Check avctx width/height more thoroughly (e.g. all values 0 except width would
[libav.git] / libavformat / rtpdec.c
CommitLineData
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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{
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57 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
58 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
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59 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
60 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
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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);
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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{
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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
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87 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
88 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
89 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
90 s->last_rtcp_timestamp = AV_RB32(buf + 16);
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91
92 buf += payload_len;
93 len -= payload_len;
94 break;
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95 case RTCP_BYE:
96 return -RTCP_BYE;
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97 default:
98 return -1;
99 }
100 }
b20359f5 101 return -1;
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102}
103
104#define RTP_SEQ_MOD (1<<16)
105
106/**
107* called on parse open packet
108*/
109static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
110{
111 memset(s, 0, sizeof(RTPStatistics));
112 s->max_seq= base_sequence;
113 s->probation= 1;
114}
115
116/**
117* called whenever there is a large jump in sequence numbers, or when they get out of probation...
118*/
119static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
120{
121 s->max_seq= seq;
122 s->cycles= 0;
123 s->base_seq= seq -1;
124 s->bad_seq= RTP_SEQ_MOD + 1;
125 s->received= 0;
126 s->expected_prior= 0;
127 s->received_prior= 0;
128 s->jitter= 0;
129 s->transit= 0;
130}
131
132/**
133* returns 1 if we should handle this packet.
134*/
135static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
136{
137 uint16_t udelta= seq - s->max_seq;
138 const int MAX_DROPOUT= 3000;
139 const int MAX_MISORDER = 100;
140 const int MIN_SEQUENTIAL = 2;
141
142 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
143 if(s->probation)
144 {
145 if(seq==s->max_seq + 1) {
146 s->probation--;
147 s->max_seq= seq;
148 if(s->probation==0) {
149 rtp_init_sequence(s, seq);
150 s->received++;
151 return 1;
152 }
153 } else {
154 s->probation= MIN_SEQUENTIAL - 1;
155 s->max_seq = seq;
156 }
157 } else if (udelta < MAX_DROPOUT) {
158 // in order, with permissible gap
159 if(seq < s->max_seq) {
160 //sequence number wrapped; count antother 64k cycles
161 s->cycles += RTP_SEQ_MOD;
162 }
163 s->max_seq= seq;
164 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
165 // sequence made a large jump...
166 if(seq==s->bad_seq) {
167 // two sequential packets-- assume that the other side restarted without telling us; just resync.
168 rtp_init_sequence(s, seq);
169 } else {
170 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
171 return 0;
172 }
173 } else {
174 // duplicate or reordered packet...
175 }
176 s->received++;
177 return 1;
178}
179
180#if 0
181/**
182* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
183* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
184* never change. I left this in in case someone else can see a way. (rdm)
185*/
186static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
187{
188 uint32_t transit= arrival_timestamp - sent_timestamp;
189 int d;
190 s->transit= transit;
191 d= FFABS(transit - s->transit);
192 s->jitter += d - ((s->jitter + 8)>>4);
193}
194#endif
195
196int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
197{
198 ByteIOContext *pb;
199 uint8_t *buf;
200 int len;
201 int rtcp_bytes;
202 RTPStatistics *stats= &s->statistics;
203 uint32_t lost;
204 uint32_t extended_max;
205 uint32_t expected_interval;
206 uint32_t received_interval;
207 uint32_t lost_interval;
208 uint32_t expected;
209 uint32_t fraction;
210 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
211
212 if (!s->rtp_ctx || (count < 1))
213 return -1;
214
215 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
216 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
217 s->octet_count += count;
218 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
219 RTCP_TX_RATIO_DEN;
220 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
221 if (rtcp_bytes < 28)
222 return -1;
223 s->last_octet_count = s->octet_count;
224
225 if (url_open_dyn_buf(&pb) < 0)
226 return -1;
227
228 // Receiver Report
229 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 230 put_byte(pb, RTCP_RR);
8eb793c4 231 put_be16(pb, 7); /* length in words - 1 */
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232 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
233 put_be32(pb, s->ssrc + 1);
234 put_be32(pb, s->ssrc); // server SSRC
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235 // some placeholders we should really fill...
236 // RFC 1889/p64
237 extended_max= stats->cycles + stats->max_seq;
238 expected= extended_max - stats->base_seq + 1;
239 lost= expected - stats->received;
240 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
241 expected_interval= expected - stats->expected_prior;
242 stats->expected_prior= expected;
243 received_interval= stats->received - stats->received_prior;
244 stats->received_prior= stats->received;
245 lost_interval= expected_interval - received_interval;
246 if (expected_interval==0 || lost_interval<=0) fraction= 0;
247 else fraction = (lost_interval<<8)/expected_interval;
248
249 fraction= (fraction<<24) | lost;
250
251 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
252 put_be32(pb, extended_max); /* max sequence received */
253 put_be32(pb, stats->jitter>>4); /* jitter */
254
255 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
256 {
257 put_be32(pb, 0); /* last SR timestamp */
258 put_be32(pb, 0); /* delay since last SR */
259 } else {
260 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
261 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
262
263 put_be32(pb, middle_32_bits); /* last SR timestamp */
264 put_be32(pb, delay_since_last); /* delay since last SR */
265 }
266
267 // CNAME
268 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
7f3468d3 269 put_byte(pb, RTCP_SDES);
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270 len = strlen(s->hostname);
271 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
272 put_be32(pb, s->ssrc);
273 put_byte(pb, 0x01);
274 put_byte(pb, len);
275 put_buffer(pb, s->hostname, len);
276 // padding
277 for (len = (6 + len) % 4; len % 4; len++) {
278 put_byte(pb, 0);
279 }
280
281 put_flush_packet(pb);
282 len = url_close_dyn_buf(pb, &buf);
283 if ((len > 0) && buf) {
284 int result;
e8420626 285 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 286 result= url_write(s->rtp_ctx, buf, len);
e8420626 287 dprintf(s->ic, "result from url_write: %d\n", result);
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288 av_free(buf);
289 }
290 return 0;
291}
292
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MS
293void rtp_send_punch_packets(URLContext* rtp_handle)
294{
295 ByteIOContext *pb;
296 uint8_t *buf;
297 int len;
298
299 /* Send a small RTP packet */
300 if (url_open_dyn_buf(&pb) < 0)
301 return;
302
303 put_byte(pb, (RTP_VERSION << 6));
304 put_byte(pb, 0); /* Payload type */
305 put_be16(pb, 0); /* Seq */
306 put_be32(pb, 0); /* Timestamp */
307 put_be32(pb, 0); /* SSRC */
308
309 put_flush_packet(pb);
310 len = url_close_dyn_buf(pb, &buf);
311 if ((len > 0) && buf)
312 url_write(rtp_handle, buf, len);
313 av_free(buf);
314
315 /* Send a minimal RTCP RR */
316 if (url_open_dyn_buf(&pb) < 0)
317 return;
318
319 put_byte(pb, (RTP_VERSION << 6));
7f3468d3 320 put_byte(pb, RTCP_RR); /* receiver report */
9c8fa20d
MS
321 put_be16(pb, 1); /* length in words - 1 */
322 put_be32(pb, 0); /* our own SSRC */
323
324 put_flush_packet(pb);
325 len = url_close_dyn_buf(pb, &buf);
326 if ((len > 0) && buf)
327 url_write(rtp_handle, buf, len);
328 av_free(buf);
329}
330
331
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332/**
333 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
334 * MPEG2TS streams to indicate that they should be demuxed inside the
335 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 336 */
ca937a55 337RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type)
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338{
339 RTPDemuxContext *s;
340
341 s = av_mallocz(sizeof(RTPDemuxContext));
342 if (!s)
343 return NULL;
344 s->payload_type = payload_type;
345 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 346 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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347 s->ic = s1;
348 s->st = st;
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349 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
350 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 351 s->ts = ff_mpegts_parse_open(s->ic);
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352 if (s->ts == NULL) {
353 av_free(s);
354 return NULL;
355 }
356 } else {
26efefc5 357 av_set_pts_info(st, 32, 1, 90000);
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358 switch(st->codec->codec_id) {
359 case CODEC_ID_MPEG1VIDEO:
360 case CODEC_ID_MPEG2VIDEO:
361 case CODEC_ID_MP2:
362 case CODEC_ID_MP3:
363 case CODEC_ID_MPEG4:
45aa9080 364 case CODEC_ID_H263:
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365 case CODEC_ID_H264:
366 st->need_parsing = AVSTREAM_PARSE_FULL;
367 break;
368 default:
72415b2a 369 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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LA
370 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
371 }
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372 break;
373 }
374 }
375 // needed to send back RTCP RR in RTSP sessions
376 s->rtp_ctx = rtpc;
377 gethostname(s->hostname, sizeof(s->hostname));
378 return s;
379}
380
99a1d191
RB
381void
382rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
383 RTPDynamicProtocolHandler *handler)
384{
385 s->dynamic_protocol_context = ctx;
386 s->parse_packet = handler->parse_packet;
387}
388
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389/**
390 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
391 */
392static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
393{
d74c6145 394 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
fba7815d
LA
395 int64_t addend;
396 int delta_timestamp;
397
398 /* compute pts from timestamp with received ntp_time */
399 delta_timestamp = timestamp - s->last_rtcp_timestamp;
400 /* convert to the PTS timebase */
2cab6b48 401 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 402 pkt->pts = s->range_start_offset + addend + delta_timestamp;
fba7815d 403 }
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404}
405
406/**
407 * Parse an RTP or RTCP packet directly sent as a buffer.
408 * @param s RTP parse context.
409 * @param pkt returned packet
410 * @param buf input buffer or NULL to read the next packets
411 * @param len buffer len
412 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
413 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
414 */
415int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
416 const uint8_t *buf, int len)
417{
418 unsigned int ssrc, h;
f841a0fc 419 int payload_type, seq, ret, flags = 0;
8eb793c4
LA
420 AVStream *st;
421 uint32_t timestamp;
422 int rv= 0;
423
424 if (!buf) {
425 /* return the next packets, if any */
426 if(s->st && s->parse_packet) {
d74c6145
MS
427 /* timestamp should be overwritten by parse_packet, if not,
428 * the packet is left with pts == AV_NOPTS_VALUE */
429 timestamp = RTP_NOTS_VALUE;
1a45a9f4 430 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 431 s->st, pkt, &timestamp, NULL, 0, flags);
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432 finalize_packet(s, pkt, timestamp);
433 return rv;
434 } else {
435 // TODO: Move to a dynamic packet handler (like above)
436 if (s->read_buf_index >= s->read_buf_size)
437 return -1;
9125806e 438 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
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439 s->read_buf_size - s->read_buf_index);
440 if (ret < 0)
441 return -1;
442 s->read_buf_index += ret;
443 if (s->read_buf_index < s->read_buf_size)
444 return 1;
445 else
446 return 0;
447 }
448 }
449
450 if (len < 12)
451 return -1;
452
453 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
454 return -1;
7f3468d3 455 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
b20359f5 456 return rtcp_parse_packet(s, buf, len);
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LA
457 }
458 payload_type = buf[1] & 0x7f;
144ae29d
RB
459 if (buf[1] & 0x80)
460 flags |= RTP_FLAG_MARKER;
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LA
461 seq = AV_RB16(buf + 2);
462 timestamp = AV_RB32(buf + 4);
463 ssrc = AV_RB32(buf + 8);
464 /* store the ssrc in the RTPDemuxContext */
465 s->ssrc = ssrc;
466
467 /* NOTE: we can handle only one payload type */
468 if (s->payload_type != payload_type)
469 return -1;
470
471 st = s->st;
472 // only do something with this if all the rtp checks pass...
473 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
474 {
475 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
476 payload_type, seq, ((s->seq + 1) & 0xffff));
477 return -1;
478 }
479
480 s->seq = seq;
481 len -= 12;
482 buf += 12;
483
484 if (!st) {
485 /* specific MPEG2TS demux support */
9125806e 486 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
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LA
487 if (ret < 0)
488 return -1;
489 if (ret < len) {
490 s->read_buf_size = len - ret;
491 memcpy(s->buf, buf + ret, s->read_buf_size);
492 s->read_buf_index = 0;
493 return 1;
494 }
f3e71942 495 return 0;
b4e3330c 496 } else if (s->parse_packet) {
1a45a9f4 497 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 498 s->st, pkt, &timestamp, buf, len, flags);
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LA
499 } else {
500 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
501 switch(st->codec->codec_id) {
502 case CODEC_ID_MP2:
76faff6e 503 case CODEC_ID_MP3:
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LA
504 /* better than nothing: skip mpeg audio RTP header */
505 if (len <= 4)
506 return -1;
507 h = AV_RB32(buf);
508 len -= 4;
509 buf += 4;
510 av_new_packet(pkt, len);
511 memcpy(pkt->data, buf, len);
512 break;
513 case CODEC_ID_MPEG1VIDEO:
514 case CODEC_ID_MPEG2VIDEO:
515 /* better than nothing: skip mpeg video RTP header */
516 if (len <= 4)
517 return -1;
518 h = AV_RB32(buf);
519 buf += 4;
520 len -= 4;
521 if (h & (1 << 26)) {
522 /* mpeg2 */
523 if (len <= 4)
524 return -1;
525 buf += 4;
526 len -= 4;
527 }
528 av_new_packet(pkt, len);
529 memcpy(pkt->data, buf, len);
530 break;
8eb793c4 531 default:
f739b36d
RB
532 av_new_packet(pkt, len);
533 memcpy(pkt->data, buf, len);
8eb793c4
LA
534 break;
535 }
eafb17d1
RB
536
537 pkt->stream_index = st->index;
f3e71942 538 }
8eb793c4 539
95f03cf3
RB
540 // now perform timestamp things....
541 finalize_packet(s, pkt, timestamp);
f3e71942 542
8eb793c4
LA
543 return rv;
544}
545
546void rtp_parse_close(RTPDemuxContext *s)
547{
8eb793c4 548 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 549 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
550 }
551 av_free(s);
552}
016bc031
JA
553
554int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
555 int (*parse_fmtp)(AVStream *stream,
556 PayloadContext *data,
557 char *attr, char *value))
558{
559 char attr[256];
824535e3 560 char *value;
016bc031 561 int res;
824535e3
JA
562 int value_size = strlen(p) + 1;
563
564 if (!(value = av_malloc(value_size))) {
565 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
566 return AVERROR(ENOMEM);
567 }
016bc031
JA
568
569 // remove protocol identifier
570 while (*p && *p == ' ') p++; // strip spaces
571 while (*p && *p != ' ') p++; // eat protocol identifier
572 while (*p && *p == ' ') p++; // strip trailing spaces
573
574 while (ff_rtsp_next_attr_and_value(&p,
575 attr, sizeof(attr),
824535e3 576 value, value_size)) {
016bc031
JA
577
578 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
579 if (res < 0 && res != AVERROR_PATCHWELCOME) {
580 av_free(value);
016bc031 581 return res;
824535e3 582 }
016bc031 583 }
824535e3 584 av_free(value);
016bc031
JA
585 return 0;
586}