Rename RTP payload contexts to PayloadContext, suggested by Luca in
[libav.git] / libavformat / rtpdec.c
CommitLineData
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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 */
245976da 21
7246177d 22/* needed for gethostname() */
d0feff2a 23#define _XOPEN_SOURCE 600
7246177d 24
245976da 25#include "libavcodec/bitstream.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
32#include "rtp_internal.h"
33#include "rtp_h264.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 'url_open_dyn_packet_buf')
44*/
45
46/* statistics functions */
47RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
48
49static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
50static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
51
0369d2b0 52void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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53{
54 handler->next= RTPFirstDynamicPayloadHandler;
55 RTPFirstDynamicPayloadHandler= handler;
56}
57
58void av_register_rtp_dynamic_payload_handlers(void)
59{
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60 ff_register_dynamic_payload_handler(&mp4v_es_handler);
61 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
62 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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63}
64
65static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
66{
67 if (buf[1] != 200)
68 return -1;
69 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
70 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
71 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
72 s->last_rtcp_timestamp = AV_RB32(buf + 16);
73 return 0;
74}
75
76#define RTP_SEQ_MOD (1<<16)
77
78/**
79* called on parse open packet
80*/
81static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
82{
83 memset(s, 0, sizeof(RTPStatistics));
84 s->max_seq= base_sequence;
85 s->probation= 1;
86}
87
88/**
89* called whenever there is a large jump in sequence numbers, or when they get out of probation...
90*/
91static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
92{
93 s->max_seq= seq;
94 s->cycles= 0;
95 s->base_seq= seq -1;
96 s->bad_seq= RTP_SEQ_MOD + 1;
97 s->received= 0;
98 s->expected_prior= 0;
99 s->received_prior= 0;
100 s->jitter= 0;
101 s->transit= 0;
102}
103
104/**
105* returns 1 if we should handle this packet.
106*/
107static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
108{
109 uint16_t udelta= seq - s->max_seq;
110 const int MAX_DROPOUT= 3000;
111 const int MAX_MISORDER = 100;
112 const int MIN_SEQUENTIAL = 2;
113
114 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
115 if(s->probation)
116 {
117 if(seq==s->max_seq + 1) {
118 s->probation--;
119 s->max_seq= seq;
120 if(s->probation==0) {
121 rtp_init_sequence(s, seq);
122 s->received++;
123 return 1;
124 }
125 } else {
126 s->probation= MIN_SEQUENTIAL - 1;
127 s->max_seq = seq;
128 }
129 } else if (udelta < MAX_DROPOUT) {
130 // in order, with permissible gap
131 if(seq < s->max_seq) {
132 //sequence number wrapped; count antother 64k cycles
133 s->cycles += RTP_SEQ_MOD;
134 }
135 s->max_seq= seq;
136 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
137 // sequence made a large jump...
138 if(seq==s->bad_seq) {
139 // two sequential packets-- assume that the other side restarted without telling us; just resync.
140 rtp_init_sequence(s, seq);
141 } else {
142 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
143 return 0;
144 }
145 } else {
146 // duplicate or reordered packet...
147 }
148 s->received++;
149 return 1;
150}
151
152#if 0
153/**
154* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
155* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
156* never change. I left this in in case someone else can see a way. (rdm)
157*/
158static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
159{
160 uint32_t transit= arrival_timestamp - sent_timestamp;
161 int d;
162 s->transit= transit;
163 d= FFABS(transit - s->transit);
164 s->jitter += d - ((s->jitter + 8)>>4);
165}
166#endif
167
168int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
169{
170 ByteIOContext *pb;
171 uint8_t *buf;
172 int len;
173 int rtcp_bytes;
174 RTPStatistics *stats= &s->statistics;
175 uint32_t lost;
176 uint32_t extended_max;
177 uint32_t expected_interval;
178 uint32_t received_interval;
179 uint32_t lost_interval;
180 uint32_t expected;
181 uint32_t fraction;
182 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
183
184 if (!s->rtp_ctx || (count < 1))
185 return -1;
186
187 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
188 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
189 s->octet_count += count;
190 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
191 RTCP_TX_RATIO_DEN;
192 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
193 if (rtcp_bytes < 28)
194 return -1;
195 s->last_octet_count = s->octet_count;
196
197 if (url_open_dyn_buf(&pb) < 0)
198 return -1;
199
200 // Receiver Report
201 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
202 put_byte(pb, 201);
203 put_be16(pb, 7); /* length in words - 1 */
204 put_be32(pb, s->ssrc); // our own SSRC
205 put_be32(pb, s->ssrc); // XXX: should be the server's here!
206 // some placeholders we should really fill...
207 // RFC 1889/p64
208 extended_max= stats->cycles + stats->max_seq;
209 expected= extended_max - stats->base_seq + 1;
210 lost= expected - stats->received;
211 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
212 expected_interval= expected - stats->expected_prior;
213 stats->expected_prior= expected;
214 received_interval= stats->received - stats->received_prior;
215 stats->received_prior= stats->received;
216 lost_interval= expected_interval - received_interval;
217 if (expected_interval==0 || lost_interval<=0) fraction= 0;
218 else fraction = (lost_interval<<8)/expected_interval;
219
220 fraction= (fraction<<24) | lost;
221
222 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
223 put_be32(pb, extended_max); /* max sequence received */
224 put_be32(pb, stats->jitter>>4); /* jitter */
225
226 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
227 {
228 put_be32(pb, 0); /* last SR timestamp */
229 put_be32(pb, 0); /* delay since last SR */
230 } else {
231 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
232 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
233
234 put_be32(pb, middle_32_bits); /* last SR timestamp */
235 put_be32(pb, delay_since_last); /* delay since last SR */
236 }
237
238 // CNAME
239 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
240 put_byte(pb, 202);
241 len = strlen(s->hostname);
242 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
243 put_be32(pb, s->ssrc);
244 put_byte(pb, 0x01);
245 put_byte(pb, len);
246 put_buffer(pb, s->hostname, len);
247 // padding
248 for (len = (6 + len) % 4; len % 4; len++) {
249 put_byte(pb, 0);
250 }
251
252 put_flush_packet(pb);
253 len = url_close_dyn_buf(pb, &buf);
254 if ((len > 0) && buf) {
255 int result;
e8420626 256 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 257 result= url_write(s->rtp_ctx, buf, len);
e8420626 258 dprintf(s->ic, "result from url_write: %d\n", result);
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259 av_free(buf);
260 }
261 return 0;
262}
263
264/**
265 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
266 * MPEG2TS streams to indicate that they should be demuxed inside the
267 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
268 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
269 */
270RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
271{
272 RTPDemuxContext *s;
273
274 s = av_mallocz(sizeof(RTPDemuxContext));
275 if (!s)
276 return NULL;
277 s->payload_type = payload_type;
278 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
279 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
280 s->ic = s1;
281 s->st = st;
282 s->rtp_payload_data = rtp_payload_data;
283 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
284 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
285 s->ts = mpegts_parse_open(s->ic);
286 if (s->ts == NULL) {
287 av_free(s);
288 return NULL;
289 }
290 } else {
26efefc5 291 av_set_pts_info(st, 32, 1, 90000);
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292 switch(st->codec->codec_id) {
293 case CODEC_ID_MPEG1VIDEO:
294 case CODEC_ID_MPEG2VIDEO:
295 case CODEC_ID_MP2:
296 case CODEC_ID_MP3:
297 case CODEC_ID_MPEG4:
298 case CODEC_ID_H264:
299 st->need_parsing = AVSTREAM_PARSE_FULL;
300 break;
301 default:
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302 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
303 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
304 }
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305 break;
306 }
307 }
308 // needed to send back RTCP RR in RTSP sessions
309 s->rtp_ctx = rtpc;
310 gethostname(s->hostname, sizeof(s->hostname));
311 return s;
312}
313
314static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
315{
316 int au_headers_length, au_header_size, i;
317 GetBitContext getbitcontext;
318 rtp_payload_data_t *infos;
319
320 infos = s->rtp_payload_data;
321
322 if (infos == NULL)
323 return -1;
324
bd107136 325 /* decode the first 2 bytes where the AUHeader sections are stored
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326 length in bits */
327 au_headers_length = AV_RB16(buf);
328
329 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
330 return -1;
331
332 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
333
334 /* skip AU headers length section (2 bytes) */
335 buf += 2;
336
337 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
338
339 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
340 au_header_size = infos->sizelength + infos->indexlength;
341 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
342 return -1;
343
344 infos->nb_au_headers = au_headers_length / au_header_size;
345 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
346
347 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
348 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
349 but does when sending the whole as one big packet... */
350 infos->au_headers[0].size = 0;
351 infos->au_headers[0].index = 0;
352 for (i = 0; i < infos->nb_au_headers; ++i) {
353 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
354 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
355 }
356
357 infos->nb_au_headers = 1;
358
359 return 0;
360}
361
362/**
363 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
364 */
365static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
366{
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367 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
368 int64_t addend;
369 int delta_timestamp;
370
371 /* compute pts from timestamp with received ntp_time */
372 delta_timestamp = timestamp - s->last_rtcp_timestamp;
373 /* convert to the PTS timebase */
374 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);
375 pkt->pts = addend + delta_timestamp;
376 }
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377 pkt->stream_index = s->st->index;
378}
379
380/**
381 * Parse an RTP or RTCP packet directly sent as a buffer.
382 * @param s RTP parse context.
383 * @param pkt returned packet
384 * @param buf input buffer or NULL to read the next packets
385 * @param len buffer len
386 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
387 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
388 */
389int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
390 const uint8_t *buf, int len)
391{
392 unsigned int ssrc, h;
f841a0fc 393 int payload_type, seq, ret, flags = 0;
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394 AVStream *st;
395 uint32_t timestamp;
396 int rv= 0;
397
398 if (!buf) {
399 /* return the next packets, if any */
400 if(s->st && s->parse_packet) {
401 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
f841a0fc 402 rv= s->parse_packet(s, pkt, &timestamp, NULL, 0, flags);
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403 finalize_packet(s, pkt, timestamp);
404 return rv;
405 } else {
406 // TODO: Move to a dynamic packet handler (like above)
407 if (s->read_buf_index >= s->read_buf_size)
408 return -1;
409 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
410 s->read_buf_size - s->read_buf_index);
411 if (ret < 0)
412 return -1;
413 s->read_buf_index += ret;
414 if (s->read_buf_index < s->read_buf_size)
415 return 1;
416 else
417 return 0;
418 }
419 }
420
421 if (len < 12)
422 return -1;
423
424 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
425 return -1;
426 if (buf[1] >= 200 && buf[1] <= 204) {
427 rtcp_parse_packet(s, buf, len);
428 return -1;
429 }
430 payload_type = buf[1] & 0x7f;
431 seq = AV_RB16(buf + 2);
432 timestamp = AV_RB32(buf + 4);
433 ssrc = AV_RB32(buf + 8);
434 /* store the ssrc in the RTPDemuxContext */
435 s->ssrc = ssrc;
436
437 /* NOTE: we can handle only one payload type */
438 if (s->payload_type != payload_type)
439 return -1;
440
441 st = s->st;
442 // only do something with this if all the rtp checks pass...
443 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
444 {
445 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
446 payload_type, seq, ((s->seq + 1) & 0xffff));
447 return -1;
448 }
449
450 s->seq = seq;
451 len -= 12;
452 buf += 12;
453
454 if (!st) {
455 /* specific MPEG2TS demux support */
456 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
457 if (ret < 0)
458 return -1;
459 if (ret < len) {
460 s->read_buf_size = len - ret;
461 memcpy(s->buf, buf + ret, s->read_buf_size);
462 s->read_buf_index = 0;
463 return 1;
464 }
b4e3330c 465 } else if (s->parse_packet) {
f841a0fc 466 rv = s->parse_packet(s, pkt, &timestamp, buf, len, flags);
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467 } else {
468 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
469 switch(st->codec->codec_id) {
470 case CODEC_ID_MP2:
471 /* better than nothing: skip mpeg audio RTP header */
472 if (len <= 4)
473 return -1;
474 h = AV_RB32(buf);
475 len -= 4;
476 buf += 4;
477 av_new_packet(pkt, len);
478 memcpy(pkt->data, buf, len);
479 break;
480 case CODEC_ID_MPEG1VIDEO:
481 case CODEC_ID_MPEG2VIDEO:
482 /* better than nothing: skip mpeg video RTP header */
483 if (len <= 4)
484 return -1;
485 h = AV_RB32(buf);
486 buf += 4;
487 len -= 4;
488 if (h & (1 << 26)) {
489 /* mpeg2 */
490 if (len <= 4)
491 return -1;
492 buf += 4;
493 len -= 4;
494 }
495 av_new_packet(pkt, len);
496 memcpy(pkt->data, buf, len);
497 break;
498 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
499 // timestamps.
500 // TODO: Put this into a dynamic packet handler...
501 case CODEC_ID_AAC:
502 if (rtp_parse_mp4_au(s, buf))
503 return -1;
504 {
505 rtp_payload_data_t *infos = s->rtp_payload_data;
506 if (infos == NULL)
507 return -1;
508 buf += infos->au_headers_length_bytes + 2;
509 len -= infos->au_headers_length_bytes + 2;
510
511 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
512 one au_header */
513 av_new_packet(pkt, infos->au_headers[0].size);
514 memcpy(pkt->data, buf, infos->au_headers[0].size);
515 buf += infos->au_headers[0].size;
516 len -= infos->au_headers[0].size;
517 }
518 s->read_buf_size = len;
519 rv= 0;
520 break;
521 default:
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522 av_new_packet(pkt, len);
523 memcpy(pkt->data, buf, len);
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524 break;
525 }
526
527 // now perform timestamp things....
528 finalize_packet(s, pkt, timestamp);
529 }
530 return rv;
531}
532
533void rtp_parse_close(RTPDemuxContext *s)
534{
535 // TODO: fold this into the protocol specific data fields.
536 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
537 mpegts_parse_close(s->ts);
538 }
539 av_free(s);
540}