FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavformat/rdt.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 0 245 0.0%
Functions: 0 16 0.0%
Branches: 0 148 0.0%

Line Branch Exec Source
1 /*
2 * Realmedia RTSP protocol (RDT) support.
3 * Copyright (c) 2007 Ronald S. Bultje
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 */
21
22 /**
23 * @file
24 * @brief Realmedia RTSP protocol (RDT) support
25 * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
26 */
27
28 #include "avformat.h"
29 #include "libavutil/avstring.h"
30 #include "libavutil/mem.h"
31 #include "demux.h"
32 #include "rtpdec.h"
33 #include "rdt.h"
34 #include "libavutil/base64.h"
35 #include "libavutil/md5.h"
36 #include "rm.h"
37 #include "internal.h"
38 #include "avio_internal.h"
39 #include "libavcodec/get_bits.h"
40
41 struct RDTDemuxContext {
42 AVFormatContext *ic; /**< the containing (RTSP) demux context */
43 /** Each RDT stream-set (represented by one RTSPStream) can contain
44 * multiple streams (of the same content, but with possibly different
45 * codecs/bitrates). Each such stream is represented by one AVStream
46 * in the AVFormatContext, and this variable points to the offset in
47 * that array such that the first is the first stream of this set. */
48 AVStream **streams;
49 int n_streams; /**< streams with identical content in this set */
50 void *dynamic_protocol_context;
51 DynamicPayloadPacketHandlerProc parse_packet;
52 uint32_t prev_timestamp;
53 int prev_set_id, prev_stream_id;
54 };
55
56 RDTDemuxContext *
57 ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,
58 void *priv_data, const RTPDynamicProtocolHandler *handler)
59 {
60 RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
61 if (!s)
62 return NULL;
63
64 s->ic = ic;
65 s->streams = &ic->streams[first_stream_of_set_idx];
66 do {
67 s->n_streams++;
68 } while (first_stream_of_set_idx + s->n_streams < ic->nb_streams &&
69 s->streams[s->n_streams]->id == s->streams[0]->id);
70 s->prev_set_id = -1;
71 s->prev_stream_id = -1;
72 s->prev_timestamp = -1;
73 s->parse_packet = handler ? handler->parse_packet : NULL;
74 s->dynamic_protocol_context = priv_data;
75
76 return s;
77 }
78
79 void
80 ff_rdt_parse_close(RDTDemuxContext *s)
81 {
82 av_free(s);
83 }
84
85 struct PayloadContext {
86 AVFormatContext *rmctx;
87 int nb_rmst;
88 RMStream **rmst;
89 uint8_t *mlti_data;
90 unsigned int mlti_data_size;
91 char buffer[RTP_MAX_PACKET_LENGTH + AV_INPUT_BUFFER_PADDING_SIZE];
92 int audio_pkt_cnt; /**< remaining audio packets in rmdec */
93 };
94
95 void
96 ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
97 const char *challenge)
98 {
99 int ch_len = strlen (challenge), i;
100 unsigned char zres[16],
101 buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 };
102 #define XOR_TABLE_SIZE 37
103 static const unsigned char xor_table[XOR_TABLE_SIZE] = {
104 0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53,
105 0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
106 0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
107 0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02,
108 0x10, 0x57, 0x05, 0x18, 0x54 };
109
110 /* some (length) checks */
111 if (ch_len == 40) /* what a hack... */
112 ch_len = 32;
113 else if (ch_len > 56)
114 ch_len = 56;
115 memcpy(buf + 8, challenge, ch_len);
116
117 /* xor challenge bytewise with xor_table */
118 for (i = 0; i < XOR_TABLE_SIZE; i++)
119 buf[8 + i] ^= xor_table[i];
120
121 av_md5_sum(zres, buf, 64);
122 ff_data_to_hex(response, zres, 16, 1);
123
124 /* add tail */
125 strcpy (response + 32, "01d0a8e3");
126
127 /* calculate checksum */
128 for (i = 0; i < 8; i++)
129 chksum[i] = response[i * 4];
130 chksum[8] = 0;
131 }
132
133 static int
134 rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
135 {
136 FFIOContext pb0;
137 AVIOContext *const pb = &pb0.pub;
138 unsigned int size;
139 uint32_t tag;
140
141 /**
142 * Layout of the MLTI chunk:
143 * 4: MLTI
144 * 2: number of streams
145 * Then for each stream ([number_of_streams] times):
146 * 2: mdpr index
147 * 2: number of mdpr chunks
148 * Then for each mdpr chunk ([number_of_mdpr_chunks] times):
149 * 4: size
150 * [size]: data
151 * we skip MDPR chunks until we reach the one of the stream
152 * we're interested in, and forward that ([size]+[data]) to
153 * the RM demuxer to parse the stream-specific header data.
154 */
155 if (!rdt->mlti_data)
156 return -1;
157 ffio_init_read_context(&pb0, rdt->mlti_data, rdt->mlti_data_size);
158 tag = avio_rl32(pb);
159 if (tag == MKTAG('M', 'L', 'T', 'I')) {
160 int num, chunk_nr;
161
162 /* read index of MDPR chunk numbers */
163 num = avio_rb16(pb);
164 if (rule_nr < 0 || rule_nr >= num)
165 return -1;
166 avio_skip(pb, rule_nr * 2);
167 chunk_nr = avio_rb16(pb);
168 avio_skip(pb, (num - 1 - rule_nr) * 2);
169
170 /* read MDPR chunks */
171 num = avio_rb16(pb);
172 if (chunk_nr >= num)
173 return -1;
174 while (chunk_nr--)
175 avio_skip(pb, avio_rb32(pb));
176 size = avio_rb32(pb);
177 } else {
178 size = rdt->mlti_data_size;
179 avio_seek(pb, 0, SEEK_SET);
180 }
181 if (ff_rm_read_mdpr_codecdata(rdt->rmctx, pb, st, rdt->rmst[st->index], size, NULL) < 0)
182 return -1;
183
184 return 0;
185 }
186
187 /**
188 * Actual data handling.
189 */
190
191 int
192 ff_rdt_parse_header(const uint8_t *buf, int len,
193 int *pset_id, int *pseq_no, int *pstream_id,
194 int *pis_keyframe, uint32_t *ptimestamp)
195 {
196 GetBitContext gb;
197 int consumed = 0, set_id, seq_no, stream_id, is_keyframe,
198 len_included, need_reliable;
199 uint32_t timestamp;
200
201 /* skip status packets */
202 while (len >= 5 && buf[1] == 0xFF /* status packet */) {
203 int pkt_len;
204
205 if (!(buf[0] & 0x80))
206 return -1; /* not followed by a data packet */
207
208 pkt_len = AV_RB16(buf+3);
209 if (pkt_len > len)
210 return AVERROR_INVALIDDATA;
211 buf += pkt_len;
212 len -= pkt_len;
213 consumed += pkt_len;
214 }
215 if (len < 16)
216 return -1;
217 /**
218 * Layout of the header (in bits):
219 * 1: len_included
220 * Flag indicating whether this header includes a length field;
221 * this can be used to concatenate multiple RDT packets in a
222 * single UDP/TCP data frame and is used to precede RDT data
223 * by stream status packets
224 * 1: need_reliable
225 * Flag indicating whether this header includes a "reliable
226 * sequence number"; these are apparently sequence numbers of
227 * data packets alone. For data packets, this flag is always
228 * set, according to the Real documentation [1]
229 * 5: set_id
230 * ID of a set of streams of identical content, possibly with
231 * different codecs or bitrates
232 * 1: is_reliable
233 * Flag set for certain streams deemed less tolerable for packet
234 * loss
235 * 16: seq_no
236 * Packet sequence number; if >=0xFF00, this is a non-data packet
237 * containing stream status info, the second byte indicates the
238 * type of status packet (see wireshark docs / source code [2])
239 * if (len_included) {
240 * 16: packet_len
241 * } else {
242 * packet_len = remainder of UDP/TCP frame
243 * }
244 * 1: is_back_to_back
245 * Back-to-Back flag; used for timing, set for one in every 10
246 * packets, according to the Real documentation [1]
247 * 1: is_slow_data
248 * Slow-data flag; currently unused, according to Real docs [1]
249 * 5: stream_id
250 * ID of the stream within this particular set of streams
251 * 1: is_no_keyframe
252 * Non-keyframe flag (unset if packet belongs to a keyframe)
253 * 32: timestamp (PTS)
254 * if (set_id == 0x1F) {
255 * 16: set_id (extended set-of-streams ID; see set_id)
256 * }
257 * if (need_reliable) {
258 * 16: reliable_seq_no
259 * Reliable sequence number (see need_reliable)
260 * }
261 * if (stream_id == 0x3F) {
262 * 16: stream_id (extended stream ID; see stream_id)
263 * }
264 * [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt
265 * [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
266 * http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
267 */
268 init_get_bits(&gb, buf, len << 3);
269 len_included = get_bits1(&gb);
270 need_reliable = get_bits1(&gb);
271 set_id = get_bits(&gb, 5);
272 skip_bits(&gb, 1);
273 seq_no = get_bits(&gb, 16);
274 if (len_included)
275 skip_bits(&gb, 16);
276 skip_bits(&gb, 2);
277 stream_id = get_bits(&gb, 5);
278 is_keyframe = !get_bits1(&gb);
279 timestamp = get_bits_long(&gb, 32);
280 if (set_id == 0x1f)
281 set_id = get_bits(&gb, 16);
282 if (need_reliable)
283 skip_bits(&gb, 16);
284 if (stream_id == 0x1f)
285 stream_id = get_bits(&gb, 16);
286
287 if (pset_id) *pset_id = set_id;
288 if (pseq_no) *pseq_no = seq_no;
289 if (pstream_id) *pstream_id = stream_id;
290 if (pis_keyframe) *pis_keyframe = is_keyframe;
291 if (ptimestamp) *ptimestamp = timestamp;
292
293 return consumed + (get_bits_count(&gb) >> 3);
294 }
295
296 /**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
297 static int
298 rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st,
299 AVPacket *pkt, uint32_t *timestamp,
300 const uint8_t *buf, int len, uint16_t rtp_seq, int flags)
301 {
302 int seq = 1, res;
303
304 if (rdt->audio_pkt_cnt == 0) {
305 FFIOContext pb;
306 int pos, rmflags;
307
308 ffio_init_read_context(&pb, buf, len);
309 rmflags = (flags & RTP_FLAG_KEY) ? 2 : 0;
310 res = ff_rm_parse_packet(rdt->rmctx, &pb.pub, st, rdt->rmst[st->index],
311 len, pkt, &seq, rmflags, *timestamp);
312 pos = avio_tell(&pb.pub);
313 if (res < 0)
314 return res;
315 if (res > 0) {
316 if (st->codecpar->codec_id == AV_CODEC_ID_AAC) {
317 memcpy (rdt->buffer, buf + pos, len - pos);
318 rdt->rmctx->pb = avio_alloc_context (rdt->buffer, len - pos, 0,
319 NULL, NULL, NULL, NULL);
320 }
321 goto get_cache;
322 }
323 } else {
324 get_cache:
325 rdt->audio_pkt_cnt =
326 ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb,
327 st, rdt->rmst[st->index], pkt);
328 if (rdt->audio_pkt_cnt == 0 &&
329 st->codecpar->codec_id == AV_CODEC_ID_AAC)
330 avio_context_free(&rdt->rmctx->pb);
331 }
332 pkt->stream_index = st->index;
333 pkt->pts = *timestamp;
334
335 return rdt->audio_pkt_cnt > 0;
336 }
337
338 int
339 ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
340 uint8_t **bufptr, int len)
341 {
342 uint8_t *buf = bufptr ? *bufptr : NULL;
343 int seq_no, flags = 0, stream_id, set_id, is_keyframe;
344 uint32_t timestamp;
345 int rv= 0;
346
347 if (!s->parse_packet)
348 return -1;
349
350 if (!buf && s->prev_stream_id != -1) {
351 /* return the next packets, if any */
352 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
353 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
354 s->streams[s->prev_stream_id],
355 pkt, &timestamp, NULL, 0, 0, flags);
356 return rv;
357 }
358
359 if (len < 12)
360 return -1;
361 rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, &timestamp);
362 if (rv < 0)
363 return rv;
364 if (is_keyframe &&
365 (set_id != s->prev_set_id || timestamp != s->prev_timestamp ||
366 stream_id != s->prev_stream_id)) {
367 flags |= RTP_FLAG_KEY;
368 s->prev_set_id = set_id;
369 s->prev_timestamp = timestamp;
370 }
371 s->prev_stream_id = stream_id;
372 buf += rv;
373 len -= rv;
374
375 if (s->prev_stream_id >= s->n_streams) {
376 s->prev_stream_id = -1;
377 return -1;
378 }
379
380 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
381 s->streams[s->prev_stream_id],
382 pkt, &timestamp, buf, len, 0, flags);
383
384 return rv;
385 }
386
387 void
388 ff_rdt_subscribe_rule (char *cmd, int size,
389 int stream_nr, int rule_nr)
390 {
391 av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d",
392 stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
393 }
394
395 static unsigned char *
396 rdt_parse_b64buf (unsigned int *target_len, const char *p)
397 {
398 unsigned char *target;
399 int len = strlen(p);
400 if (*p == '\"') {
401 p++;
402 len -= 2; /* skip embracing " at start/end */
403 }
404 *target_len = len * 3 / 4;
405 target = av_mallocz(*target_len + AV_INPUT_BUFFER_PADDING_SIZE);
406 if (!target)
407 return NULL;
408 av_base64_decode(target, p, *target_len);
409 return target;
410 }
411
412 static int
413 rdt_parse_sdp_line (AVFormatContext *s, int st_index,
414 PayloadContext *rdt, const char *line)
415 {
416 AVStream *stream = s->streams[st_index];
417 const char *p = line;
418
419 if (av_strstart(p, "OpaqueData:buffer;", &p)) {
420 rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
421 } else if (av_strstart(p, "StartTime:integer;", &p))
422 ffstream(stream)->first_dts = atoi(p);
423 else if (av_strstart(p, "ASMRuleBook:string;", &p)) {
424 int n, first = -1;
425
426 for (n = 0; n < s->nb_streams; n++)
427 if (s->streams[n]->id == stream->id) {
428 int count = s->streams[n]->index + 1, err;
429 if (first == -1) first = n;
430 if (rdt->nb_rmst < count) {
431 if ((err = av_reallocp(&rdt->rmst,
432 count * sizeof(*rdt->rmst))) < 0) {
433 rdt->nb_rmst = 0;
434 return err;
435 }
436 memset(rdt->rmst + rdt->nb_rmst, 0,
437 (count - rdt->nb_rmst) * sizeof(*rdt->rmst));
438 rdt->nb_rmst = count;
439 }
440 rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream();
441 if (!rdt->rmst[s->streams[n]->index])
442 return AVERROR(ENOMEM);
443 rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2);
444 }
445 }
446
447 return 0;
448 }
449
450 static void
451 real_parse_asm_rule(AVStream *st, const char *p, const char *end)
452 {
453 do {
454 /* can be either averagebandwidth= or AverageBandwidth= */
455 if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%"SCNd64, &st->codecpar->bit_rate) == 1)
456 break;
457 if (!(p = strchr(p, ',')) || p > end)
458 p = end;
459 p++;
460 } while (p < end);
461 }
462
463 static AVStream *
464 add_dstream(AVFormatContext *s, AVStream *orig_st)
465 {
466 AVStream *st;
467
468 if (!(st = avformat_new_stream(s, NULL)))
469 return NULL;
470 st->id = orig_st->id;
471 st->codecpar->codec_type = orig_st->codecpar->codec_type;
472 ffstream(st)->first_dts = ffstream(orig_st)->first_dts;
473
474 return st;
475 }
476
477 static void
478 real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
479 const char *p)
480 {
481 const char *end;
482 int n_rules = 0, odd = 0;
483 AVStream *st;
484
485 /**
486 * The ASMRuleBook contains a list of comma-separated strings per rule,
487 * and each rule is separated by a ;. The last one also has a ; at the
488 * end so we can use it as delimiter.
489 * Every rule occurs twice, once for when the RTSP packet header marker
490 * is set and once for if it isn't. We only read the first because we
491 * don't care much (that's what the "odd" variable is for).
492 * Each rule contains a set of one or more statements, optionally
493 * preceded by a single condition. If there's a condition, the rule
494 * starts with a '#'. Multiple conditions are merged between brackets,
495 * so there are never multiple conditions spread out over separate
496 * statements. Generally, these conditions are bitrate limits (min/max)
497 * for multi-bitrate streams.
498 */
499 if (*p == '\"') p++;
500 while (1) {
501 if (!(end = strchr(p, ';')))
502 break;
503 if (!odd && end != p) {
504 if (n_rules > 0)
505 st = add_dstream(s, orig_st);
506 else
507 st = orig_st;
508 if (!st)
509 break;
510 real_parse_asm_rule(st, p, end);
511 n_rules++;
512 }
513 p = end + 1;
514 odd ^= 1;
515 }
516 }
517
518 void
519 ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index,
520 const char *line)
521 {
522 const char *p = line;
523
524 if (av_strstart(p, "ASMRuleBook:string;", &p))
525 real_parse_asm_rulebook(s, s->streams[stream_index], p);
526 }
527
528
529
530 static av_cold int rdt_init(AVFormatContext *s, int st_index, PayloadContext *rdt)
531 {
532 int ret;
533
534 rdt->rmctx = avformat_alloc_context();
535 if (!rdt->rmctx)
536 return AVERROR(ENOMEM);
537
538 if ((ret = ff_copy_whiteblacklists(rdt->rmctx, s)) < 0)
539 return ret;
540
541 return avformat_open_input(&rdt->rmctx, "", &ff_rdt_demuxer.p, NULL);
542 }
543
544 static void
545 rdt_close_context (PayloadContext *rdt)
546 {
547 int i;
548
549 for (i = 0; i < rdt->nb_rmst; i++)
550 if (rdt->rmst[i]) {
551 ff_rm_free_rmstream(rdt->rmst[i]);
552 av_freep(&rdt->rmst[i]);
553 }
554 if (rdt->rmctx)
555 avformat_close_input(&rdt->rmctx);
556 av_freep(&rdt->mlti_data);
557 av_freep(&rdt->rmst);
558 }
559
560 #define RDT_HANDLER(n, s, t) \
561 const RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
562 .enc_name = s, \
563 .codec_type = t, \
564 .codec_id = AV_CODEC_ID_NONE, \
565 .priv_data_size = sizeof(PayloadContext), \
566 .init = rdt_init, \
567 .parse_sdp_a_line = rdt_parse_sdp_line, \
568 .close = rdt_close_context, \
569 .parse_packet = rdt_parse_packet \
570 }
571
572 RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
573 RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
574 RDT_HANDLER(video, "x-pn-realvideo", AVMEDIA_TYPE_VIDEO);
575 RDT_HANDLER(audio, "x-pn-realaudio", AVMEDIA_TYPE_AUDIO);
576
577