rtpdec: Read RTCP compound packets
[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>
30#include "network.h"
31
302879cb 32#include "rtpdec.h"
965a3ddb 33#include "rtpdec_formats.h"
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34
35//#define DEBUG
36
37/* TODO: - add RTCP statistics reporting (should be optional).
38
39 - add support for h263/mpeg4 packetized output : IDEA: send a
40 buffer to 'rtp_write_packet' contains all the packets for ONE
41 frame. Each packet should have a four byte header containing
42 the length in big endian format (same trick as
43 'url_open_dyn_packet_buf')
44*/
45
46/* statistics functions */
47RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
48
0369d2b0 49void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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50{
51 handler->next= RTPFirstDynamicPayloadHandler;
52 RTPFirstDynamicPayloadHandler= handler;
53}
54
55void av_register_rtp_dynamic_payload_handlers(void)
56{
9b3788ef
JA
57 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
58 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
556aa7a1
RB
59 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
60 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
45aa9080
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61 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
62 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 63 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 64 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
887af2aa 65 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
a59096e4 66 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
4449df6b 67 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
1ddc176e 68 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
51291e60 69 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
e9fce261
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70
71 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
72 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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73}
74
75static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
76{
ff328c02
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77 int payload_len;
78 while (len >= 2) {
79 switch (buf[1]) {
80 case RTCP_SR:
81 if (len < 16) {
82 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
83 return AVERROR_INVALIDDATA;
84 }
85 payload_len = (AV_RB16(buf + 2) + 1) * 4;
86
8eb793c4 87 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
2cab6b48
MS
88 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
89 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
8eb793c4 90 s->last_rtcp_timestamp = AV_RB32(buf + 16);
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91
92 buf += payload_len;
93 len -= payload_len;
94 break;
95 default:
96 return -1;
97 }
98 }
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99 return 0;
100}
101
102#define RTP_SEQ_MOD (1<<16)
103
104/**
105* called on parse open packet
106*/
107static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
108{
109 memset(s, 0, sizeof(RTPStatistics));
110 s->max_seq= base_sequence;
111 s->probation= 1;
112}
113
114/**
115* called whenever there is a large jump in sequence numbers, or when they get out of probation...
116*/
117static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
118{
119 s->max_seq= seq;
120 s->cycles= 0;
121 s->base_seq= seq -1;
122 s->bad_seq= RTP_SEQ_MOD + 1;
123 s->received= 0;
124 s->expected_prior= 0;
125 s->received_prior= 0;
126 s->jitter= 0;
127 s->transit= 0;
128}
129
130/**
131* returns 1 if we should handle this packet.
132*/
133static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
134{
135 uint16_t udelta= seq - s->max_seq;
136 const int MAX_DROPOUT= 3000;
137 const int MAX_MISORDER = 100;
138 const int MIN_SEQUENTIAL = 2;
139
140 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
141 if(s->probation)
142 {
143 if(seq==s->max_seq + 1) {
144 s->probation--;
145 s->max_seq= seq;
146 if(s->probation==0) {
147 rtp_init_sequence(s, seq);
148 s->received++;
149 return 1;
150 }
151 } else {
152 s->probation= MIN_SEQUENTIAL - 1;
153 s->max_seq = seq;
154 }
155 } else if (udelta < MAX_DROPOUT) {
156 // in order, with permissible gap
157 if(seq < s->max_seq) {
158 //sequence number wrapped; count antother 64k cycles
159 s->cycles += RTP_SEQ_MOD;
160 }
161 s->max_seq= seq;
162 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
163 // sequence made a large jump...
164 if(seq==s->bad_seq) {
165 // two sequential packets-- assume that the other side restarted without telling us; just resync.
166 rtp_init_sequence(s, seq);
167 } else {
168 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
169 return 0;
170 }
171 } else {
172 // duplicate or reordered packet...
173 }
174 s->received++;
175 return 1;
176}
177
178#if 0
179/**
180* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
181* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
182* never change. I left this in in case someone else can see a way. (rdm)
183*/
184static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
185{
186 uint32_t transit= arrival_timestamp - sent_timestamp;
187 int d;
188 s->transit= transit;
189 d= FFABS(transit - s->transit);
190 s->jitter += d - ((s->jitter + 8)>>4);
191}
192#endif
193
194int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
195{
196 ByteIOContext *pb;
197 uint8_t *buf;
198 int len;
199 int rtcp_bytes;
200 RTPStatistics *stats= &s->statistics;
201 uint32_t lost;
202 uint32_t extended_max;
203 uint32_t expected_interval;
204 uint32_t received_interval;
205 uint32_t lost_interval;
206 uint32_t expected;
207 uint32_t fraction;
208 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
209
210 if (!s->rtp_ctx || (count < 1))
211 return -1;
212
213 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
214 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
215 s->octet_count += count;
216 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
217 RTCP_TX_RATIO_DEN;
218 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
219 if (rtcp_bytes < 28)
220 return -1;
221 s->last_octet_count = s->octet_count;
222
223 if (url_open_dyn_buf(&pb) < 0)
224 return -1;
225
226 // Receiver Report
227 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 228 put_byte(pb, RTCP_RR);
8eb793c4 229 put_be16(pb, 7); /* length in words - 1 */
952139a3
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230 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
231 put_be32(pb, s->ssrc + 1);
232 put_be32(pb, s->ssrc); // server SSRC
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233 // some placeholders we should really fill...
234 // RFC 1889/p64
235 extended_max= stats->cycles + stats->max_seq;
236 expected= extended_max - stats->base_seq + 1;
237 lost= expected - stats->received;
238 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
239 expected_interval= expected - stats->expected_prior;
240 stats->expected_prior= expected;
241 received_interval= stats->received - stats->received_prior;
242 stats->received_prior= stats->received;
243 lost_interval= expected_interval - received_interval;
244 if (expected_interval==0 || lost_interval<=0) fraction= 0;
245 else fraction = (lost_interval<<8)/expected_interval;
246
247 fraction= (fraction<<24) | lost;
248
249 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
250 put_be32(pb, extended_max); /* max sequence received */
251 put_be32(pb, stats->jitter>>4); /* jitter */
252
253 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
254 {
255 put_be32(pb, 0); /* last SR timestamp */
256 put_be32(pb, 0); /* delay since last SR */
257 } else {
258 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
259 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
260
261 put_be32(pb, middle_32_bits); /* last SR timestamp */
262 put_be32(pb, delay_since_last); /* delay since last SR */
263 }
264
265 // CNAME
266 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 267 put_byte(pb, RTCP_SDES);
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268 len = strlen(s->hostname);
269 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
270 put_be32(pb, s->ssrc);
271 put_byte(pb, 0x01);
272 put_byte(pb, len);
273 put_buffer(pb, s->hostname, len);
274 // padding
275 for (len = (6 + len) % 4; len % 4; len++) {
276 put_byte(pb, 0);
277 }
278
279 put_flush_packet(pb);
280 len = url_close_dyn_buf(pb, &buf);
281 if ((len > 0) && buf) {
282 int result;
e8420626 283 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 284 result= url_write(s->rtp_ctx, buf, len);
e8420626 285 dprintf(s->ic, "result from url_write: %d\n", result);
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286 av_free(buf);
287 }
288 return 0;
289}
290
9c8fa20d
MS
291void rtp_send_punch_packets(URLContext* rtp_handle)
292{
293 ByteIOContext *pb;
294 uint8_t *buf;
295 int len;
296
297 /* Send a small RTP packet */
298 if (url_open_dyn_buf(&pb) < 0)
299 return;
300
301 put_byte(pb, (RTP_VERSION << 6));
302 put_byte(pb, 0); /* Payload type */
303 put_be16(pb, 0); /* Seq */
304 put_be32(pb, 0); /* Timestamp */
305 put_be32(pb, 0); /* SSRC */
306
307 put_flush_packet(pb);
308 len = url_close_dyn_buf(pb, &buf);
309 if ((len > 0) && buf)
310 url_write(rtp_handle, buf, len);
311 av_free(buf);
312
313 /* Send a minimal RTCP RR */
314 if (url_open_dyn_buf(&pb) < 0)
315 return;
316
317 put_byte(pb, (RTP_VERSION << 6));
7f3468d3 318 put_byte(pb, RTCP_RR); /* receiver report */
9c8fa20d
MS
319 put_be16(pb, 1); /* length in words - 1 */
320 put_be32(pb, 0); /* our own SSRC */
321
322 put_flush_packet(pb);
323 len = url_close_dyn_buf(pb, &buf);
324 if ((len > 0) && buf)
325 url_write(rtp_handle, buf, len);
326 av_free(buf);
327}
328
329
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330/**
331 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
332 * MPEG2TS streams to indicate that they should be demuxed inside the
333 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 334 */
ca937a55 335RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type)
8eb793c4
LA
336{
337 RTPDemuxContext *s;
338
339 s = av_mallocz(sizeof(RTPDemuxContext));
340 if (!s)
341 return NULL;
342 s->payload_type = payload_type;
343 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 344 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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345 s->ic = s1;
346 s->st = st;
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347 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
348 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 349 s->ts = ff_mpegts_parse_open(s->ic);
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LA
350 if (s->ts == NULL) {
351 av_free(s);
352 return NULL;
353 }
354 } else {
26efefc5 355 av_set_pts_info(st, 32, 1, 90000);
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LA
356 switch(st->codec->codec_id) {
357 case CODEC_ID_MPEG1VIDEO:
358 case CODEC_ID_MPEG2VIDEO:
359 case CODEC_ID_MP2:
360 case CODEC_ID_MP3:
361 case CODEC_ID_MPEG4:
45aa9080 362 case CODEC_ID_H263:
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LA
363 case CODEC_ID_H264:
364 st->need_parsing = AVSTREAM_PARSE_FULL;
365 break;
366 default:
72415b2a 367 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
d6b9e57a
LA
368 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
369 }
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370 break;
371 }
372 }
373 // needed to send back RTCP RR in RTSP sessions
374 s->rtp_ctx = rtpc;
375 gethostname(s->hostname, sizeof(s->hostname));
376 return s;
377}
378
99a1d191
RB
379void
380rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
381 RTPDynamicProtocolHandler *handler)
382{
383 s->dynamic_protocol_context = ctx;
384 s->parse_packet = handler->parse_packet;
385}
386
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387/**
388 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
389 */
390static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
391{
d74c6145 392 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
fba7815d
LA
393 int64_t addend;
394 int delta_timestamp;
395
396 /* compute pts from timestamp with received ntp_time */
397 delta_timestamp = timestamp - s->last_rtcp_timestamp;
398 /* convert to the PTS timebase */
2cab6b48 399 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 400 pkt->pts = s->range_start_offset + addend + delta_timestamp;
fba7815d 401 }
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402}
403
404/**
405 * Parse an RTP or RTCP packet directly sent as a buffer.
406 * @param s RTP parse context.
407 * @param pkt returned packet
408 * @param buf input buffer or NULL to read the next packets
409 * @param len buffer len
410 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
411 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
412 */
413int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
414 const uint8_t *buf, int len)
415{
416 unsigned int ssrc, h;
f841a0fc 417 int payload_type, seq, ret, flags = 0;
8eb793c4
LA
418 AVStream *st;
419 uint32_t timestamp;
420 int rv= 0;
421
422 if (!buf) {
423 /* return the next packets, if any */
424 if(s->st && s->parse_packet) {
d74c6145
MS
425 /* timestamp should be overwritten by parse_packet, if not,
426 * the packet is left with pts == AV_NOPTS_VALUE */
427 timestamp = RTP_NOTS_VALUE;
1a45a9f4 428 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 429 s->st, pkt, &timestamp, NULL, 0, flags);
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LA
430 finalize_packet(s, pkt, timestamp);
431 return rv;
432 } else {
433 // TODO: Move to a dynamic packet handler (like above)
434 if (s->read_buf_index >= s->read_buf_size)
435 return -1;
9125806e 436 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
8eb793c4
LA
437 s->read_buf_size - s->read_buf_index);
438 if (ret < 0)
439 return -1;
440 s->read_buf_index += ret;
441 if (s->read_buf_index < s->read_buf_size)
442 return 1;
443 else
444 return 0;
445 }
446 }
447
448 if (len < 12)
449 return -1;
450
451 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
452 return -1;
7f3468d3 453 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
8eb793c4
LA
454 rtcp_parse_packet(s, buf, len);
455 return -1;
456 }
457 payload_type = buf[1] & 0x7f;
144ae29d
RB
458 if (buf[1] & 0x80)
459 flags |= RTP_FLAG_MARKER;
8eb793c4
LA
460 seq = AV_RB16(buf + 2);
461 timestamp = AV_RB32(buf + 4);
462 ssrc = AV_RB32(buf + 8);
463 /* store the ssrc in the RTPDemuxContext */
464 s->ssrc = ssrc;
465
466 /* NOTE: we can handle only one payload type */
467 if (s->payload_type != payload_type)
468 return -1;
469
470 st = s->st;
471 // only do something with this if all the rtp checks pass...
472 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
473 {
474 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
475 payload_type, seq, ((s->seq + 1) & 0xffff));
476 return -1;
477 }
478
479 s->seq = seq;
480 len -= 12;
481 buf += 12;
482
483 if (!st) {
484 /* specific MPEG2TS demux support */
9125806e 485 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
8eb793c4
LA
486 if (ret < 0)
487 return -1;
488 if (ret < len) {
489 s->read_buf_size = len - ret;
490 memcpy(s->buf, buf + ret, s->read_buf_size);
491 s->read_buf_index = 0;
492 return 1;
493 }
f3e71942 494 return 0;
b4e3330c 495 } else if (s->parse_packet) {
1a45a9f4 496 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 497 s->st, pkt, &timestamp, buf, len, flags);
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LA
498 } else {
499 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
500 switch(st->codec->codec_id) {
501 case CODEC_ID_MP2:
76faff6e 502 case CODEC_ID_MP3:
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LA
503 /* better than nothing: skip mpeg audio RTP header */
504 if (len <= 4)
505 return -1;
506 h = AV_RB32(buf);
507 len -= 4;
508 buf += 4;
509 av_new_packet(pkt, len);
510 memcpy(pkt->data, buf, len);
511 break;
512 case CODEC_ID_MPEG1VIDEO:
513 case CODEC_ID_MPEG2VIDEO:
514 /* better than nothing: skip mpeg video RTP header */
515 if (len <= 4)
516 return -1;
517 h = AV_RB32(buf);
518 buf += 4;
519 len -= 4;
520 if (h & (1 << 26)) {
521 /* mpeg2 */
522 if (len <= 4)
523 return -1;
524 buf += 4;
525 len -= 4;
526 }
527 av_new_packet(pkt, len);
528 memcpy(pkt->data, buf, len);
529 break;
8eb793c4 530 default:
f739b36d
RB
531 av_new_packet(pkt, len);
532 memcpy(pkt->data, buf, len);
8eb793c4
LA
533 break;
534 }
eafb17d1
RB
535
536 pkt->stream_index = st->index;
f3e71942 537 }
8eb793c4 538
95f03cf3
RB
539 // now perform timestamp things....
540 finalize_packet(s, pkt, timestamp);
f3e71942 541
8eb793c4
LA
542 return rv;
543}
544
545void rtp_parse_close(RTPDemuxContext *s)
546{
8eb793c4 547 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 548 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
549 }
550 av_free(s);
551}
016bc031
JA
552
553int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
554 int (*parse_fmtp)(AVStream *stream,
555 PayloadContext *data,
556 char *attr, char *value))
557{
558 char attr[256];
824535e3 559 char *value;
016bc031 560 int res;
824535e3
JA
561 int value_size = strlen(p) + 1;
562
563 if (!(value = av_malloc(value_size))) {
564 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
565 return AVERROR(ENOMEM);
566 }
016bc031
JA
567
568 // remove protocol identifier
569 while (*p && *p == ' ') p++; // strip spaces
570 while (*p && *p != ' ') p++; // eat protocol identifier
571 while (*p && *p == ' ') p++; // strip trailing spaces
572
573 while (ff_rtsp_next_attr_and_value(&p,
574 attr, sizeof(attr),
824535e3 575 value, value_size)) {
016bc031
JA
576
577 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
578 if (res < 0 && res != AVERROR_PATCHWELCOME) {
579 av_free(value);
016bc031 580 return res;
824535e3 581 }
016bc031 582 }
824535e3 583 av_free(value);
016bc031
JA
584 return 0;
585}