Rename the "enc" variable, which refers to the AVCodecContext of a
[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"
e9fce261 33#include "rtp_asf.h"
8eb793c4 34#include "rtp_h264.h"
e6327fba 35#include "rtp_vorbis.h"
556aa7a1 36#include "rtpdec_amr.h"
45aa9080 37#include "rtpdec_h263.h"
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38
39//#define DEBUG
40
41/* TODO: - add RTCP statistics reporting (should be optional).
42
43 - add support for h263/mpeg4 packetized output : IDEA: send a
44 buffer to 'rtp_write_packet' contains all the packets for ONE
45 frame. Each packet should have a four byte header containing
46 the length in big endian format (same trick as
47 'url_open_dyn_packet_buf')
48*/
49
50/* statistics functions */
51RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
52
53static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
54static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
55
0369d2b0 56void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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57{
58 handler->next= RTPFirstDynamicPayloadHandler;
59 RTPFirstDynamicPayloadHandler= handler;
60}
61
62void av_register_rtp_dynamic_payload_handlers(void)
63{
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64 ff_register_dynamic_payload_handler(&mp4v_es_handler);
65 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
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66 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
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68 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 70 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 71 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
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72
73 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
74 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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75}
76
77static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
78{
79 if (buf[1] != 200)
80 return -1;
81 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
82 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
83 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
84 s->last_rtcp_timestamp = AV_RB32(buf + 16);
85 return 0;
86}
87
88#define RTP_SEQ_MOD (1<<16)
89
90/**
91* called on parse open packet
92*/
93static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
94{
95 memset(s, 0, sizeof(RTPStatistics));
96 s->max_seq= base_sequence;
97 s->probation= 1;
98}
99
100/**
101* called whenever there is a large jump in sequence numbers, or when they get out of probation...
102*/
103static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
104{
105 s->max_seq= seq;
106 s->cycles= 0;
107 s->base_seq= seq -1;
108 s->bad_seq= RTP_SEQ_MOD + 1;
109 s->received= 0;
110 s->expected_prior= 0;
111 s->received_prior= 0;
112 s->jitter= 0;
113 s->transit= 0;
114}
115
116/**
117* returns 1 if we should handle this packet.
118*/
119static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
120{
121 uint16_t udelta= seq - s->max_seq;
122 const int MAX_DROPOUT= 3000;
123 const int MAX_MISORDER = 100;
124 const int MIN_SEQUENTIAL = 2;
125
126 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
127 if(s->probation)
128 {
129 if(seq==s->max_seq + 1) {
130 s->probation--;
131 s->max_seq= seq;
132 if(s->probation==0) {
133 rtp_init_sequence(s, seq);
134 s->received++;
135 return 1;
136 }
137 } else {
138 s->probation= MIN_SEQUENTIAL - 1;
139 s->max_seq = seq;
140 }
141 } else if (udelta < MAX_DROPOUT) {
142 // in order, with permissible gap
143 if(seq < s->max_seq) {
144 //sequence number wrapped; count antother 64k cycles
145 s->cycles += RTP_SEQ_MOD;
146 }
147 s->max_seq= seq;
148 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
149 // sequence made a large jump...
150 if(seq==s->bad_seq) {
151 // two sequential packets-- assume that the other side restarted without telling us; just resync.
152 rtp_init_sequence(s, seq);
153 } else {
154 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
155 return 0;
156 }
157 } else {
158 // duplicate or reordered packet...
159 }
160 s->received++;
161 return 1;
162}
163
164#if 0
165/**
166* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
167* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
168* never change. I left this in in case someone else can see a way. (rdm)
169*/
170static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
171{
172 uint32_t transit= arrival_timestamp - sent_timestamp;
173 int d;
174 s->transit= transit;
175 d= FFABS(transit - s->transit);
176 s->jitter += d - ((s->jitter + 8)>>4);
177}
178#endif
179
180int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
181{
182 ByteIOContext *pb;
183 uint8_t *buf;
184 int len;
185 int rtcp_bytes;
186 RTPStatistics *stats= &s->statistics;
187 uint32_t lost;
188 uint32_t extended_max;
189 uint32_t expected_interval;
190 uint32_t received_interval;
191 uint32_t lost_interval;
192 uint32_t expected;
193 uint32_t fraction;
194 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
195
196 if (!s->rtp_ctx || (count < 1))
197 return -1;
198
199 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
200 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
201 s->octet_count += count;
202 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
203 RTCP_TX_RATIO_DEN;
204 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
205 if (rtcp_bytes < 28)
206 return -1;
207 s->last_octet_count = s->octet_count;
208
209 if (url_open_dyn_buf(&pb) < 0)
210 return -1;
211
212 // Receiver Report
213 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
214 put_byte(pb, 201);
215 put_be16(pb, 7); /* length in words - 1 */
216 put_be32(pb, s->ssrc); // our own SSRC
217 put_be32(pb, s->ssrc); // XXX: should be the server's here!
218 // some placeholders we should really fill...
219 // RFC 1889/p64
220 extended_max= stats->cycles + stats->max_seq;
221 expected= extended_max - stats->base_seq + 1;
222 lost= expected - stats->received;
223 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
224 expected_interval= expected - stats->expected_prior;
225 stats->expected_prior= expected;
226 received_interval= stats->received - stats->received_prior;
227 stats->received_prior= stats->received;
228 lost_interval= expected_interval - received_interval;
229 if (expected_interval==0 || lost_interval<=0) fraction= 0;
230 else fraction = (lost_interval<<8)/expected_interval;
231
232 fraction= (fraction<<24) | lost;
233
234 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
235 put_be32(pb, extended_max); /* max sequence received */
236 put_be32(pb, stats->jitter>>4); /* jitter */
237
238 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
239 {
240 put_be32(pb, 0); /* last SR timestamp */
241 put_be32(pb, 0); /* delay since last SR */
242 } else {
243 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
244 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
245
246 put_be32(pb, middle_32_bits); /* last SR timestamp */
247 put_be32(pb, delay_since_last); /* delay since last SR */
248 }
249
250 // CNAME
251 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
252 put_byte(pb, 202);
253 len = strlen(s->hostname);
254 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
255 put_be32(pb, s->ssrc);
256 put_byte(pb, 0x01);
257 put_byte(pb, len);
258 put_buffer(pb, s->hostname, len);
259 // padding
260 for (len = (6 + len) % 4; len % 4; len++) {
261 put_byte(pb, 0);
262 }
263
264 put_flush_packet(pb);
265 len = url_close_dyn_buf(pb, &buf);
266 if ((len > 0) && buf) {
267 int result;
e8420626 268 dprintf(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 269 result= url_write(s->rtp_ctx, buf, len);
e8420626 270 dprintf(s->ic, "result from url_write: %d\n", result);
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271 av_free(buf);
272 }
273 return 0;
274}
275
276/**
277 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
278 * MPEG2TS streams to indicate that they should be demuxed inside the
279 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
280 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
281 */
be73a544 282RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
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283{
284 RTPDemuxContext *s;
285
286 s = av_mallocz(sizeof(RTPDemuxContext));
287 if (!s)
288 return NULL;
289 s->payload_type = payload_type;
290 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
291 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
292 s->ic = s1;
293 s->st = st;
294 s->rtp_payload_data = rtp_payload_data;
295 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
296 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 297 s->ts = ff_mpegts_parse_open(s->ic);
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298 if (s->ts == NULL) {
299 av_free(s);
300 return NULL;
301 }
302 } else {
26efefc5 303 av_set_pts_info(st, 32, 1, 90000);
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304 switch(st->codec->codec_id) {
305 case CODEC_ID_MPEG1VIDEO:
306 case CODEC_ID_MPEG2VIDEO:
307 case CODEC_ID_MP2:
308 case CODEC_ID_MP3:
309 case CODEC_ID_MPEG4:
45aa9080 310 case CODEC_ID_H263:
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311 case CODEC_ID_H264:
312 st->need_parsing = AVSTREAM_PARSE_FULL;
313 break;
314 default:
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315 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
316 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
317 }
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318 break;
319 }
320 }
321 // needed to send back RTCP RR in RTSP sessions
322 s->rtp_ctx = rtpc;
323 gethostname(s->hostname, sizeof(s->hostname));
324 return s;
325}
326
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327void
328rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
329 RTPDynamicProtocolHandler *handler)
330{
331 s->dynamic_protocol_context = ctx;
332 s->parse_packet = handler->parse_packet;
333}
334
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335static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
336{
337 int au_headers_length, au_header_size, i;
338 GetBitContext getbitcontext;
be73a544 339 RTPPayloadData *infos;
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340
341 infos = s->rtp_payload_data;
342
343 if (infos == NULL)
344 return -1;
345
bd107136 346 /* decode the first 2 bytes where the AUHeader sections are stored
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347 length in bits */
348 au_headers_length = AV_RB16(buf);
349
350 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
351 return -1;
352
353 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
354
355 /* skip AU headers length section (2 bytes) */
356 buf += 2;
357
358 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
359
360 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
361 au_header_size = infos->sizelength + infos->indexlength;
362 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
363 return -1;
364
365 infos->nb_au_headers = au_headers_length / au_header_size;
366 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
367
368 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
369 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
370 but does when sending the whole as one big packet... */
371 infos->au_headers[0].size = 0;
372 infos->au_headers[0].index = 0;
373 for (i = 0; i < infos->nb_au_headers; ++i) {
374 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
375 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
376 }
377
378 infos->nb_au_headers = 1;
379
380 return 0;
381}
382
383/**
384 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
385 */
386static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
387{
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388 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
389 int64_t addend;
390 int delta_timestamp;
391
392 /* compute pts from timestamp with received ntp_time */
393 delta_timestamp = timestamp - s->last_rtcp_timestamp;
394 /* convert to the PTS timebase */
395 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);
396 pkt->pts = addend + delta_timestamp;
397 }
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398}
399
400/**
401 * Parse an RTP or RTCP packet directly sent as a buffer.
402 * @param s RTP parse context.
403 * @param pkt returned packet
404 * @param buf input buffer or NULL to read the next packets
405 * @param len buffer len
406 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
407 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
408 */
409int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
410 const uint8_t *buf, int len)
411{
412 unsigned int ssrc, h;
f841a0fc 413 int payload_type, seq, ret, flags = 0;
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414 AVStream *st;
415 uint32_t timestamp;
416 int rv= 0;
417
418 if (!buf) {
419 /* return the next packets, if any */
420 if(s->st && s->parse_packet) {
421 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
1a45a9f4 422 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 423 s->st, pkt, &timestamp, NULL, 0, flags);
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424 finalize_packet(s, pkt, timestamp);
425 return rv;
426 } else {
427 // TODO: Move to a dynamic packet handler (like above)
428 if (s->read_buf_index >= s->read_buf_size)
429 return -1;
9125806e 430 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
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431 s->read_buf_size - s->read_buf_index);
432 if (ret < 0)
433 return -1;
434 s->read_buf_index += ret;
435 if (s->read_buf_index < s->read_buf_size)
436 return 1;
437 else
438 return 0;
439 }
440 }
441
442 if (len < 12)
443 return -1;
444
445 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
446 return -1;
447 if (buf[1] >= 200 && buf[1] <= 204) {
448 rtcp_parse_packet(s, buf, len);
449 return -1;
450 }
451 payload_type = buf[1] & 0x7f;
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452 if (buf[1] & 0x80)
453 flags |= RTP_FLAG_MARKER;
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454 seq = AV_RB16(buf + 2);
455 timestamp = AV_RB32(buf + 4);
456 ssrc = AV_RB32(buf + 8);
457 /* store the ssrc in the RTPDemuxContext */
458 s->ssrc = ssrc;
459
460 /* NOTE: we can handle only one payload type */
461 if (s->payload_type != payload_type)
462 return -1;
463
464 st = s->st;
465 // only do something with this if all the rtp checks pass...
466 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
467 {
468 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
469 payload_type, seq, ((s->seq + 1) & 0xffff));
470 return -1;
471 }
472
473 s->seq = seq;
474 len -= 12;
475 buf += 12;
476
477 if (!st) {
478 /* specific MPEG2TS demux support */
9125806e 479 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
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480 if (ret < 0)
481 return -1;
482 if (ret < len) {
483 s->read_buf_size = len - ret;
484 memcpy(s->buf, buf + ret, s->read_buf_size);
485 s->read_buf_index = 0;
486 return 1;
487 }
f3e71942 488 return 0;
b4e3330c 489 } else if (s->parse_packet) {
1a45a9f4 490 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 491 s->st, pkt, &timestamp, buf, len, flags);
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492 } else {
493 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
494 switch(st->codec->codec_id) {
495 case CODEC_ID_MP2:
76faff6e 496 case CODEC_ID_MP3:
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497 /* better than nothing: skip mpeg audio RTP header */
498 if (len <= 4)
499 return -1;
500 h = AV_RB32(buf);
501 len -= 4;
502 buf += 4;
503 av_new_packet(pkt, len);
504 memcpy(pkt->data, buf, len);
505 break;
506 case CODEC_ID_MPEG1VIDEO:
507 case CODEC_ID_MPEG2VIDEO:
508 /* better than nothing: skip mpeg video RTP header */
509 if (len <= 4)
510 return -1;
511 h = AV_RB32(buf);
512 buf += 4;
513 len -= 4;
514 if (h & (1 << 26)) {
515 /* mpeg2 */
516 if (len <= 4)
517 return -1;
518 buf += 4;
519 len -= 4;
520 }
521 av_new_packet(pkt, len);
522 memcpy(pkt->data, buf, len);
523 break;
524 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
525 // timestamps.
526 // TODO: Put this into a dynamic packet handler...
527 case CODEC_ID_AAC:
528 if (rtp_parse_mp4_au(s, buf))
529 return -1;
530 {
be73a544 531 RTPPayloadData *infos = s->rtp_payload_data;
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532 if (infos == NULL)
533 return -1;
534 buf += infos->au_headers_length_bytes + 2;
535 len -= infos->au_headers_length_bytes + 2;
536
537 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
538 one au_header */
539 av_new_packet(pkt, infos->au_headers[0].size);
540 memcpy(pkt->data, buf, infos->au_headers[0].size);
541 buf += infos->au_headers[0].size;
542 len -= infos->au_headers[0].size;
543 }
544 s->read_buf_size = len;
545 rv= 0;
546 break;
547 default:
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548 av_new_packet(pkt, len);
549 memcpy(pkt->data, buf, len);
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550 break;
551 }
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552
553 pkt->stream_index = st->index;
f3e71942 554 }
8eb793c4 555
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556 // now perform timestamp things....
557 finalize_packet(s, pkt, timestamp);
f3e71942 558
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559 return rv;
560}
561
562void rtp_parse_close(RTPDemuxContext *s)
563{
564 // TODO: fold this into the protocol specific data fields.
565 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 566 ff_mpegts_parse_close(s->ts);
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567 }
568 av_free(s);
569}