rtpdec: Simplify finalize_packet
[libav.git] / libavformat / rtpdec.c
1 /*
2 * RTP input format
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
7 * Libav 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 * Libav 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 Libav; 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 "libavcodec/get_bits.h"
25 #include "avformat.h"
26 #include "mpegts.h"
27 #include "url.h"
28
29 #include <unistd.h>
30 #include "network.h"
31
32 #include "rtpdec.h"
33 #include "rtpdec_formats.h"
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 'ffio_open_dyn_packet_buf')
44 */
45
46 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
47 .enc_name = "X-MP3-draft-00",
48 .codec_type = AVMEDIA_TYPE_AUDIO,
49 .codec_id = CODEC_ID_MP3ADU,
50 };
51
52 /* statistics functions */
53 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
54
55 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
56 {
57 handler->next= RTPFirstDynamicPayloadHandler;
58 RTPFirstDynamicPayloadHandler= handler;
59 }
60
61 void av_register_rtp_dynamic_payload_handlers(void)
62 {
63 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
78
79 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
81
82 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
84 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
86 }
87
88 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
89 enum AVMediaType codec_type)
90 {
91 RTPDynamicProtocolHandler *handler;
92 for (handler = RTPFirstDynamicPayloadHandler;
93 handler; handler = handler->next)
94 if (!av_strcasecmp(name, handler->enc_name) &&
95 codec_type == handler->codec_type)
96 return handler;
97 return NULL;
98 }
99
100 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
101 enum AVMediaType codec_type)
102 {
103 RTPDynamicProtocolHandler *handler;
104 for (handler = RTPFirstDynamicPayloadHandler;
105 handler; handler = handler->next)
106 if (handler->static_payload_id && handler->static_payload_id == id &&
107 codec_type == handler->codec_type)
108 return handler;
109 return NULL;
110 }
111
112 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
113 {
114 int payload_len;
115 while (len >= 4) {
116 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
117
118 switch (buf[1]) {
119 case RTCP_SR:
120 if (payload_len < 20) {
121 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
122 return AVERROR_INVALIDDATA;
123 }
124
125 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
126 s->last_rtcp_timestamp = AV_RB32(buf + 16);
127 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
128 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
129 if (!s->base_timestamp)
130 s->base_timestamp = s->last_rtcp_timestamp;
131 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
132 }
133
134 break;
135 case RTCP_BYE:
136 return -RTCP_BYE;
137 }
138
139 buf += payload_len;
140 len -= payload_len;
141 }
142 return -1;
143 }
144
145 #define RTP_SEQ_MOD (1<<16)
146
147 /**
148 * called on parse open packet
149 */
150 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
151 {
152 memset(s, 0, sizeof(RTPStatistics));
153 s->max_seq= base_sequence;
154 s->probation= 1;
155 }
156
157 /**
158 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
159 */
160 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
161 {
162 s->max_seq= seq;
163 s->cycles= 0;
164 s->base_seq= seq -1;
165 s->bad_seq= RTP_SEQ_MOD + 1;
166 s->received= 0;
167 s->expected_prior= 0;
168 s->received_prior= 0;
169 s->jitter= 0;
170 s->transit= 0;
171 }
172
173 /**
174 * returns 1 if we should handle this packet.
175 */
176 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
177 {
178 uint16_t udelta= seq - s->max_seq;
179 const int MAX_DROPOUT= 3000;
180 const int MAX_MISORDER = 100;
181 const int MIN_SEQUENTIAL = 2;
182
183 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
184 if(s->probation)
185 {
186 if(seq==s->max_seq + 1) {
187 s->probation--;
188 s->max_seq= seq;
189 if(s->probation==0) {
190 rtp_init_sequence(s, seq);
191 s->received++;
192 return 1;
193 }
194 } else {
195 s->probation= MIN_SEQUENTIAL - 1;
196 s->max_seq = seq;
197 }
198 } else if (udelta < MAX_DROPOUT) {
199 // in order, with permissible gap
200 if(seq < s->max_seq) {
201 //sequence number wrapped; count antother 64k cycles
202 s->cycles += RTP_SEQ_MOD;
203 }
204 s->max_seq= seq;
205 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
206 // sequence made a large jump...
207 if(seq==s->bad_seq) {
208 // two sequential packets-- assume that the other side restarted without telling us; just resync.
209 rtp_init_sequence(s, seq);
210 } else {
211 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
212 return 0;
213 }
214 } else {
215 // duplicate or reordered packet...
216 }
217 s->received++;
218 return 1;
219 }
220
221 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
222 {
223 AVIOContext *pb;
224 uint8_t *buf;
225 int len;
226 int rtcp_bytes;
227 RTPStatistics *stats= &s->statistics;
228 uint32_t lost;
229 uint32_t extended_max;
230 uint32_t expected_interval;
231 uint32_t received_interval;
232 uint32_t lost_interval;
233 uint32_t expected;
234 uint32_t fraction;
235 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
236
237 if (!s->rtp_ctx || (count < 1))
238 return -1;
239
240 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
241 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
242 s->octet_count += count;
243 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
244 RTCP_TX_RATIO_DEN;
245 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
246 if (rtcp_bytes < 28)
247 return -1;
248 s->last_octet_count = s->octet_count;
249
250 if (avio_open_dyn_buf(&pb) < 0)
251 return -1;
252
253 // Receiver Report
254 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
255 avio_w8(pb, RTCP_RR);
256 avio_wb16(pb, 7); /* length in words - 1 */
257 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
258 avio_wb32(pb, s->ssrc + 1);
259 avio_wb32(pb, s->ssrc); // server SSRC
260 // some placeholders we should really fill...
261 // RFC 1889/p64
262 extended_max= stats->cycles + stats->max_seq;
263 expected= extended_max - stats->base_seq + 1;
264 lost= expected - stats->received;
265 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
266 expected_interval= expected - stats->expected_prior;
267 stats->expected_prior= expected;
268 received_interval= stats->received - stats->received_prior;
269 stats->received_prior= stats->received;
270 lost_interval= expected_interval - received_interval;
271 if (expected_interval==0 || lost_interval<=0) fraction= 0;
272 else fraction = (lost_interval<<8)/expected_interval;
273
274 fraction= (fraction<<24) | lost;
275
276 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
277 avio_wb32(pb, extended_max); /* max sequence received */
278 avio_wb32(pb, stats->jitter>>4); /* jitter */
279
280 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
281 {
282 avio_wb32(pb, 0); /* last SR timestamp */
283 avio_wb32(pb, 0); /* delay since last SR */
284 } else {
285 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
286 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
287
288 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
289 avio_wb32(pb, delay_since_last); /* delay since last SR */
290 }
291
292 // CNAME
293 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
294 avio_w8(pb, RTCP_SDES);
295 len = strlen(s->hostname);
296 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
297 avio_wb32(pb, s->ssrc);
298 avio_w8(pb, 0x01);
299 avio_w8(pb, len);
300 avio_write(pb, s->hostname, len);
301 // padding
302 for (len = (6 + len) % 4; len % 4; len++) {
303 avio_w8(pb, 0);
304 }
305
306 avio_flush(pb);
307 len = avio_close_dyn_buf(pb, &buf);
308 if ((len > 0) && buf) {
309 int av_unused result;
310 av_dlog(s->ic, "sending %d bytes of RR\n", len);
311 result= ffurl_write(s->rtp_ctx, buf, len);
312 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
313 av_free(buf);
314 }
315 return 0;
316 }
317
318 void ff_rtp_send_punch_packets(URLContext* rtp_handle)
319 {
320 AVIOContext *pb;
321 uint8_t *buf;
322 int len;
323
324 /* Send a small RTP packet */
325 if (avio_open_dyn_buf(&pb) < 0)
326 return;
327
328 avio_w8(pb, (RTP_VERSION << 6));
329 avio_w8(pb, 0); /* Payload type */
330 avio_wb16(pb, 0); /* Seq */
331 avio_wb32(pb, 0); /* Timestamp */
332 avio_wb32(pb, 0); /* SSRC */
333
334 avio_flush(pb);
335 len = avio_close_dyn_buf(pb, &buf);
336 if ((len > 0) && buf)
337 ffurl_write(rtp_handle, buf, len);
338 av_free(buf);
339
340 /* Send a minimal RTCP RR */
341 if (avio_open_dyn_buf(&pb) < 0)
342 return;
343
344 avio_w8(pb, (RTP_VERSION << 6));
345 avio_w8(pb, RTCP_RR); /* receiver report */
346 avio_wb16(pb, 1); /* length in words - 1 */
347 avio_wb32(pb, 0); /* our own SSRC */
348
349 avio_flush(pb);
350 len = avio_close_dyn_buf(pb, &buf);
351 if ((len > 0) && buf)
352 ffurl_write(rtp_handle, buf, len);
353 av_free(buf);
354 }
355
356
357 /**
358 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
359 * MPEG2TS streams to indicate that they should be demuxed inside the
360 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
361 */
362 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
363 {
364 RTPDemuxContext *s;
365
366 s = av_mallocz(sizeof(RTPDemuxContext));
367 if (!s)
368 return NULL;
369 s->payload_type = payload_type;
370 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
371 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
372 s->ic = s1;
373 s->st = st;
374 s->queue_size = queue_size;
375 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
376 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
377 s->ts = ff_mpegts_parse_open(s->ic);
378 if (s->ts == NULL) {
379 av_free(s);
380 return NULL;
381 }
382 } else {
383 switch(st->codec->codec_id) {
384 case CODEC_ID_MPEG1VIDEO:
385 case CODEC_ID_MPEG2VIDEO:
386 case CODEC_ID_MP2:
387 case CODEC_ID_MP3:
388 case CODEC_ID_MPEG4:
389 case CODEC_ID_H263:
390 case CODEC_ID_H264:
391 st->need_parsing = AVSTREAM_PARSE_FULL;
392 break;
393 case CODEC_ID_ADPCM_G722:
394 /* According to RFC 3551, the stream clock rate is 8000
395 * even if the sample rate is 16000. */
396 if (st->codec->sample_rate == 8000)
397 st->codec->sample_rate = 16000;
398 break;
399 default:
400 break;
401 }
402 }
403 // needed to send back RTCP RR in RTSP sessions
404 s->rtp_ctx = rtpc;
405 gethostname(s->hostname, sizeof(s->hostname));
406 return s;
407 }
408
409 void
410 ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
411 RTPDynamicProtocolHandler *handler)
412 {
413 s->dynamic_protocol_context = ctx;
414 s->parse_packet = handler->parse_packet;
415 }
416
417 /**
418 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
419 */
420 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
421 {
422 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
423 return; /* Timestamp already set by depacketizer */
424 if (timestamp == RTP_NOTS_VALUE)
425 return;
426
427 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
428 int64_t addend;
429 int delta_timestamp;
430
431 /* compute pts from timestamp with received ntp_time */
432 delta_timestamp = timestamp - s->last_rtcp_timestamp;
433 /* convert to the PTS timebase */
434 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);
435 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
436 delta_timestamp;
437 return;
438 }
439
440 if (!s->base_timestamp)
441 s->base_timestamp = timestamp;
442 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
443 }
444
445 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
446 const uint8_t *buf, int len)
447 {
448 unsigned int ssrc, h;
449 int payload_type, seq, ret, flags = 0;
450 int ext;
451 AVStream *st;
452 uint32_t timestamp;
453 int rv= 0;
454
455 ext = buf[0] & 0x10;
456 payload_type = buf[1] & 0x7f;
457 if (buf[1] & 0x80)
458 flags |= RTP_FLAG_MARKER;
459 seq = AV_RB16(buf + 2);
460 timestamp = AV_RB32(buf + 4);
461 ssrc = AV_RB32(buf + 8);
462 /* store the ssrc in the RTPDemuxContext */
463 s->ssrc = ssrc;
464
465 /* NOTE: we can handle only one payload type */
466 if (s->payload_type != payload_type)
467 return -1;
468
469 st = s->st;
470 // only do something with this if all the rtp checks pass...
471 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
472 {
473 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
474 payload_type, seq, ((s->seq + 1) & 0xffff));
475 return -1;
476 }
477
478 if (buf[0] & 0x20) {
479 int padding = buf[len - 1];
480 if (len >= 12 + padding)
481 len -= padding;
482 }
483
484 s->seq = seq;
485 len -= 12;
486 buf += 12;
487
488 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
489 if (ext) {
490 if (len < 4)
491 return -1;
492 /* calculate the header extension length (stored as number
493 * of 32-bit words) */
494 ext = (AV_RB16(buf + 2) + 1) << 2;
495
496 if (len < ext)
497 return -1;
498 // skip past RTP header extension
499 len -= ext;
500 buf += ext;
501 }
502
503 if (!st) {
504 /* specific MPEG2TS demux support */
505 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
506 /* The only error that can be returned from ff_mpegts_parse_packet
507 * is "no more data to return from the provided buffer", so return
508 * AVERROR(EAGAIN) for all errors */
509 if (ret < 0)
510 return AVERROR(EAGAIN);
511 if (ret < len) {
512 s->read_buf_size = len - ret;
513 memcpy(s->buf, buf + ret, s->read_buf_size);
514 s->read_buf_index = 0;
515 return 1;
516 }
517 return 0;
518 } else if (s->parse_packet) {
519 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
520 s->st, pkt, &timestamp, buf, len, flags);
521 } else {
522 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
523 switch(st->codec->codec_id) {
524 case CODEC_ID_MP2:
525 case CODEC_ID_MP3:
526 /* better than nothing: skip mpeg audio RTP header */
527 if (len <= 4)
528 return -1;
529 h = AV_RB32(buf);
530 len -= 4;
531 buf += 4;
532 av_new_packet(pkt, len);
533 memcpy(pkt->data, buf, len);
534 break;
535 case CODEC_ID_MPEG1VIDEO:
536 case CODEC_ID_MPEG2VIDEO:
537 /* better than nothing: skip mpeg video RTP header */
538 if (len <= 4)
539 return -1;
540 h = AV_RB32(buf);
541 buf += 4;
542 len -= 4;
543 if (h & (1 << 26)) {
544 /* mpeg2 */
545 if (len <= 4)
546 return -1;
547 buf += 4;
548 len -= 4;
549 }
550 av_new_packet(pkt, len);
551 memcpy(pkt->data, buf, len);
552 break;
553 default:
554 av_new_packet(pkt, len);
555 memcpy(pkt->data, buf, len);
556 break;
557 }
558
559 pkt->stream_index = st->index;
560 }
561
562 // now perform timestamp things....
563 finalize_packet(s, pkt, timestamp);
564
565 return rv;
566 }
567
568 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
569 {
570 while (s->queue) {
571 RTPPacket *next = s->queue->next;
572 av_free(s->queue->buf);
573 av_free(s->queue);
574 s->queue = next;
575 }
576 s->seq = 0;
577 s->queue_len = 0;
578 s->prev_ret = 0;
579 }
580
581 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
582 {
583 uint16_t seq = AV_RB16(buf + 2);
584 RTPPacket *cur = s->queue, *prev = NULL, *packet;
585
586 /* Find the correct place in the queue to insert the packet */
587 while (cur) {
588 int16_t diff = seq - cur->seq;
589 if (diff < 0)
590 break;
591 prev = cur;
592 cur = cur->next;
593 }
594
595 packet = av_mallocz(sizeof(*packet));
596 if (!packet)
597 return;
598 packet->recvtime = av_gettime();
599 packet->seq = seq;
600 packet->len = len;
601 packet->buf = buf;
602 packet->next = cur;
603 if (prev)
604 prev->next = packet;
605 else
606 s->queue = packet;
607 s->queue_len++;
608 }
609
610 static int has_next_packet(RTPDemuxContext *s)
611 {
612 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
613 }
614
615 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
616 {
617 return s->queue ? s->queue->recvtime : 0;
618 }
619
620 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
621 {
622 int rv;
623 RTPPacket *next;
624
625 if (s->queue_len <= 0)
626 return -1;
627
628 if (!has_next_packet(s))
629 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
630 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
631
632 /* Parse the first packet in the queue, and dequeue it */
633 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
634 next = s->queue->next;
635 av_free(s->queue->buf);
636 av_free(s->queue);
637 s->queue = next;
638 s->queue_len--;
639 return rv;
640 }
641
642 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
643 uint8_t **bufptr, int len)
644 {
645 uint8_t* buf = bufptr ? *bufptr : NULL;
646 int ret, flags = 0;
647 uint32_t timestamp;
648 int rv= 0;
649
650 if (!buf) {
651 /* If parsing of the previous packet actually returned 0 or an error,
652 * there's nothing more to be parsed from that packet, but we may have
653 * indicated that we can return the next enqueued packet. */
654 if (s->prev_ret <= 0)
655 return rtp_parse_queued_packet(s, pkt);
656 /* return the next packets, if any */
657 if(s->st && s->parse_packet) {
658 /* timestamp should be overwritten by parse_packet, if not,
659 * the packet is left with pts == AV_NOPTS_VALUE */
660 timestamp = RTP_NOTS_VALUE;
661 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
662 s->st, pkt, &timestamp, NULL, 0, flags);
663 finalize_packet(s, pkt, timestamp);
664 return rv;
665 } else {
666 // TODO: Move to a dynamic packet handler (like above)
667 if (s->read_buf_index >= s->read_buf_size)
668 return AVERROR(EAGAIN);
669 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
670 s->read_buf_size - s->read_buf_index);
671 if (ret < 0)
672 return AVERROR(EAGAIN);
673 s->read_buf_index += ret;
674 if (s->read_buf_index < s->read_buf_size)
675 return 1;
676 else
677 return 0;
678 }
679 }
680
681 if (len < 12)
682 return -1;
683
684 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
685 return -1;
686 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
687 return rtcp_parse_packet(s, buf, len);
688 }
689
690 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
691 /* First packet, or no reordering */
692 return rtp_parse_packet_internal(s, pkt, buf, len);
693 } else {
694 uint16_t seq = AV_RB16(buf + 2);
695 int16_t diff = seq - s->seq;
696 if (diff < 0) {
697 /* Packet older than the previously emitted one, drop */
698 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
699 "RTP: dropping old packet received too late\n");
700 return -1;
701 } else if (diff <= 1) {
702 /* Correct packet */
703 rv = rtp_parse_packet_internal(s, pkt, buf, len);
704 return rv;
705 } else {
706 /* Still missing some packet, enqueue this one. */
707 enqueue_packet(s, buf, len);
708 *bufptr = NULL;
709 /* Return the first enqueued packet if the queue is full,
710 * even if we're missing something */
711 if (s->queue_len >= s->queue_size)
712 return rtp_parse_queued_packet(s, pkt);
713 return -1;
714 }
715 }
716 }
717
718 /**
719 * Parse an RTP or RTCP packet directly sent as a buffer.
720 * @param s RTP parse context.
721 * @param pkt returned packet
722 * @param bufptr pointer to the input buffer or NULL to read the next packets
723 * @param len buffer len
724 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
725 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
726 */
727 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
728 uint8_t **bufptr, int len)
729 {
730 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
731 s->prev_ret = rv;
732 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
733 rv = rtp_parse_queued_packet(s, pkt);
734 return rv ? rv : has_next_packet(s);
735 }
736
737 void ff_rtp_parse_close(RTPDemuxContext *s)
738 {
739 ff_rtp_reset_packet_queue(s);
740 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
741 ff_mpegts_parse_close(s->ts);
742 }
743 av_free(s);
744 }
745
746 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
747 int (*parse_fmtp)(AVStream *stream,
748 PayloadContext *data,
749 char *attr, char *value))
750 {
751 char attr[256];
752 char *value;
753 int res;
754 int value_size = strlen(p) + 1;
755
756 if (!(value = av_malloc(value_size))) {
757 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
758 return AVERROR(ENOMEM);
759 }
760
761 // remove protocol identifier
762 while (*p && *p == ' ') p++; // strip spaces
763 while (*p && *p != ' ') p++; // eat protocol identifier
764 while (*p && *p == ' ') p++; // strip trailing spaces
765
766 while (ff_rtsp_next_attr_and_value(&p,
767 attr, sizeof(attr),
768 value, value_size)) {
769
770 res = parse_fmtp(stream, data, attr, value);
771 if (res < 0 && res != AVERROR_PATCHWELCOME) {
772 av_free(value);
773 return res;
774 }
775 }
776 av_free(value);
777 return 0;
778 }