(pb, 0); /* last SR timestamp */
364 avio_wb32(pb, 0); /* delay since last SR */
365 } else {
366 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
367 uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
368 65536, AV_TIME_BASE);
369
370 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
371 avio_wb32(pb, delay_since_last); /* delay since last SR */
372 }
373
374 // CNAME
375 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
376 avio_w8(pb, RTCP_SDES);
377 len = strlen(s->hostname);
378 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
379 avio_wb32(pb, s->ssrc + 1);
380 avio_w8(pb, 0x01);
381 avio_w8(pb, len);
382 avio_write(pb, s->hostname, len);
383 avio_w8(pb, 0); /* END */
384 // padding
385 for (len = (7 + len) % 4; len % 4; len++)
386 avio_w8(pb, 0);
387
388 avio_flush(pb);
389 if (!fd)
390 return 0;
391 len = avio_close_dyn_buf(pb, &buf);
392 if ((len > 0) && buf) {
393 int av_unused result;
394 av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
395 result = ffurl_write(fd, buf, len);
396 av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
397 av_free(buf);
398 }
399 return 0;
400 }
401
ff_rtp_send_punch_packets(URLContext * rtp_handle)402 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
403 {
404 AVIOContext *pb;
405 uint8_t *buf;
406 int len;
407
408 /* Send a small RTP packet */
409 if (avio_open_dyn_buf(&pb) < 0)
410 return;
411
412 avio_w8(pb, (RTP_VERSION << 6));
413 avio_w8(pb, 0); /* Payload type */
414 avio_wb16(pb, 0); /* Seq */
415 avio_wb32(pb, 0); /* Timestamp */
416 avio_wb32(pb, 0); /* SSRC */
417
418 len = avio_close_dyn_buf(pb, &buf);
419 if ((len > 0) && buf)
420 ffurl_write(rtp_handle, buf, len);
421 av_free(buf);
422
423 /* Send a minimal RTCP RR */
424 if (avio_open_dyn_buf(&pb) < 0)
425 return;
426
427 avio_w8(pb, (RTP_VERSION << 6));
428 avio_w8(pb, RTCP_RR); /* receiver report */
429 avio_wb16(pb, 1); /* length in words - 1 */
430 avio_wb32(pb, 0); /* our own SSRC */
431
432 len = avio_close_dyn_buf(pb, &buf);
433 if ((len > 0) && buf)
434 ffurl_write(rtp_handle, buf, len);
435 av_free(buf);
436 }
437
find_missing_packets(RTPDemuxContext * s,uint16_t * first_missing,uint16_t * missing_mask)438 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
439 uint16_t *missing_mask)
440 {
441 int i;
442 uint16_t next_seq = s->seq + 1;
443 RTPPacket *pkt = s->queue;
444
445 if (!pkt || pkt->seq == next_seq)
446 return 0;
447
448 *missing_mask = 0;
449 for (i = 1; i <= 16; i++) {
450 uint16_t missing_seq = next_seq + i;
451 while (pkt) {
452 int16_t diff = pkt->seq - missing_seq;
453 if (diff >= 0)
454 break;
455 pkt = pkt->next;
456 }
457 if (!pkt)
458 break;
459 if (pkt->seq == missing_seq)
460 continue;
461 *missing_mask |= 1 << (i - 1);
462 }
463
464 *first_missing = next_seq;
465 return 1;
466 }
467
ff_rtp_send_rtcp_feedback(RTPDemuxContext * s,URLContext * fd,AVIOContext * avio)468 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
469 AVIOContext *avio)
470 {
471 int len, need_keyframe, missing_packets;
472 AVIOContext *pb;
473 uint8_t *buf;
474 int64_t now;
475 uint16_t first_missing = 0, missing_mask = 0;
476
477 if (!fd && !avio)
478 return -1;
479
480 need_keyframe = s->handler && s->handler->need_keyframe &&
481 s->handler->need_keyframe(s->dynamic_protocol_context);
482 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
483
484 if (!need_keyframe && !missing_packets)
485 return 0;
486
487 /* Send new feedback if enough time has elapsed since the last
488 * feedback packet. */
489
490 now = av_gettime_relative();
491 if (s->last_feedback_time &&
492 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
493 return 0;
494 s->last_feedback_time = now;
495
496 if (!fd)
497 pb = avio;
498 else if (avio_open_dyn_buf(&pb) < 0)
499 return -1;
500
501 if (need_keyframe) {
502 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
503 avio_w8(pb, RTCP_PSFB);
504 avio_wb16(pb, 2); /* length in words - 1 */
505 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
506 avio_wb32(pb, s->ssrc + 1);
507 avio_wb32(pb, s->ssrc); // server SSRC
508 }
509
510 if (missing_packets) {
511 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
512 avio_w8(pb, RTCP_RTPFB);
513 avio_wb16(pb, 3); /* length in words - 1 */
514 avio_wb32(pb, s->ssrc + 1);
515 avio_wb32(pb, s->ssrc); // server SSRC
516
517 avio_wb16(pb, first_missing);
518 avio_wb16(pb, missing_mask);
519 }
520
521 avio_flush(pb);
522 if (!fd)
523 return 0;
524 len = avio_close_dyn_buf(pb, &buf);
525 if (len > 0 && buf) {
526 ffurl_write(fd, buf, len);
527 av_free(buf);
528 }
529 return 0;
530 }
531
532 /**
533 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
534 * MPEG-2 TS streams.
535 */
ff_rtp_parse_open(AVFormatContext * s1,AVStream * st,int payload_type,int queue_size)536 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
537 int payload_type, int queue_size)
538 {
539 RTPDemuxContext *s;
540
541 s = av_mallocz(sizeof(RTPDemuxContext));
542 if (!s)
543 return NULL;
544 s->payload_type = payload_type;
545 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
546 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
547 s->ic = s1;
548 s->st = st;
549 s->queue_size = queue_size;
550
551 av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
552 s->queue_size);
553
554 rtp_init_statistics(&s->statistics, 0);
555 if (st) {
556 switch (st->codecpar->codec_id) {
557 case AV_CODEC_ID_ADPCM_G722:
558 /* According to RFC 3551, the stream clock rate is 8000
559 * even if the sample rate is 16000. */
560 if (st->codecpar->sample_rate == 8000)
561 st->codecpar->sample_rate = 16000;
562 break;
563 default:
564 break;
565 }
566 }
567 // needed to send back RTCP RR in RTSP sessions
568 gethostname(s->hostname, sizeof(s->hostname));
569 return s;
570 }
571
ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext * s,PayloadContext * ctx,const RTPDynamicProtocolHandler * handler)572 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
573 const RTPDynamicProtocolHandler *handler)
574 {
575 s->dynamic_protocol_context = ctx;
576 s->handler = handler;
577 }
578
ff_rtp_parse_set_crypto(RTPDemuxContext * s,const char * suite,const char * params)579 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
580 const char *params)
581 {
582 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
583 s->srtp_enabled = 1;
584 }
585
586 /**
587 * This was the second switch in rtp_parse packet.
588 * Normalizes time, if required, sets stream_index, etc.
589 */
finalize_packet(RTPDemuxContext * s,AVPacket * pkt,uint32_t timestamp)590 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
591 {
592 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
593 return; /* Timestamp already set by depacketizer */
594 if (timestamp == RTP_NOTS_VALUE)
595 return;
596
597 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
598 int64_t addend;
599 int delta_timestamp;
600
601 /* compute pts from timestamp with received ntp_time */
602 delta_timestamp = timestamp - s->last_rtcp_timestamp;
603 /* convert to the PTS timebase */
604 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
605 s->st->time_base.den,
606 (uint64_t) s->st->time_base.num << 32);
607 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
608 delta_timestamp;
609 return;
610 }
611
612 if (!s->base_timestamp)
613 s->base_timestamp = timestamp;
614 /* assume that the difference is INT32_MIN < x < INT32_MAX,
615 * but allow the first timestamp to exceed INT32_MAX */
616 if (!s->timestamp)
617 s->unwrapped_timestamp += timestamp;
618 else
619 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
620 s->timestamp = timestamp;
621 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
622 s->base_timestamp;
623 }
624
rtp_parse_packet_internal(RTPDemuxContext * s,AVPacket * pkt,const uint8_t * buf,int len)625 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
626 const uint8_t *buf, int len)
627 {
628 unsigned int ssrc;
629 int payload_type, seq, flags = 0;
630 int ext, csrc;
631 AVStream *st;
632 uint32_t timestamp;
633 int rv = 0;
634
635 csrc = buf[0] & 0x0f;
636 ext = buf[0] & 0x10;
637 payload_type = buf[1] & 0x7f;
638 if (buf[1] & 0x80)
639 flags |= RTP_FLAG_MARKER;
640 seq = AV_RB16(buf + 2);
641 timestamp = AV_RB32(buf + 4);
642 ssrc = AV_RB32(buf + 8);
643 /* store the ssrc in the RTPDemuxContext */
644 s->ssrc = ssrc;
645
646 /* NOTE: we can handle only one payload type */
647 if (s->payload_type != payload_type)
648 return -1;
649
650 st = s->st;
651 // only do something with this if all the rtp checks pass...
652 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
653 av_log(s->ic, AV_LOG_ERROR,
654 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
655 payload_type, seq, ((s->seq + 1) & 0xffff));
656 return -1;
657 }
658
659 if (buf[0] & 0x20) {
660 int padding = buf[len - 1];
661 if (len >= 12 + padding)
662 len -= padding;
663 }
664
665 s->seq = seq;
666 len -= 12;
667 buf += 12;
668
669 len -= 4 * csrc;
670 buf += 4 * csrc;
671 if (len < 0)
672 return AVERROR_INVALIDDATA;
673
674 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
675 if (ext) {
676 if (len < 4)
677 return -1;
678 /* calculate the header extension length (stored as number
679 * of 32-bit words) */
680 ext = (AV_RB16(buf + 2) + 1) << 2;
681
682 if (len < ext)
683 return -1;
684 // skip past RTP header extension
685 len -= ext;
686 buf += ext;
687 }
688
689 if (s->handler && s->handler->parse_packet) {
690 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
691 s->st, pkt, ×tamp, buf, len, seq,
692 flags);
693 } else if (st) {
694 if ((rv = av_new_packet(pkt, len)) < 0)
695 return rv;
696 memcpy(pkt->data, buf, len);
697 pkt->stream_index = st->index;
698 } else {
699 return AVERROR(EINVAL);
700 }
701
702 // now perform timestamp things....
703 finalize_packet(s, pkt, timestamp);
704
705 return rv;
706 }
707
ff_rtp_reset_packet_queue(RTPDemuxContext * s)708 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
709 {
710 while (s->queue) {
711 RTPPacket *next = s->queue->next;
712 av_freep(&s->queue->buf);
713 av_freep(&s->queue);
714 s->queue = next;
715 }
716 s->seq = 0;
717 s->queue_len = 0;
718 s->prev_ret = 0;
719 }
720
enqueue_packet(RTPDemuxContext * s,uint8_t * buf,int len)721 static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
722 {
723 uint16_t seq = AV_RB16(buf + 2);
724 RTPPacket **cur = &s->queue, *packet;
725
726 /* Find the correct place in the queue to insert the packet */
727 while (*cur) {
728 int16_t diff = seq - (*cur)->seq;
729 if (diff < 0)
730 break;
731 cur = &(*cur)->next;
732 }
733
734 packet = av_mallocz(sizeof(*packet));
735 if (!packet)
736 return AVERROR(ENOMEM);
737 packet->recvtime = av_gettime_relative();
738 packet->seq = seq;
739 packet->len = len;
740 packet->buf = buf;
741 packet->next = *cur;
742 *cur = packet;
743 s->queue_len++;
744
745 return 0;
746 }
747
has_next_packet(RTPDemuxContext * s)748 static int has_next_packet(RTPDemuxContext *s)
749 {
750 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
751 }
752
ff_rtp_queued_packet_time(RTPDemuxContext * s)753 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
754 {
755 return s->queue ? s->queue->recvtime : 0;
756 }
757
rtp_parse_queued_packet(RTPDemuxContext * s,AVPacket * pkt)758 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
759 {
760 int rv;
761 RTPPacket *next;
762
763 if (s->queue_len <= 0)
764 return -1;
765
766 if (!has_next_packet(s))
767 av_log(s->ic, AV_LOG_WARNING,
768 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
769
770 /* Parse the first packet in the queue, and dequeue it */
771 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
772 next = s->queue->next;
773 av_freep(&s->queue->buf);
774 av_freep(&s->queue);
775 s->queue = next;
776 s->queue_len--;
777 return rv;
778 }
779
rtp_parse_one_packet(RTPDemuxContext * s,AVPacket * pkt,uint8_t ** bufptr,int len)780 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
781 uint8_t **bufptr, int len)
782 {
783 uint8_t *buf = bufptr ? *bufptr : NULL;
784 int flags = 0;
785 uint32_t timestamp;
786 int rv = 0;
787
788 if (!buf) {
789 /* If parsing of the previous packet actually returned 0 or an error,
790 * there's nothing more to be parsed from that packet, but we may have
791 * indicated that we can return the next enqueued packet. */
792 if (s->prev_ret <= 0)
793 return rtp_parse_queued_packet(s, pkt);
794 /* return the next packets, if any */
795 if (s->handler && s->handler->parse_packet) {
796 /* timestamp should be overwritten by parse_packet, if not,
797 * the packet is left with pts == AV_NOPTS_VALUE */
798 timestamp = RTP_NOTS_VALUE;
799 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
800 s->st, pkt, ×tamp, NULL, 0, 0,
801 flags);
802 finalize_packet(s, pkt, timestamp);
803 return rv;
804 }
805 }
806
807 if (len < 12)
808 return -1;
809
810 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
811 return -1;
812 if (RTP_PT_IS_RTCP(buf[1])) {
813 return rtcp_parse_packet(s, buf, len);
814 }
815
816 if (s->st) {
817 int64_t received = av_gettime_relative();
818 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
819 s->st->time_base);
820 timestamp = AV_RB32(buf + 4);
821 // Calculate the jitter immediately, before queueing the packet
822 // into the reordering queue.
823 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
824 }
825
826 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
827 /* First packet, or no reordering */
828 return rtp_parse_packet_internal(s, pkt, buf, len);
829 } else {
830 uint16_t seq = AV_RB16(buf + 2);
831 int16_t diff = seq - s->seq;
832 if (diff < 0) {
833 /* Packet older than the previously emitted one, drop */
834 av_log(s->ic, AV_LOG_WARNING,
835 "RTP: dropping old packet received too late\n");
836 return -1;
837 } else if (diff <= 1) {
838 /* Correct packet */
839 rv = rtp_parse_packet_internal(s, pkt, buf, len);
840 return rv;
841 } else {
842 /* Still missing some packet, enqueue this one. */
843 rv = enqueue_packet(s, buf, len);
844 if (rv < 0)
845 return rv;
846 *bufptr = NULL;
847 /* Return the first enqueued packet if the queue is full,
848 * even if we're missing something */
849 if (s->queue_len >= s->queue_size) {
850 av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
851 return rtp_parse_queued_packet(s, pkt);
852 }
853 return -1;
854 }
855 }
856 }
857
858 /**
859 * Parse an RTP or RTCP packet directly sent as a buffer.
860 * @param s RTP parse context.
861 * @param pkt returned packet
862 * @param bufptr pointer to the input buffer or NULL to read the next packets
863 * @param len buffer len
864 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
865 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
866 */
ff_rtp_parse_packet(RTPDemuxContext * s,AVPacket * pkt,uint8_t ** bufptr,int len)867 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
868 uint8_t **bufptr, int len)
869 {
870 int rv;
871 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
872 return -1;
873 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
874 s->prev_ret = rv;
875 while (rv < 0 && has_next_packet(s))
876 rv = rtp_parse_queued_packet(s, pkt);
877 return rv ? rv : has_next_packet(s);
878 }
879
ff_rtp_parse_close(RTPDemuxContext * s)880 void ff_rtp_parse_close(RTPDemuxContext *s)
881 {
882 ff_rtp_reset_packet_queue(s);
883 ff_srtp_free(&s->srtp);
884 av_free(s);
885 }
886
ff_parse_fmtp(AVFormatContext * s,AVStream * stream,PayloadContext * data,const char * p,int (* parse_fmtp)(AVFormatContext * s,AVStream * stream,PayloadContext * data,const char * attr,const char * value))887 int ff_parse_fmtp(AVFormatContext *s,
888 AVStream *stream, PayloadContext *data, const char *p,
889 int (*parse_fmtp)(AVFormatContext *s,
890 AVStream *stream,
891 PayloadContext *data,
892 const char *attr, const char *value))
893 {
894 char attr[256];
895 char *value;
896 int res;
897 int value_size = strlen(p) + 1;
898
899 if (!(value = av_malloc(value_size))) {
900 av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
901 return AVERROR(ENOMEM);
902 }
903
904 // remove protocol identifier
905 while (*p && *p == ' ')
906 p++; // strip spaces
907 while (*p && *p != ' ')
908 p++; // eat protocol identifier
909 while (*p && *p == ' ')
910 p++; // strip trailing spaces
911
912 while (ff_rtsp_next_attr_and_value(&p,
913 attr, sizeof(attr),
914 value, value_size)) {
915 res = parse_fmtp(s, stream, data, attr, value);
916 if (res < 0 && res != AVERROR_PATCHWELCOME) {
917 av_free(value);
918 return res;
919 }
920 }
921 av_free(value);
922 return 0;
923 }
924
ff_rtp_finalize_packet(AVPacket * pkt,AVIOContext ** dyn_buf,int stream_idx)925 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
926 {
927 int ret;
928 av_init_packet(pkt);
929
930 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
931 pkt->stream_index = stream_idx;
932 *dyn_buf = NULL;
933 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
934 av_freep(&pkt->data);
935 return ret;
936 }
937 return pkt->size;
938 }
939