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path: root/libavformat/rtpdec.c (plain)
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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
36static RTPDynamicProtocolHandler l24_dynamic_handler = {
37 .enc_name = "L24",
38 .codec_type = AVMEDIA_TYPE_AUDIO,
39 .codec_id = AV_CODEC_ID_PCM_S24BE,
40};
41
42static RTPDynamicProtocolHandler gsm_dynamic_handler = {
43 .enc_name = "GSM",
44 .codec_type = AVMEDIA_TYPE_AUDIO,
45 .codec_id = AV_CODEC_ID_GSM,
46};
47
48static 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
54static RTPDynamicProtocolHandler speex_dynamic_handler = {
55 .enc_name = "speex",
56 .codec_type = AVMEDIA_TYPE_AUDIO,
57 .codec_id = AV_CODEC_ID_SPEEX,
58};
59
60static RTPDynamicProtocolHandler opus_dynamic_handler = {
61 .enc_name = "opus",
62 .codec_type = AVMEDIA_TYPE_AUDIO,
63 .codec_id = AV_CODEC_ID_OPUS,
64};
65
66static 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
72static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
73
74void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
75{
76 handler->next = rtp_first_dynamic_payload_handler;
77 rtp_first_dynamic_payload_handler = handler;
78}
79
80void ff_register_rtp_dynamic_payload_handlers(void)
81{
82 ff_register_dynamic_payload_handler(&ff_ac3_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_dv_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_g726le_16_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_g726le_24_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_g726le_32_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_g726le_40_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler);
95 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
96 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
97 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
98 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
99 ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler);
100 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
101 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
102 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
103 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
104 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
105 ff_register_dynamic_payload_handler(&ff_mpeg_audio_robust_dynamic_handler);
106 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
107 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
108 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
109 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
110 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
111 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
112 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
113 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
114 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
115 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
116 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
117 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
118 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
119 ff_register_dynamic_payload_handler(&ff_vc2hq_dynamic_handler);
120 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
121 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
122 ff_register_dynamic_payload_handler(&ff_vp9_dynamic_handler);
123 ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
124 ff_register_dynamic_payload_handler(&l24_dynamic_handler);
125 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
126 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
127 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
128 ff_register_dynamic_payload_handler(&t140_dynamic_handler);
129}
130
131RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
132 enum AVMediaType codec_type)
133{
134 RTPDynamicProtocolHandler *handler;
135 for (handler = rtp_first_dynamic_payload_handler;
136 handler; handler = handler->next)
137 if (handler->enc_name &&
138 !av_strcasecmp(name, handler->enc_name) &&
139 codec_type == handler->codec_type)
140 return handler;
141 return NULL;
142}
143
144RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
145 enum AVMediaType codec_type)
146{
147 RTPDynamicProtocolHandler *handler;
148 for (handler = rtp_first_dynamic_payload_handler;
149 handler; handler = handler->next)
150 if (handler->static_payload_id && handler->static_payload_id == id &&
151 codec_type == handler->codec_type)
152 return handler;
153 return NULL;
154}
155
156static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
157 int len)
158{
159 int payload_len;
160 while (len >= 4) {
161 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
162
163 switch (buf[1]) {
164 case RTCP_SR:
165 if (payload_len < 20) {
166 av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
167 return AVERROR_INVALIDDATA;
168 }
169
170 s->last_rtcp_reception_time = av_gettime_relative();
171 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
172 s->last_rtcp_timestamp = AV_RB32(buf + 16);
173 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
174 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
175 if (!s->base_timestamp)
176 s->base_timestamp = s->last_rtcp_timestamp;
177 s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
178 }
179
180 break;
181 case RTCP_BYE:
182 return -RTCP_BYE;
183 }
184
185 buf += payload_len;
186 len -= payload_len;
187 }
188 return -1;
189}
190
191#define RTP_SEQ_MOD (1 << 16)
192
193static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
194{
195 memset(s, 0, sizeof(RTPStatistics));
196 s->max_seq = base_sequence;
197 s->probation = 1;
198}
199
200/*
201 * Called whenever there is a large jump in sequence numbers,
202 * or when they get out of probation...
203 */
204static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
205{
206 s->max_seq = seq;
207 s->cycles = 0;
208 s->base_seq = seq - 1;
209 s->bad_seq = RTP_SEQ_MOD + 1;
210 s->received = 0;
211 s->expected_prior = 0;
212 s->received_prior = 0;
213 s->jitter = 0;
214 s->transit = 0;
215}
216
217/* Returns 1 if we should handle this packet. */
218static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
219{
220 uint16_t udelta = seq - s->max_seq;
221 const int MAX_DROPOUT = 3000;
222 const int MAX_MISORDER = 100;
223 const int MIN_SEQUENTIAL = 2;
224
225 /* source not valid until MIN_SEQUENTIAL packets with sequence
226 * seq. numbers have been received */
227 if (s->probation) {
228 if (seq == s->max_seq + 1) {
229 s->probation--;
230 s->max_seq = seq;
231 if (s->probation == 0) {
232 rtp_init_sequence(s, seq);
233 s->received++;
234 return 1;
235 }
236 } else {
237 s->probation = MIN_SEQUENTIAL - 1;
238 s->max_seq = seq;
239 }
240 } else if (udelta < MAX_DROPOUT) {
241 // in order, with permissible gap
242 if (seq < s->max_seq) {
243 // sequence number wrapped; count another 64k cycles
244 s->cycles += RTP_SEQ_MOD;
245 }
246 s->max_seq = seq;
247 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
248 // sequence made a large jump...
249 if (seq == s->bad_seq) {
250 /* two sequential packets -- assume that the other side
251 * restarted without telling us; just resync. */
252 rtp_init_sequence(s, seq);
253 } else {
254 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
255 return 0;
256 }
257 } else {
258 // duplicate or reordered packet...
259 }
260 s->received++;
261 return 1;
262}
263
264static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
265 uint32_t arrival_timestamp)
266{
267 // Most of this is pretty straight from RFC 3550 appendix A.8
268 uint32_t transit = arrival_timestamp - sent_timestamp;
269 uint32_t prev_transit = s->transit;
270 int32_t d = transit - prev_transit;
271 // Doing the FFABS() call directly on the "transit - prev_transit"
272 // expression doesn't work, since it's an unsigned expression. Doing the
273 // transit calculation in unsigned is desired though, since it most
274 // probably will need to wrap around.
275 d = FFABS(d);
276 s->transit = transit;
277 if (!prev_transit)
278 return;
279 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
280}
281
282int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
283 AVIOContext *avio, int count)
284{
285 AVIOContext *pb;
286 uint8_t *buf;
287 int len;
288 int rtcp_bytes;
289 RTPStatistics *stats = &s->statistics;
290 uint32_t lost;
291 uint32_t extended_max;
292 uint32_t expected_interval;
293 uint32_t received_interval;
294 int32_t lost_interval;
295 uint32_t expected;
296 uint32_t fraction;
297
298 if ((!fd && !avio) || (count < 1))
299 return -1;
300
301 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
302 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
303 s->octet_count += count;
304 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
305 RTCP_TX_RATIO_DEN;
306 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
307 if (rtcp_bytes < 28)
308 return -1;
309 s->last_octet_count = s->octet_count;
310
311 if (!fd)
312 pb = avio;
313 else if (avio_open_dyn_buf(&pb) < 0)
314 return -1;
315
316 // Receiver Report
317 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
318 avio_w8(pb, RTCP_RR);
319 avio_wb16(pb, 7); /* length in words - 1 */
320 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
321 avio_wb32(pb, s->ssrc + 1);
322 avio_wb32(pb, s->ssrc); // server SSRC
323 // some placeholders we should really fill...
324 // RFC 1889/p64
325 extended_max = stats->cycles + stats->max_seq;
326 expected = extended_max - stats->base_seq;
327 lost = expected - stats->received;
328 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
329 expected_interval = expected - stats->expected_prior;
330 stats->expected_prior = expected;
331 received_interval = stats->received - stats->received_prior;
332 stats->received_prior = stats->received;
333 lost_interval = expected_interval - received_interval;
334 if (expected_interval == 0 || lost_interval <= 0)
335 fraction = 0;
336 else
337 fraction = (lost_interval << 8) / expected_interval;
338
339 fraction = (fraction << 24) | lost;
340
341 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
342 avio_wb32(pb, extended_max); /* max sequence received */
343 avio_wb32(pb, stats->jitter >> 4); /* jitter */
344
345 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
346 avio_wb32(pb, 0); /* last SR timestamp */
347 avio_wb32(pb, 0); /* delay since last SR */
348 } else {
349 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
350 uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
351 65536, AV_TIME_BASE);
352
353 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
354 avio_wb32(pb, delay_since_last); /* delay since last SR */
355 }
356
357 // CNAME
358 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
359 avio_w8(pb, RTCP_SDES);
360 len = strlen(s->hostname);
361 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
362 avio_wb32(pb, s->ssrc + 1);
363 avio_w8(pb, 0x01);
364 avio_w8(pb, len);
365 avio_write(pb, s->hostname, len);
366 avio_w8(pb, 0); /* END */
367 // padding
368 for (len = (7 + len) % 4; len % 4; len++)
369 avio_w8(pb, 0);
370
371 avio_flush(pb);
372 if (!fd)
373 return 0;
374 len = avio_close_dyn_buf(pb, &buf);
375 if ((len > 0) && buf) {
376 int av_unused result;
377 av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
378 result = ffurl_write(fd, buf, len);
379 av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
380 av_free(buf);
381 }
382 return 0;
383}
384
385void ff_rtp_send_punch_packets(URLContext *rtp_handle)
386{
387 AVIOContext *pb;
388 uint8_t *buf;
389 int len;
390
391 /* Send a small RTP packet */
392 if (avio_open_dyn_buf(&pb) < 0)
393 return;
394
395 avio_w8(pb, (RTP_VERSION << 6));
396 avio_w8(pb, 0); /* Payload type */
397 avio_wb16(pb, 0); /* Seq */
398 avio_wb32(pb, 0); /* Timestamp */
399 avio_wb32(pb, 0); /* SSRC */
400
401 avio_flush(pb);
402 len = avio_close_dyn_buf(pb, &buf);
403 if ((len > 0) && buf)
404 ffurl_write(rtp_handle, buf, len);
405 av_free(buf);
406
407 /* Send a minimal RTCP RR */
408 if (avio_open_dyn_buf(&pb) < 0)
409 return;
410
411 avio_w8(pb, (RTP_VERSION << 6));
412 avio_w8(pb, RTCP_RR); /* receiver report */
413 avio_wb16(pb, 1); /* length in words - 1 */
414 avio_wb32(pb, 0); /* our own SSRC */
415
416 avio_flush(pb);
417 len = avio_close_dyn_buf(pb, &buf);
418 if ((len > 0) && buf)
419 ffurl_write(rtp_handle, buf, len);
420 av_free(buf);
421}
422
423static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
424 uint16_t *missing_mask)
425{
426 int i;
427 uint16_t next_seq = s->seq + 1;
428 RTPPacket *pkt = s->queue;
429
430 if (!pkt || pkt->seq == next_seq)
431 return 0;
432
433 *missing_mask = 0;
434 for (i = 1; i <= 16; i++) {
435 uint16_t missing_seq = next_seq + i;
436 while (pkt) {
437 int16_t diff = pkt->seq - missing_seq;
438 if (diff >= 0)
439 break;
440 pkt = pkt->next;
441 }
442 if (!pkt)
443 break;
444 if (pkt->seq == missing_seq)
445 continue;
446 *missing_mask |= 1 << (i - 1);
447 }
448
449 *first_missing = next_seq;
450 return 1;
451}
452
453int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
454 AVIOContext *avio)
455{
456 int len, need_keyframe, missing_packets;
457 AVIOContext *pb;
458 uint8_t *buf;
459 int64_t now;
460 uint16_t first_missing = 0, missing_mask = 0;
461
462 if (!fd && !avio)
463 return -1;
464
465 need_keyframe = s->handler && s->handler->need_keyframe &&
466 s->handler->need_keyframe(s->dynamic_protocol_context);
467 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
468
469 if (!need_keyframe && !missing_packets)
470 return 0;
471
472 /* Send new feedback if enough time has elapsed since the last
473 * feedback packet. */
474
475 now = av_gettime_relative();
476 if (s->last_feedback_time &&
477 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
478 return 0;
479 s->last_feedback_time = now;
480
481 if (!fd)
482 pb = avio;
483 else if (avio_open_dyn_buf(&pb) < 0)
484 return -1;
485
486 if (need_keyframe) {
487 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
488 avio_w8(pb, RTCP_PSFB);
489 avio_wb16(pb, 2); /* length in words - 1 */
490 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
491 avio_wb32(pb, s->ssrc + 1);
492 avio_wb32(pb, s->ssrc); // server SSRC
493 }
494
495 if (missing_packets) {
496 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
497 avio_w8(pb, RTCP_RTPFB);
498 avio_wb16(pb, 3); /* length in words - 1 */
499 avio_wb32(pb, s->ssrc + 1);
500 avio_wb32(pb, s->ssrc); // server SSRC
501
502 avio_wb16(pb, first_missing);
503 avio_wb16(pb, missing_mask);
504 }
505
506 avio_flush(pb);
507 if (!fd)
508 return 0;
509 len = avio_close_dyn_buf(pb, &buf);
510 if (len > 0 && buf) {
511 ffurl_write(fd, buf, len);
512 av_free(buf);
513 }
514 return 0;
515}
516
517/**
518 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
519 * MPEG-2 TS streams.
520 */
521RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
522 int payload_type, int queue_size)
523{
524 RTPDemuxContext *s;
525
526 s = av_mallocz(sizeof(RTPDemuxContext));
527 if (!s)
528 return NULL;
529 s->payload_type = payload_type;
530 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
531 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
532 s->ic = s1;
533 s->st = st;
534 s->queue_size = queue_size;
535
536 av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
537 s->queue_size);
538
539 rtp_init_statistics(&s->statistics, 0);
540 if (st) {
541 switch (st->codecpar->codec_id) {
542 case AV_CODEC_ID_ADPCM_G722:
543 /* According to RFC 3551, the stream clock rate is 8000
544 * even if the sample rate is 16000. */
545 if (st->codecpar->sample_rate == 8000)
546 st->codecpar->sample_rate = 16000;
547 break;
548 default:
549 break;
550 }
551 }
552 // needed to send back RTCP RR in RTSP sessions
553 gethostname(s->hostname, sizeof(s->hostname));
554 return s;
555}
556
557void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
558 RTPDynamicProtocolHandler *handler)
559{
560 s->dynamic_protocol_context = ctx;
561 s->handler = handler;
562}
563
564void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
565 const char *params)
566{
567 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
568 s->srtp_enabled = 1;
569}
570
571/**
572 * This was the second switch in rtp_parse packet.
573 * Normalizes time, if required, sets stream_index, etc.
574 */
575static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
576{
577 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
578 return; /* Timestamp already set by depacketizer */
579 if (timestamp == RTP_NOTS_VALUE)
580 return;
581
582 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
583 int64_t addend;
584 int delta_timestamp;
585
586 /* compute pts from timestamp with received ntp_time */
587 delta_timestamp = timestamp - s->last_rtcp_timestamp;
588 /* convert to the PTS timebase */
589 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
590 s->st->time_base.den,
591 (uint64_t) s->st->time_base.num << 32);
592 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
593 delta_timestamp;
594 return;
595 }
596
597 if (!s->base_timestamp)
598 s->base_timestamp = timestamp;
599 /* assume that the difference is INT32_MIN < x < INT32_MAX,
600 * but allow the first timestamp to exceed INT32_MAX */
601 if (!s->timestamp)
602 s->unwrapped_timestamp += timestamp;
603 else
604 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
605 s->timestamp = timestamp;
606 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
607 s->base_timestamp;
608}
609
610static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
611 const uint8_t *buf, int len)
612{
613 unsigned int ssrc;
614 int payload_type, seq, flags = 0;
615 int ext, csrc;
616 AVStream *st;
617 uint32_t timestamp;
618 int rv = 0;
619
620 csrc = buf[0] & 0x0f;
621 ext = buf[0] & 0x10;
622 payload_type = buf[1] & 0x7f;
623 if (buf[1] & 0x80)
624 flags |= RTP_FLAG_MARKER;
625 seq = AV_RB16(buf + 2);
626 timestamp = AV_RB32(buf + 4);
627 ssrc = AV_RB32(buf + 8);
628 /* store the ssrc in the RTPDemuxContext */
629 s->ssrc = ssrc;
630
631 /* NOTE: we can handle only one payload type */
632 if (s->payload_type != payload_type)
633 return -1;
634
635 st = s->st;
636 // only do something with this if all the rtp checks pass...
637 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
638 av_log(s->ic, AV_LOG_ERROR,
639 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
640 payload_type, seq, ((s->seq + 1) & 0xffff));
641 return -1;
642 }
643
644 if (buf[0] & 0x20) {
645 int padding = buf[len - 1];
646 if (len >= 12 + padding)
647 len -= padding;
648 }
649
650 s->seq = seq;
651 len -= 12;
652 buf += 12;
653
654 len -= 4 * csrc;
655 buf += 4 * csrc;
656 if (len < 0)
657 return AVERROR_INVALIDDATA;
658
659 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
660 if (ext) {
661 if (len < 4)
662 return -1;
663 /* calculate the header extension length (stored as number
664 * of 32-bit words) */
665 ext = (AV_RB16(buf + 2) + 1) << 2;
666
667 if (len < ext)
668 return -1;
669 // skip past RTP header extension
670 len -= ext;
671 buf += ext;
672 }
673
674 if (s->handler && s->handler->parse_packet) {
675 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
676 s->st, pkt, &timestamp, buf, len, seq,
677 flags);
678 } else if (st) {
679 if ((rv = av_new_packet(pkt, len)) < 0)
680 return rv;
681 memcpy(pkt->data, buf, len);
682 pkt->stream_index = st->index;
683 } else {
684 return AVERROR(EINVAL);
685 }
686
687 // now perform timestamp things....
688 finalize_packet(s, pkt, timestamp);
689
690 return rv;
691}
692
693void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
694{
695 while (s->queue) {
696 RTPPacket *next = s->queue->next;
697 av_freep(&s->queue->buf);
698 av_freep(&s->queue);
699 s->queue = next;
700 }
701 s->seq = 0;
702 s->queue_len = 0;
703 s->prev_ret = 0;
704}
705
706static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
707{
708 uint16_t seq = AV_RB16(buf + 2);
709 RTPPacket **cur = &s->queue, *packet;
710
711 /* Find the correct place in the queue to insert the packet */
712 while (*cur) {
713 int16_t diff = seq - (*cur)->seq;
714 if (diff < 0)
715 break;
716 cur = &(*cur)->next;
717 }
718
719 packet = av_mallocz(sizeof(*packet));
720 if (!packet)
721 return AVERROR(ENOMEM);
722 packet->recvtime = av_gettime_relative();
723 packet->seq = seq;
724 packet->len = len;
725 packet->buf = buf;
726 packet->next = *cur;
727 *cur = packet;
728 s->queue_len++;
729
730 return 0;
731}
732
733static int has_next_packet(RTPDemuxContext *s)
734{
735 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
736}
737
738int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
739{
740 return s->queue ? s->queue->recvtime : 0;
741}
742
743static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
744{
745 int rv;
746 RTPPacket *next;
747
748 if (s->queue_len <= 0)
749 return -1;
750
751 if (!has_next_packet(s))
752 av_log(s->ic, AV_LOG_WARNING,
753 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
754
755 /* Parse the first packet in the queue, and dequeue it */
756 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
757 next = s->queue->next;
758 av_freep(&s->queue->buf);
759 av_freep(&s->queue);
760 s->queue = next;
761 s->queue_len--;
762 return rv;
763}
764
765static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
766 uint8_t **bufptr, int len)
767{
768 uint8_t *buf = bufptr ? *bufptr : NULL;
769 int flags = 0;
770 uint32_t timestamp;
771 int rv = 0;
772
773 if (!buf) {
774 /* If parsing of the previous packet actually returned 0 or an error,
775 * there's nothing more to be parsed from that packet, but we may have
776 * indicated that we can return the next enqueued packet. */
777 if (s->prev_ret <= 0)
778 return rtp_parse_queued_packet(s, pkt);
779 /* return the next packets, if any */
780 if (s->handler && s->handler->parse_packet) {
781 /* timestamp should be overwritten by parse_packet, if not,
782 * the packet is left with pts == AV_NOPTS_VALUE */
783 timestamp = RTP_NOTS_VALUE;
784 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
785 s->st, pkt, &timestamp, NULL, 0, 0,
786 flags);
787 finalize_packet(s, pkt, timestamp);
788 return rv;
789 }
790 }
791
792 if (len < 12)
793 return -1;
794
795 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
796 return -1;
797 if (RTP_PT_IS_RTCP(buf[1])) {
798 return rtcp_parse_packet(s, buf, len);
799 }
800
801 if (s->st) {
802 int64_t received = av_gettime_relative();
803 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
804 s->st->time_base);
805 timestamp = AV_RB32(buf + 4);
806 // Calculate the jitter immediately, before queueing the packet
807 // into the reordering queue.
808 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
809 }
810
811 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
812 /* First packet, or no reordering */
813 return rtp_parse_packet_internal(s, pkt, buf, len);
814 } else {
815 uint16_t seq = AV_RB16(buf + 2);
816 int16_t diff = seq - s->seq;
817 if (diff < 0) {
818 /* Packet older than the previously emitted one, drop */
819 av_log(s->ic, AV_LOG_WARNING,
820 "RTP: dropping old packet received too late\n");
821 return -1;
822 } else if (diff <= 1) {
823 /* Correct packet */
824 rv = rtp_parse_packet_internal(s, pkt, buf, len);
825 return rv;
826 } else {
827 /* Still missing some packet, enqueue this one. */
828 rv = enqueue_packet(s, buf, len);
829 if (rv < 0)
830 return rv;
831 *bufptr = NULL;
832 /* Return the first enqueued packet if the queue is full,
833 * even if we're missing something */
834 if (s->queue_len >= s->queue_size) {
835 av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
836 return rtp_parse_queued_packet(s, pkt);
837 }
838 return -1;
839 }
840 }
841}
842
843/**
844 * Parse an RTP or RTCP packet directly sent as a buffer.
845 * @param s RTP parse context.
846 * @param pkt returned packet
847 * @param bufptr pointer to the input buffer or NULL to read the next packets
848 * @param len buffer len
849 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
850 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
851 */
852int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
853 uint8_t **bufptr, int len)
854{
855 int rv;
856 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
857 return -1;
858 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
859 s->prev_ret = rv;
860 while (rv < 0 && has_next_packet(s))
861 rv = rtp_parse_queued_packet(s, pkt);
862 return rv ? rv : has_next_packet(s);
863}
864
865void ff_rtp_parse_close(RTPDemuxContext *s)
866{
867 ff_rtp_reset_packet_queue(s);
868 ff_srtp_free(&s->srtp);
869 av_free(s);
870}
871
872int ff_parse_fmtp(AVFormatContext *s,
873 AVStream *stream, PayloadContext *data, const char *p,
874 int (*parse_fmtp)(AVFormatContext *s,
875 AVStream *stream,
876 PayloadContext *data,
877 const char *attr, const char *value))
878{
879 char attr[256];
880 char *value;
881 int res;
882 int value_size = strlen(p) + 1;
883
884 if (!(value = av_malloc(value_size))) {
885 av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
886 return AVERROR(ENOMEM);
887 }
888
889 // remove protocol identifier
890 while (*p && *p == ' ')
891 p++; // strip spaces
892 while (*p && *p != ' ')
893 p++; // eat protocol identifier
894 while (*p && *p == ' ')
895 p++; // strip trailing spaces
896
897 while (ff_rtsp_next_attr_and_value(&p,
898 attr, sizeof(attr),
899 value, value_size)) {
900 res = parse_fmtp(s, stream, data, attr, value);
901 if (res < 0 && res != AVERROR_PATCHWELCOME) {
902 av_free(value);
903 return res;
904 }
905 }
906 av_free(value);
907 return 0;
908}
909
910int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
911{
912 int ret;
913 av_init_packet(pkt);
914
915 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
916 pkt->stream_index = stream_idx;
917 *dyn_buf = NULL;
918 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
919 av_freep(&pkt->data);
920 return ret;
921 }
922 return pkt->size;
923}
924
generated by cgit at 2024-05-17 02:01:14 (GMT)