FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavformat/rdt.c
Date: 2024-02-16 17:37:06
Exec Total Coverage
Lines: 0 243 0.0%
Functions: 0 16 0.0%
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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 "rtpdec.h"
31 #include "rdt.h"
32 #include "libavutil/base64.h"
33 #include "libavutil/md5.h"
34 #include "rm.h"
35 #include "internal.h"
36 #include "avio_internal.h"
37 #include "libavcodec/get_bits.h"
38
39 struct RDTDemuxContext {
40 AVFormatContext *ic; /**< the containing (RTSP) demux context */
41 /** Each RDT stream-set (represented by one RTSPStream) can contain
42 * multiple streams (of the same content, but with possibly different
43 * codecs/bitrates). Each such stream is represented by one AVStream
44 * in the AVFormatContext, and this variable points to the offset in
45 * that array such that the first is the first stream of this set. */
46 AVStream **streams;
47 int n_streams; /**< streams with identical content in this set */
48 void *dynamic_protocol_context;
49 DynamicPayloadPacketHandlerProc parse_packet;
50 uint32_t prev_timestamp;
51 int prev_set_id, prev_stream_id;
52 };
53
54 RDTDemuxContext *
55 ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,
56 void *priv_data, const RTPDynamicProtocolHandler *handler)
57 {
58 RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
59 if (!s)
60 return NULL;
61
62 s->ic = ic;
63 s->streams = &ic->streams[first_stream_of_set_idx];
64 do {
65 s->n_streams++;
66 } while (first_stream_of_set_idx + s->n_streams < ic->nb_streams &&
67 s->streams[s->n_streams]->id == s->streams[0]->id);
68 s->prev_set_id = -1;
69 s->prev_stream_id = -1;
70 s->prev_timestamp = -1;
71 s->parse_packet = handler ? handler->parse_packet : NULL;
72 s->dynamic_protocol_context = priv_data;
73
74 return s;
75 }
76
77 void
78 ff_rdt_parse_close(RDTDemuxContext *s)
79 {
80 av_free(s);
81 }
82
83 struct PayloadContext {
84 AVFormatContext *rmctx;
85 int nb_rmst;
86 RMStream **rmst;
87 uint8_t *mlti_data;
88 unsigned int mlti_data_size;
89 char buffer[RTP_MAX_PACKET_LENGTH + AV_INPUT_BUFFER_PADDING_SIZE];
90 int audio_pkt_cnt; /**< remaining audio packets in rmdec */
91 };
92
93 void
94 ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
95 const char *challenge)
96 {
97 int ch_len = strlen (challenge), i;
98 unsigned char zres[16],
99 buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 };
100 #define XOR_TABLE_SIZE 37
101 static const unsigned char xor_table[XOR_TABLE_SIZE] = {
102 0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53,
103 0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
104 0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
105 0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02,
106 0x10, 0x57, 0x05, 0x18, 0x54 };
107
108 /* some (length) checks */
109 if (ch_len == 40) /* what a hack... */
110 ch_len = 32;
111 else if (ch_len > 56)
112 ch_len = 56;
113 memcpy(buf + 8, challenge, ch_len);
114
115 /* xor challenge bytewise with xor_table */
116 for (i = 0; i < XOR_TABLE_SIZE; i++)
117 buf[8 + i] ^= xor_table[i];
118
119 av_md5_sum(zres, buf, 64);
120 ff_data_to_hex(response, zres, 16, 1);
121
122 /* add tail */
123 strcpy (response + 32, "01d0a8e3");
124
125 /* calculate checksum */
126 for (i = 0; i < 8; i++)
127 chksum[i] = response[i * 4];
128 chksum[8] = 0;
129 }
130
131 static int
132 rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
133 {
134 FFIOContext pb0;
135 AVIOContext *const pb = &pb0.pub;
136 unsigned int size;
137 uint32_t tag;
138
139 /**
140 * Layout of the MLTI chunk:
141 * 4: MLTI
142 * 2: number of streams
143 * Then for each stream ([number_of_streams] times):
144 * 2: mdpr index
145 * 2: number of mdpr chunks
146 * Then for each mdpr chunk ([number_of_mdpr_chunks] times):
147 * 4: size
148 * [size]: data
149 * we skip MDPR chunks until we reach the one of the stream
150 * we're interested in, and forward that ([size]+[data]) to
151 * the RM demuxer to parse the stream-specific header data.
152 */
153 if (!rdt->mlti_data)
154 return -1;
155 ffio_init_read_context(&pb0, rdt->mlti_data, rdt->mlti_data_size);
156 tag = avio_rl32(pb);
157 if (tag == MKTAG('M', 'L', 'T', 'I')) {
158 int num, chunk_nr;
159
160 /* read index of MDPR chunk numbers */
161 num = avio_rb16(pb);
162 if (rule_nr < 0 || rule_nr >= num)
163 return -1;
164 avio_skip(pb, rule_nr * 2);
165 chunk_nr = avio_rb16(pb);
166 avio_skip(pb, (num - 1 - rule_nr) * 2);
167
168 /* read MDPR chunks */
169 num = avio_rb16(pb);
170 if (chunk_nr >= num)
171 return -1;
172 while (chunk_nr--)
173 avio_skip(pb, avio_rb32(pb));
174 size = avio_rb32(pb);
175 } else {
176 size = rdt->mlti_data_size;
177 avio_seek(pb, 0, SEEK_SET);
178 }
179 if (ff_rm_read_mdpr_codecdata(rdt->rmctx, pb, st, rdt->rmst[st->index], size, NULL) < 0)
180 return -1;
181
182 return 0;
183 }
184
185 /**
186 * Actual data handling.
187 */
188
189 int
190 ff_rdt_parse_header(const uint8_t *buf, int len,
191 int *pset_id, int *pseq_no, int *pstream_id,
192 int *pis_keyframe, uint32_t *ptimestamp)
193 {
194 GetBitContext gb;
195 int consumed = 0, set_id, seq_no, stream_id, is_keyframe,
196 len_included, need_reliable;
197 uint32_t timestamp;
198
199 /* skip status packets */
200 while (len >= 5 && buf[1] == 0xFF /* status packet */) {
201 int pkt_len;
202
203 if (!(buf[0] & 0x80))
204 return -1; /* not followed by a data packet */
205
206 pkt_len = AV_RB16(buf+3);
207 buf += pkt_len;
208 len -= pkt_len;
209 consumed += pkt_len;
210 }
211 if (len < 16)
212 return -1;
213 /**
214 * Layout of the header (in bits):
215 * 1: len_included
216 * Flag indicating whether this header includes a length field;
217 * this can be used to concatenate multiple RDT packets in a
218 * single UDP/TCP data frame and is used to precede RDT data
219 * by stream status packets
220 * 1: need_reliable
221 * Flag indicating whether this header includes a "reliable
222 * sequence number"; these are apparently sequence numbers of
223 * data packets alone. For data packets, this flag is always
224 * set, according to the Real documentation [1]
225 * 5: set_id
226 * ID of a set of streams of identical content, possibly with
227 * different codecs or bitrates
228 * 1: is_reliable
229 * Flag set for certain streams deemed less tolerable for packet
230 * loss
231 * 16: seq_no
232 * Packet sequence number; if >=0xFF00, this is a non-data packet
233 * containing stream status info, the second byte indicates the
234 * type of status packet (see wireshark docs / source code [2])
235 * if (len_included) {
236 * 16: packet_len
237 * } else {
238 * packet_len = remainder of UDP/TCP frame
239 * }
240 * 1: is_back_to_back
241 * Back-to-Back flag; used for timing, set for one in every 10
242 * packets, according to the Real documentation [1]
243 * 1: is_slow_data
244 * Slow-data flag; currently unused, according to Real docs [1]
245 * 5: stream_id
246 * ID of the stream within this particular set of streams
247 * 1: is_no_keyframe
248 * Non-keyframe flag (unset if packet belongs to a keyframe)
249 * 32: timestamp (PTS)
250 * if (set_id == 0x1F) {
251 * 16: set_id (extended set-of-streams ID; see set_id)
252 * }
253 * if (need_reliable) {
254 * 16: reliable_seq_no
255 * Reliable sequence number (see need_reliable)
256 * }
257 * if (stream_id == 0x3F) {
258 * 16: stream_id (extended stream ID; see stream_id)
259 * }
260 * [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt
261 * [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
262 * http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
263 */
264 init_get_bits(&gb, buf, len << 3);
265 len_included = get_bits1(&gb);
266 need_reliable = get_bits1(&gb);
267 set_id = get_bits(&gb, 5);
268 skip_bits(&gb, 1);
269 seq_no = get_bits(&gb, 16);
270 if (len_included)
271 skip_bits(&gb, 16);
272 skip_bits(&gb, 2);
273 stream_id = get_bits(&gb, 5);
274 is_keyframe = !get_bits1(&gb);
275 timestamp = get_bits_long(&gb, 32);
276 if (set_id == 0x1f)
277 set_id = get_bits(&gb, 16);
278 if (need_reliable)
279 skip_bits(&gb, 16);
280 if (stream_id == 0x1f)
281 stream_id = get_bits(&gb, 16);
282
283 if (pset_id) *pset_id = set_id;
284 if (pseq_no) *pseq_no = seq_no;
285 if (pstream_id) *pstream_id = stream_id;
286 if (pis_keyframe) *pis_keyframe = is_keyframe;
287 if (ptimestamp) *ptimestamp = timestamp;
288
289 return consumed + (get_bits_count(&gb) >> 3);
290 }
291
292 /**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
293 static int
294 rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st,
295 AVPacket *pkt, uint32_t *timestamp,
296 const uint8_t *buf, int len, uint16_t rtp_seq, int flags)
297 {
298 int seq = 1, res;
299
300 if (rdt->audio_pkt_cnt == 0) {
301 FFIOContext pb;
302 int pos, rmflags;
303
304 ffio_init_read_context(&pb, buf, len);
305 rmflags = (flags & RTP_FLAG_KEY) ? 2 : 0;
306 res = ff_rm_parse_packet(rdt->rmctx, &pb.pub, st, rdt->rmst[st->index],
307 len, pkt, &seq, rmflags, *timestamp);
308 pos = avio_tell(&pb.pub);
309 if (res < 0)
310 return res;
311 if (res > 0) {
312 if (st->codecpar->codec_id == AV_CODEC_ID_AAC) {
313 memcpy (rdt->buffer, buf + pos, len - pos);
314 rdt->rmctx->pb = avio_alloc_context (rdt->buffer, len - pos, 0,
315 NULL, NULL, NULL, NULL);
316 }
317 goto get_cache;
318 }
319 } else {
320 get_cache:
321 rdt->audio_pkt_cnt =
322 ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb,
323 st, rdt->rmst[st->index], pkt);
324 if (rdt->audio_pkt_cnt == 0 &&
325 st->codecpar->codec_id == AV_CODEC_ID_AAC)
326 avio_context_free(&rdt->rmctx->pb);
327 }
328 pkt->stream_index = st->index;
329 pkt->pts = *timestamp;
330
331 return rdt->audio_pkt_cnt > 0;
332 }
333
334 int
335 ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
336 uint8_t **bufptr, int len)
337 {
338 uint8_t *buf = bufptr ? *bufptr : NULL;
339 int seq_no, flags = 0, stream_id, set_id, is_keyframe;
340 uint32_t timestamp;
341 int rv= 0;
342
343 if (!s->parse_packet)
344 return -1;
345
346 if (!buf && s->prev_stream_id != -1) {
347 /* return the next packets, if any */
348 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
349 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
350 s->streams[s->prev_stream_id],
351 pkt, &timestamp, NULL, 0, 0, flags);
352 return rv;
353 }
354
355 if (len < 12)
356 return -1;
357 rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, &timestamp);
358 if (rv < 0)
359 return rv;
360 if (is_keyframe &&
361 (set_id != s->prev_set_id || timestamp != s->prev_timestamp ||
362 stream_id != s->prev_stream_id)) {
363 flags |= RTP_FLAG_KEY;
364 s->prev_set_id = set_id;
365 s->prev_timestamp = timestamp;
366 }
367 s->prev_stream_id = stream_id;
368 buf += rv;
369 len -= rv;
370
371 if (s->prev_stream_id >= s->n_streams) {
372 s->prev_stream_id = -1;
373 return -1;
374 }
375
376 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
377 s->streams[s->prev_stream_id],
378 pkt, &timestamp, buf, len, 0, flags);
379
380 return rv;
381 }
382
383 void
384 ff_rdt_subscribe_rule (char *cmd, int size,
385 int stream_nr, int rule_nr)
386 {
387 av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d",
388 stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
389 }
390
391 static unsigned char *
392 rdt_parse_b64buf (unsigned int *target_len, const char *p)
393 {
394 unsigned char *target;
395 int len = strlen(p);
396 if (*p == '\"') {
397 p++;
398 len -= 2; /* skip embracing " at start/end */
399 }
400 *target_len = len * 3 / 4;
401 target = av_mallocz(*target_len + AV_INPUT_BUFFER_PADDING_SIZE);
402 if (!target)
403 return NULL;
404 av_base64_decode(target, p, *target_len);
405 return target;
406 }
407
408 static int
409 rdt_parse_sdp_line (AVFormatContext *s, int st_index,
410 PayloadContext *rdt, const char *line)
411 {
412 AVStream *stream = s->streams[st_index];
413 const char *p = line;
414
415 if (av_strstart(p, "OpaqueData:buffer;", &p)) {
416 rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
417 } else if (av_strstart(p, "StartTime:integer;", &p))
418 ffstream(stream)->first_dts = atoi(p);
419 else if (av_strstart(p, "ASMRuleBook:string;", &p)) {
420 int n, first = -1;
421
422 for (n = 0; n < s->nb_streams; n++)
423 if (s->streams[n]->id == stream->id) {
424 int count = s->streams[n]->index + 1, err;
425 if (first == -1) first = n;
426 if (rdt->nb_rmst < count) {
427 if ((err = av_reallocp(&rdt->rmst,
428 count * sizeof(*rdt->rmst))) < 0) {
429 rdt->nb_rmst = 0;
430 return err;
431 }
432 memset(rdt->rmst + rdt->nb_rmst, 0,
433 (count - rdt->nb_rmst) * sizeof(*rdt->rmst));
434 rdt->nb_rmst = count;
435 }
436 rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream();
437 if (!rdt->rmst[s->streams[n]->index])
438 return AVERROR(ENOMEM);
439 rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2);
440 }
441 }
442
443 return 0;
444 }
445
446 static void
447 real_parse_asm_rule(AVStream *st, const char *p, const char *end)
448 {
449 do {
450 /* can be either averagebandwidth= or AverageBandwidth= */
451 if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%"SCNd64, &st->codecpar->bit_rate) == 1)
452 break;
453 if (!(p = strchr(p, ',')) || p > end)
454 p = end;
455 p++;
456 } while (p < end);
457 }
458
459 static AVStream *
460 add_dstream(AVFormatContext *s, AVStream *orig_st)
461 {
462 AVStream *st;
463
464 if (!(st = avformat_new_stream(s, NULL)))
465 return NULL;
466 st->id = orig_st->id;
467 st->codecpar->codec_type = orig_st->codecpar->codec_type;
468 ffstream(st)->first_dts = ffstream(orig_st)->first_dts;
469
470 return st;
471 }
472
473 static void
474 real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
475 const char *p)
476 {
477 const char *end;
478 int n_rules = 0, odd = 0;
479 AVStream *st;
480
481 /**
482 * The ASMRuleBook contains a list of comma-separated strings per rule,
483 * and each rule is separated by a ;. The last one also has a ; at the
484 * end so we can use it as delimiter.
485 * Every rule occurs twice, once for when the RTSP packet header marker
486 * is set and once for if it isn't. We only read the first because we
487 * don't care much (that's what the "odd" variable is for).
488 * Each rule contains a set of one or more statements, optionally
489 * preceded by a single condition. If there's a condition, the rule
490 * starts with a '#'. Multiple conditions are merged between brackets,
491 * so there are never multiple conditions spread out over separate
492 * statements. Generally, these conditions are bitrate limits (min/max)
493 * for multi-bitrate streams.
494 */
495 if (*p == '\"') p++;
496 while (1) {
497 if (!(end = strchr(p, ';')))
498 break;
499 if (!odd && end != p) {
500 if (n_rules > 0)
501 st = add_dstream(s, orig_st);
502 else
503 st = orig_st;
504 if (!st)
505 break;
506 real_parse_asm_rule(st, p, end);
507 n_rules++;
508 }
509 p = end + 1;
510 odd ^= 1;
511 }
512 }
513
514 void
515 ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index,
516 const char *line)
517 {
518 const char *p = line;
519
520 if (av_strstart(p, "ASMRuleBook:string;", &p))
521 real_parse_asm_rulebook(s, s->streams[stream_index], p);
522 }
523
524
525
526 static av_cold int rdt_init(AVFormatContext *s, int st_index, PayloadContext *rdt)
527 {
528 int ret;
529
530 rdt->rmctx = avformat_alloc_context();
531 if (!rdt->rmctx)
532 return AVERROR(ENOMEM);
533
534 if ((ret = ff_copy_whiteblacklists(rdt->rmctx, s)) < 0)
535 return ret;
536
537 return avformat_open_input(&rdt->rmctx, "", &ff_rdt_demuxer, NULL);
538 }
539
540 static void
541 rdt_close_context (PayloadContext *rdt)
542 {
543 int i;
544
545 for (i = 0; i < rdt->nb_rmst; i++)
546 if (rdt->rmst[i]) {
547 ff_rm_free_rmstream(rdt->rmst[i]);
548 av_freep(&rdt->rmst[i]);
549 }
550 if (rdt->rmctx)
551 avformat_close_input(&rdt->rmctx);
552 av_freep(&rdt->mlti_data);
553 av_freep(&rdt->rmst);
554 }
555
556 #define RDT_HANDLER(n, s, t) \
557 const RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
558 .enc_name = s, \
559 .codec_type = t, \
560 .codec_id = AV_CODEC_ID_NONE, \
561 .priv_data_size = sizeof(PayloadContext), \
562 .init = rdt_init, \
563 .parse_sdp_a_line = rdt_parse_sdp_line, \
564 .close = rdt_close_context, \
565 .parse_packet = rdt_parse_packet \
566 }
567
568 RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
569 RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
570 RDT_HANDLER(video, "x-pn-realvideo", AVMEDIA_TYPE_VIDEO);
571 RDT_HANDLER(audio, "x-pn-realaudio", AVMEDIA_TYPE_AUDIO);
572
573