FFmpeg: libavformat/rdt.c Source File

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rdt.c

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1 /*

2 * Realmedia RTSP protocol (RDT) support.

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 */

22 /**

23 * @file

24 * @brief Realmedia RTSP protocol (RDT) support

25 * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>

26 */

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"

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 identifical 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 };

54 RDTDemuxContext *

55 ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,

56 void *priv_data, RTPDynamicProtocolHandler *handler)

57 {

58 RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));

59 if (!s)

60 return NULL;

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;

74 return s;

75 }

77 void

78 ff_rdt_parse_close(RDTDemuxContext *s)

79 {

80 av_free(s);

81 }

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 + FF_INPUT_BUFFER_PADDING_SIZE];

90 int audio_pkt_cnt; /**< remaining audio packets in rmdec */

91 };

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 AVIOContext pb;

135 unsigned int size;

136 uint32_t tag;

137

138 /**

139 * Layout of the MLTI chunk:

140 * 4: MLTI

141 * 2: number of streams

142 * Then for each stream ([number_of_streams] times):

143 * 2: mdpr index

144 * 2: number of mdpr chunks

145 * Then for each mdpr chunk ([number_of_mdpr_chunks] times):

146 * 4: size

147 * [size]: data

148 * we skip MDPR chunks until we reach the one of the stream

149 * we're interested in, and forward that ([size]+[data]) to

150 * the RM demuxer to parse the stream-specific header data.

151 */

152 if (!rdt->mlti_data)

153 return -1;

154 ffio_init_context(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,

155 NULL, NULL, NULL, NULL);

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 AVIOContext pb;

300

301 if (rdt->audio_pkt_cnt == 0) {

302 int pos, rmflags;

303

304 ffio_init_context(&pb, (uint8_t *)buf, len, 0, NULL, NULL, NULL, NULL);

305 rmflags = (flags & RTP_FLAG_KEY) ? 2 : 0;

306 res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt,

307 &seq, rmflags, *timestamp);

308 pos = avio_tell(&pb);

309 if (res < 0)

310 return res;

311 if (res > 0) {

312 if (st->codec->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->codec->codec_id == AV_CODEC_ID_AAC)

326 av_freep(&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 + FF_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 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=%d", &st->codec->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->codec->codec_type = orig_st->codec->codec_type;

468 st->first_dts = 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_whitelists(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 static RTPDynamicProtocolHandler 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 void ff_register_rdt_dynamic_payload_handlers(void)

574 {

575 ff_register_dynamic_payload_handler(&rdt_video_handler);

576 ff_register_dynamic_payload_handler(&rdt_audio_handler);

577 ff_register_dynamic_payload_handler(&rdt_live_video_handler);

578 ff_register_dynamic_payload_handler(&rdt_live_audio_handler);

579 }

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