PlaylistFetcher.cpp revision 5cf91c5067a9c7ed3c138d4e56fb176b28f5dc3a
1/* 2 * Copyright (C) 2012 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17//#define LOG_NDEBUG 0 18#define LOG_TAG "PlaylistFetcher" 19#include <utils/Log.h> 20 21#include "PlaylistFetcher.h" 22 23#include "LiveDataSource.h" 24#include "LiveSession.h" 25#include "M3UParser.h" 26 27#include "include/avc_utils.h" 28#include "include/HTTPBase.h" 29#include "include/ID3.h" 30#include "mpeg2ts/AnotherPacketSource.h" 31 32#include <media/IStreamSource.h> 33#include <media/stagefright/foundation/ABitReader.h> 34#include <media/stagefright/foundation/ABuffer.h> 35#include <media/stagefright/foundation/ADebug.h> 36#include <media/stagefright/foundation/hexdump.h> 37#include <media/stagefright/FileSource.h> 38#include <media/stagefright/MediaDefs.h> 39#include <media/stagefright/MetaData.h> 40#include <media/stagefright/Utils.h> 41 42#include <ctype.h> 43#include <inttypes.h> 44#include <openssl/aes.h> 45#include <openssl/md5.h> 46 47namespace android { 48 49// static 50const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; 51const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; 52const int32_t PlaylistFetcher::kDownloadBlockSize = 2048; 53const int32_t PlaylistFetcher::kNumSkipFrames = 10; 54 55PlaylistFetcher::PlaylistFetcher( 56 const sp<AMessage> ¬ify, 57 const sp<LiveSession> &session, 58 const char *uri, 59 int32_t subtitleGeneration) 60 : mNotify(notify), 61 mStartTimeUsNotify(notify->dup()), 62 mSession(session), 63 mURI(uri), 64 mStreamTypeMask(0), 65 mStartTimeUs(-1ll), 66 mSegmentStartTimeUs(-1ll), 67 mDiscontinuitySeq(-1ll), 68 mStartTimeUsRelative(false), 69 mLastPlaylistFetchTimeUs(-1ll), 70 mSeqNumber(-1), 71 mNumRetries(0), 72 mStartup(true), 73 mAdaptive(false), 74 mPrepared(false), 75 mNextPTSTimeUs(-1ll), 76 mMonitorQueueGeneration(0), 77 mSubtitleGeneration(subtitleGeneration), 78 mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), 79 mFirstPTSValid(false), 80 mAbsoluteTimeAnchorUs(0ll), 81 mVideoBuffer(new AnotherPacketSource(NULL)) { 82 memset(mPlaylistHash, 0, sizeof(mPlaylistHash)); 83 mStartTimeUsNotify->setInt32("what", kWhatStartedAt); 84 mStartTimeUsNotify->setInt32("streamMask", 0); 85} 86 87PlaylistFetcher::~PlaylistFetcher() { 88} 89 90int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { 91 CHECK(mPlaylist != NULL); 92 93 int32_t firstSeqNumberInPlaylist; 94 if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( 95 "media-sequence", &firstSeqNumberInPlaylist)) { 96 firstSeqNumberInPlaylist = 0; 97 } 98 99 int32_t lastSeqNumberInPlaylist = 100 firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; 101 102 CHECK_GE(seqNumber, firstSeqNumberInPlaylist); 103 CHECK_LE(seqNumber, lastSeqNumberInPlaylist); 104 105 int64_t segmentStartUs = 0ll; 106 for (int32_t index = 0; 107 index < seqNumber - firstSeqNumberInPlaylist; ++index) { 108 sp<AMessage> itemMeta; 109 CHECK(mPlaylist->itemAt( 110 index, NULL /* uri */, &itemMeta)); 111 112 int64_t itemDurationUs; 113 CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); 114 115 segmentStartUs += itemDurationUs; 116 } 117 118 return segmentStartUs; 119} 120 121int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const { 122 int64_t nowUs = ALooper::GetNowUs(); 123 124 if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) { 125 CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY); 126 return 0ll; 127 } 128 129 if (mPlaylist->isComplete()) { 130 return (~0llu >> 1); 131 } 132 133 int32_t targetDurationSecs; 134 CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); 135 136 int64_t targetDurationUs = targetDurationSecs * 1000000ll; 137 138 int64_t minPlaylistAgeUs; 139 140 switch (mRefreshState) { 141 case INITIAL_MINIMUM_RELOAD_DELAY: 142 { 143 size_t n = mPlaylist->size(); 144 if (n > 0) { 145 sp<AMessage> itemMeta; 146 CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta)); 147 148 int64_t itemDurationUs; 149 CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); 150 151 minPlaylistAgeUs = itemDurationUs; 152 break; 153 } 154 155 // fall through 156 } 157 158 case FIRST_UNCHANGED_RELOAD_ATTEMPT: 159 { 160 minPlaylistAgeUs = targetDurationUs / 2; 161 break; 162 } 163 164 case SECOND_UNCHANGED_RELOAD_ATTEMPT: 165 { 166 minPlaylistAgeUs = (targetDurationUs * 3) / 2; 167 break; 168 } 169 170 case THIRD_UNCHANGED_RELOAD_ATTEMPT: 171 { 172 minPlaylistAgeUs = targetDurationUs * 3; 173 break; 174 } 175 176 default: 177 TRESPASS(); 178 break; 179 } 180 181 int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs; 182 return delayUs > 0ll ? delayUs : 0ll; 183} 184 185status_t PlaylistFetcher::decryptBuffer( 186 size_t playlistIndex, const sp<ABuffer> &buffer, 187 bool first) { 188 sp<AMessage> itemMeta; 189 bool found = false; 190 AString method; 191 192 for (ssize_t i = playlistIndex; i >= 0; --i) { 193 AString uri; 194 CHECK(mPlaylist->itemAt(i, &uri, &itemMeta)); 195 196 if (itemMeta->findString("cipher-method", &method)) { 197 found = true; 198 break; 199 } 200 } 201 202 if (!found) { 203 method = "NONE"; 204 } 205 buffer->meta()->setString("cipher-method", method.c_str()); 206 207 if (method == "NONE") { 208 return OK; 209 } else if (!(method == "AES-128")) { 210 ALOGE("Unsupported cipher method '%s'", method.c_str()); 211 return ERROR_UNSUPPORTED; 212 } 213 214 AString keyURI; 215 if (!itemMeta->findString("cipher-uri", &keyURI)) { 216 ALOGE("Missing key uri"); 217 return ERROR_MALFORMED; 218 } 219 220 ssize_t index = mAESKeyForURI.indexOfKey(keyURI); 221 222 sp<ABuffer> key; 223 if (index >= 0) { 224 key = mAESKeyForURI.valueAt(index); 225 } else { 226 ssize_t err = mSession->fetchFile(keyURI.c_str(), &key); 227 228 if (err < 0) { 229 ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str()); 230 return ERROR_IO; 231 } else if (key->size() != 16) { 232 ALOGE("key file '%s' wasn't 16 bytes in size.", keyURI.c_str()); 233 return ERROR_MALFORMED; 234 } 235 236 mAESKeyForURI.add(keyURI, key); 237 } 238 239 AES_KEY aes_key; 240 if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) { 241 ALOGE("failed to set AES decryption key."); 242 return UNKNOWN_ERROR; 243 } 244 245 size_t n = buffer->size(); 246 if (!n) { 247 return OK; 248 } 249 CHECK(n % 16 == 0); 250 251 if (first) { 252 // If decrypting the first block in a file, read the iv from the manifest 253 // or derive the iv from the file's sequence number. 254 255 AString iv; 256 if (itemMeta->findString("cipher-iv", &iv)) { 257 if ((!iv.startsWith("0x") && !iv.startsWith("0X")) 258 || iv.size() != 16 * 2 + 2) { 259 ALOGE("malformed cipher IV '%s'.", iv.c_str()); 260 return ERROR_MALFORMED; 261 } 262 263 memset(mAESInitVec, 0, sizeof(mAESInitVec)); 264 for (size_t i = 0; i < 16; ++i) { 265 char c1 = tolower(iv.c_str()[2 + 2 * i]); 266 char c2 = tolower(iv.c_str()[3 + 2 * i]); 267 if (!isxdigit(c1) || !isxdigit(c2)) { 268 ALOGE("malformed cipher IV '%s'.", iv.c_str()); 269 return ERROR_MALFORMED; 270 } 271 uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; 272 uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; 273 274 mAESInitVec[i] = nibble1 << 4 | nibble2; 275 } 276 } else { 277 memset(mAESInitVec, 0, sizeof(mAESInitVec)); 278 mAESInitVec[15] = mSeqNumber & 0xff; 279 mAESInitVec[14] = (mSeqNumber >> 8) & 0xff; 280 mAESInitVec[13] = (mSeqNumber >> 16) & 0xff; 281 mAESInitVec[12] = (mSeqNumber >> 24) & 0xff; 282 } 283 } 284 285 AES_cbc_encrypt( 286 buffer->data(), buffer->data(), buffer->size(), 287 &aes_key, mAESInitVec, AES_DECRYPT); 288 289 return OK; 290} 291 292status_t PlaylistFetcher::checkDecryptPadding(const sp<ABuffer> &buffer) { 293 status_t err; 294 AString method; 295 CHECK(buffer->meta()->findString("cipher-method", &method)); 296 if (method == "NONE") { 297 return OK; 298 } 299 300 uint8_t padding = 0; 301 if (buffer->size() > 0) { 302 padding = buffer->data()[buffer->size() - 1]; 303 } 304 305 if (padding > 16) { 306 return ERROR_MALFORMED; 307 } 308 309 for (size_t i = buffer->size() - padding; i < padding; i++) { 310 if (buffer->data()[i] != padding) { 311 return ERROR_MALFORMED; 312 } 313 } 314 315 buffer->setRange(buffer->offset(), buffer->size() - padding); 316 return OK; 317} 318 319void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) { 320 int64_t maxDelayUs = delayUsToRefreshPlaylist(); 321 if (maxDelayUs < minDelayUs) { 322 maxDelayUs = minDelayUs; 323 } 324 if (delayUs > maxDelayUs) { 325 ALOGV("Need to refresh playlist in %" PRId64 , maxDelayUs); 326 delayUs = maxDelayUs; 327 } 328 sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id()); 329 msg->setInt32("generation", mMonitorQueueGeneration); 330 msg->post(delayUs); 331} 332 333void PlaylistFetcher::cancelMonitorQueue() { 334 ++mMonitorQueueGeneration; 335} 336 337void PlaylistFetcher::startAsync( 338 const sp<AnotherPacketSource> &audioSource, 339 const sp<AnotherPacketSource> &videoSource, 340 const sp<AnotherPacketSource> &subtitleSource, 341 int64_t startTimeUs, 342 int64_t segmentStartTimeUs, 343 int32_t startDiscontinuitySeq, 344 bool adaptive) { 345 sp<AMessage> msg = new AMessage(kWhatStart, id()); 346 347 uint32_t streamTypeMask = 0ul; 348 349 if (audioSource != NULL) { 350 msg->setPointer("audioSource", audioSource.get()); 351 streamTypeMask |= LiveSession::STREAMTYPE_AUDIO; 352 } 353 354 if (videoSource != NULL) { 355 msg->setPointer("videoSource", videoSource.get()); 356 streamTypeMask |= LiveSession::STREAMTYPE_VIDEO; 357 } 358 359 if (subtitleSource != NULL) { 360 msg->setPointer("subtitleSource", subtitleSource.get()); 361 streamTypeMask |= LiveSession::STREAMTYPE_SUBTITLES; 362 } 363 364 msg->setInt32("streamTypeMask", streamTypeMask); 365 msg->setInt64("startTimeUs", startTimeUs); 366 msg->setInt64("segmentStartTimeUs", segmentStartTimeUs); 367 msg->setInt32("startDiscontinuitySeq", startDiscontinuitySeq); 368 msg->setInt32("adaptive", adaptive); 369 msg->post(); 370} 371 372void PlaylistFetcher::pauseAsync() { 373 (new AMessage(kWhatPause, id()))->post(); 374} 375 376void PlaylistFetcher::stopAsync(bool clear) { 377 sp<AMessage> msg = new AMessage(kWhatStop, id()); 378 msg->setInt32("clear", clear); 379 msg->post(); 380} 381 382void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> ¶ms) { 383 AMessage* msg = new AMessage(kWhatResumeUntil, id()); 384 msg->setMessage("params", params); 385 msg->post(); 386} 387 388void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { 389 switch (msg->what()) { 390 case kWhatStart: 391 { 392 status_t err = onStart(msg); 393 394 sp<AMessage> notify = mNotify->dup(); 395 notify->setInt32("what", kWhatStarted); 396 notify->setInt32("err", err); 397 notify->post(); 398 break; 399 } 400 401 case kWhatPause: 402 { 403 onPause(); 404 405 sp<AMessage> notify = mNotify->dup(); 406 notify->setInt32("what", kWhatPaused); 407 notify->post(); 408 break; 409 } 410 411 case kWhatStop: 412 { 413 onStop(msg); 414 415 sp<AMessage> notify = mNotify->dup(); 416 notify->setInt32("what", kWhatStopped); 417 notify->post(); 418 break; 419 } 420 421 case kWhatMonitorQueue: 422 case kWhatDownloadNext: 423 { 424 int32_t generation; 425 CHECK(msg->findInt32("generation", &generation)); 426 427 if (generation != mMonitorQueueGeneration) { 428 // Stale event 429 break; 430 } 431 432 if (msg->what() == kWhatMonitorQueue) { 433 onMonitorQueue(); 434 } else { 435 onDownloadNext(); 436 } 437 break; 438 } 439 440 case kWhatResumeUntil: 441 { 442 onResumeUntil(msg); 443 break; 444 } 445 446 default: 447 TRESPASS(); 448 } 449} 450 451status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { 452 mPacketSources.clear(); 453 454 uint32_t streamTypeMask; 455 CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask)); 456 457 int64_t startTimeUs; 458 int64_t segmentStartTimeUs; 459 int32_t startDiscontinuitySeq; 460 int32_t adaptive; 461 CHECK(msg->findInt64("startTimeUs", &startTimeUs)); 462 CHECK(msg->findInt64("segmentStartTimeUs", &segmentStartTimeUs)); 463 CHECK(msg->findInt32("startDiscontinuitySeq", &startDiscontinuitySeq)); 464 CHECK(msg->findInt32("adaptive", &adaptive)); 465 466 if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) { 467 void *ptr; 468 CHECK(msg->findPointer("audioSource", &ptr)); 469 470 mPacketSources.add( 471 LiveSession::STREAMTYPE_AUDIO, 472 static_cast<AnotherPacketSource *>(ptr)); 473 } 474 475 if (streamTypeMask & LiveSession::STREAMTYPE_VIDEO) { 476 void *ptr; 477 CHECK(msg->findPointer("videoSource", &ptr)); 478 479 mPacketSources.add( 480 LiveSession::STREAMTYPE_VIDEO, 481 static_cast<AnotherPacketSource *>(ptr)); 482 } 483 484 if (streamTypeMask & LiveSession::STREAMTYPE_SUBTITLES) { 485 void *ptr; 486 CHECK(msg->findPointer("subtitleSource", &ptr)); 487 488 mPacketSources.add( 489 LiveSession::STREAMTYPE_SUBTITLES, 490 static_cast<AnotherPacketSource *>(ptr)); 491 } 492 493 mStreamTypeMask = streamTypeMask; 494 495 mSegmentStartTimeUs = segmentStartTimeUs; 496 mDiscontinuitySeq = startDiscontinuitySeq; 497 498 if (startTimeUs >= 0) { 499 mStartTimeUs = startTimeUs; 500 mSeqNumber = -1; 501 mStartup = true; 502 mPrepared = false; 503 mAdaptive = adaptive; 504 } 505 506 postMonitorQueue(); 507 508 return OK; 509} 510 511void PlaylistFetcher::onPause() { 512 cancelMonitorQueue(); 513} 514 515void PlaylistFetcher::onStop(const sp<AMessage> &msg) { 516 cancelMonitorQueue(); 517 518 int32_t clear; 519 CHECK(msg->findInt32("clear", &clear)); 520 if (clear) { 521 for (size_t i = 0; i < mPacketSources.size(); i++) { 522 sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); 523 packetSource->clear(); 524 } 525 } 526 527 mPacketSources.clear(); 528 mStreamTypeMask = 0; 529} 530 531// Resume until we have reached the boundary timestamps listed in `msg`; when 532// the remaining time is too short (within a resume threshold) stop immediately 533// instead. 534status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) { 535 sp<AMessage> params; 536 CHECK(msg->findMessage("params", ¶ms)); 537 538 bool stop = false; 539 for (size_t i = 0; i < mPacketSources.size(); i++) { 540 sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); 541 542 const char *stopKey; 543 int streamType = mPacketSources.keyAt(i); 544 switch (streamType) { 545 case LiveSession::STREAMTYPE_VIDEO: 546 stopKey = "timeUsVideo"; 547 break; 548 549 case LiveSession::STREAMTYPE_AUDIO: 550 stopKey = "timeUsAudio"; 551 break; 552 553 case LiveSession::STREAMTYPE_SUBTITLES: 554 stopKey = "timeUsSubtitle"; 555 break; 556 557 default: 558 TRESPASS(); 559 } 560 561 // Don't resume if we would stop within a resume threshold. 562 int32_t discontinuitySeq; 563 int64_t latestTimeUs = 0, stopTimeUs = 0; 564 sp<AMessage> latestMeta = packetSource->getLatestDequeuedMeta(); 565 if (latestMeta != NULL 566 && latestMeta->findInt32("discontinuitySeq", &discontinuitySeq) 567 && discontinuitySeq == mDiscontinuitySeq 568 && latestMeta->findInt64("timeUs", &latestTimeUs) 569 && params->findInt64(stopKey, &stopTimeUs) 570 && stopTimeUs - latestTimeUs < resumeThreshold(latestMeta)) { 571 stop = true; 572 } 573 } 574 575 if (stop) { 576 for (size_t i = 0; i < mPacketSources.size(); i++) { 577 mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer()); 578 } 579 stopAsync(/* clear = */ false); 580 return OK; 581 } 582 583 mStopParams = params; 584 postMonitorQueue(); 585 586 return OK; 587} 588 589void PlaylistFetcher::notifyError(status_t err) { 590 sp<AMessage> notify = mNotify->dup(); 591 notify->setInt32("what", kWhatError); 592 notify->setInt32("err", err); 593 notify->post(); 594} 595 596void PlaylistFetcher::queueDiscontinuity( 597 ATSParser::DiscontinuityType type, const sp<AMessage> &extra) { 598 for (size_t i = 0; i < mPacketSources.size(); ++i) { 599 // do not discard buffer upon #EXT-X-DISCONTINUITY tag 600 // (seek will discard buffer by abandoning old fetchers) 601 mPacketSources.valueAt(i)->queueDiscontinuity( 602 type, extra, false /* discard */); 603 } 604} 605 606void PlaylistFetcher::onMonitorQueue() { 607 bool downloadMore = false; 608 refreshPlaylist(); 609 610 int32_t targetDurationSecs; 611 int64_t targetDurationUs = kMinBufferedDurationUs; 612 if (mPlaylist != NULL) { 613 if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( 614 "target-duration", &targetDurationSecs)) { 615 ALOGE("Playlist is missing required EXT-X-TARGETDURATION tag"); 616 notifyError(ERROR_MALFORMED); 617 return; 618 } 619 targetDurationUs = targetDurationSecs * 1000000ll; 620 } 621 622 // buffer at least 3 times the target duration, or up to 10 seconds 623 int64_t durationToBufferUs = targetDurationUs * 3; 624 if (durationToBufferUs > kMinBufferedDurationUs) { 625 durationToBufferUs = kMinBufferedDurationUs; 626 } 627 628 int64_t bufferedDurationUs = 0ll; 629 status_t finalResult = NOT_ENOUGH_DATA; 630 if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) { 631 sp<AnotherPacketSource> packetSource = 632 mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); 633 634 bufferedDurationUs = 635 packetSource->getBufferedDurationUs(&finalResult); 636 finalResult = OK; 637 } else { 638 // Use max stream duration to prevent us from waiting on a non-existent stream; 639 // when we cannot make out from the manifest what streams are included in a playlist 640 // we might assume extra streams. 641 for (size_t i = 0; i < mPacketSources.size(); ++i) { 642 if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { 643 continue; 644 } 645 646 int64_t bufferedStreamDurationUs = 647 mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); 648 ALOGV("buffered %" PRId64 " for stream %d", 649 bufferedStreamDurationUs, mPacketSources.keyAt(i)); 650 if (bufferedStreamDurationUs > bufferedDurationUs) { 651 bufferedDurationUs = bufferedStreamDurationUs; 652 } 653 } 654 } 655 downloadMore = (bufferedDurationUs < durationToBufferUs); 656 657 // signal start if buffered up at least the target size 658 if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) { 659 mPrepared = true; 660 661 ALOGV("prepared, buffered=%" PRId64 " > %" PRId64 "", 662 bufferedDurationUs, targetDurationUs); 663 sp<AMessage> msg = mNotify->dup(); 664 msg->setInt32("what", kWhatTemporarilyDoneFetching); 665 msg->post(); 666 } 667 668 if (finalResult == OK && downloadMore) { 669 ALOGV("monitoring, buffered=%" PRId64 " < %" PRId64 "", 670 bufferedDurationUs, durationToBufferUs); 671 // delay the next download slightly; hopefully this gives other concurrent fetchers 672 // a better chance to run. 673 // onDownloadNext(); 674 sp<AMessage> msg = new AMessage(kWhatDownloadNext, id()); 675 msg->setInt32("generation", mMonitorQueueGeneration); 676 msg->post(1000l); 677 } else { 678 // Nothing to do yet, try again in a second. 679 680 sp<AMessage> msg = mNotify->dup(); 681 msg->setInt32("what", kWhatTemporarilyDoneFetching); 682 msg->post(); 683 684 int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2; 685 ALOGV("pausing for %" PRId64 ", buffered=%" PRId64 " > %" PRId64 "", 686 delayUs, bufferedDurationUs, durationToBufferUs); 687 // :TRICKY: need to enforce minimum delay because the delay to 688 // refresh the playlist will become 0 689 postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0); 690 } 691} 692 693status_t PlaylistFetcher::refreshPlaylist() { 694 if (delayUsToRefreshPlaylist() <= 0) { 695 bool unchanged; 696 sp<M3UParser> playlist = mSession->fetchPlaylist( 697 mURI.c_str(), mPlaylistHash, &unchanged); 698 699 if (playlist == NULL) { 700 if (unchanged) { 701 // We succeeded in fetching the playlist, but it was 702 // unchanged from the last time we tried. 703 704 if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) { 705 mRefreshState = (RefreshState)(mRefreshState + 1); 706 } 707 } else { 708 ALOGE("failed to load playlist at url '%s'", uriDebugString(mURI).c_str()); 709 return ERROR_IO; 710 } 711 } else { 712 mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; 713 mPlaylist = playlist; 714 715 if (mPlaylist->isComplete() || mPlaylist->isEvent()) { 716 updateDuration(); 717 } 718 } 719 720 mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); 721 } 722 return OK; 723} 724 725// static 726bool PlaylistFetcher::bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer) { 727 return buffer->size() > 0 && buffer->data()[0] == 0x47; 728} 729 730void PlaylistFetcher::onDownloadNext() { 731 status_t err = refreshPlaylist(); 732 int32_t firstSeqNumberInPlaylist = 0; 733 int32_t lastSeqNumberInPlaylist = 0; 734 bool discontinuity = false; 735 736 if (mPlaylist != NULL) { 737 if (mPlaylist->meta() != NULL) { 738 mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist); 739 } 740 741 lastSeqNumberInPlaylist = 742 firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; 743 744 if (mDiscontinuitySeq < 0) { 745 mDiscontinuitySeq = mPlaylist->getDiscontinuitySeq(); 746 } 747 } 748 749 if (mPlaylist != NULL && mSeqNumber < 0) { 750 CHECK_GE(mStartTimeUs, 0ll); 751 752 if (mSegmentStartTimeUs < 0) { 753 if (!mPlaylist->isComplete() && !mPlaylist->isEvent()) { 754 // If this is a live session, start 3 segments from the end on connect 755 mSeqNumber = lastSeqNumberInPlaylist - 3; 756 if (mSeqNumber < firstSeqNumberInPlaylist) { 757 mSeqNumber = firstSeqNumberInPlaylist; 758 } 759 } else { 760 mSeqNumber = getSeqNumberForTime(mStartTimeUs); 761 mStartTimeUs -= getSegmentStartTimeUs(mSeqNumber); 762 } 763 mStartTimeUsRelative = true; 764 ALOGV("Initial sequence number for time %" PRId64 " is %d from (%d .. %d)", 765 mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist, 766 lastSeqNumberInPlaylist); 767 } else { 768 mSeqNumber = getSeqNumberForTime(mSegmentStartTimeUs); 769 if (mAdaptive) { 770 // avoid double fetch/decode 771 mSeqNumber += 1; 772 } 773 ssize_t minSeq = getSeqNumberForDiscontinuity(mDiscontinuitySeq); 774 if (mSeqNumber < minSeq) { 775 mSeqNumber = minSeq; 776 } 777 778 if (mSeqNumber < firstSeqNumberInPlaylist) { 779 mSeqNumber = firstSeqNumberInPlaylist; 780 } 781 782 if (mSeqNumber > lastSeqNumberInPlaylist) { 783 mSeqNumber = lastSeqNumberInPlaylist; 784 } 785 ALOGV("Initial sequence number for live event %d from (%d .. %d)", 786 mSeqNumber, firstSeqNumberInPlaylist, 787 lastSeqNumberInPlaylist); 788 } 789 } 790 791 // if mPlaylist is NULL then err must be non-OK; but the other way around might not be true 792 if (mSeqNumber < firstSeqNumberInPlaylist 793 || mSeqNumber > lastSeqNumberInPlaylist 794 || err != OK) { 795 if ((err != OK || !mPlaylist->isComplete()) && mNumRetries < kMaxNumRetries) { 796 ++mNumRetries; 797 798 if (mSeqNumber > lastSeqNumberInPlaylist || err != OK) { 799 // make sure we reach this retry logic on refresh failures 800 // by adding an err != OK clause to all enclosing if's. 801 802 // refresh in increasing fraction (1/2, 1/3, ...) of the 803 // playlist's target duration or 3 seconds, whichever is less 804 int64_t delayUs = kMaxMonitorDelayUs; 805 if (mPlaylist != NULL && mPlaylist->meta() != NULL) { 806 int32_t targetDurationSecs; 807 CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); 808 delayUs = mPlaylist->size() * targetDurationSecs * 809 1000000ll / (1 + mNumRetries); 810 } 811 if (delayUs > kMaxMonitorDelayUs) { 812 delayUs = kMaxMonitorDelayUs; 813 } 814 ALOGV("sequence number high: %d from (%d .. %d), " 815 "monitor in %" PRId64 " (retry=%d)", 816 mSeqNumber, firstSeqNumberInPlaylist, 817 lastSeqNumberInPlaylist, delayUs, mNumRetries); 818 postMonitorQueue(delayUs); 819 return; 820 } 821 822 if (err != OK) { 823 notifyError(err); 824 return; 825 } 826 827 // we've missed the boat, let's start 3 segments prior to the latest sequence 828 // number available and signal a discontinuity. 829 830 ALOGI("We've missed the boat, restarting playback." 831 " mStartup=%d, was looking for %d in %d-%d", 832 mStartup, mSeqNumber, firstSeqNumberInPlaylist, 833 lastSeqNumberInPlaylist); 834 if (mStopParams != NULL) { 835 // we should have kept on fetching until we hit the boundaries in mStopParams, 836 // but since the segments we are supposed to fetch have already rolled off 837 // the playlist, i.e. we have already missed the boat, we inevitably have to 838 // skip. 839 for (size_t i = 0; i < mPacketSources.size(); i++) { 840 sp<ABuffer> formatChange = mSession->createFormatChangeBuffer(); 841 mPacketSources.valueAt(i)->queueAccessUnit(formatChange); 842 } 843 stopAsync(/* clear = */ false); 844 return; 845 } 846 mSeqNumber = lastSeqNumberInPlaylist - 3; 847 if (mSeqNumber < firstSeqNumberInPlaylist) { 848 mSeqNumber = firstSeqNumberInPlaylist; 849 } 850 discontinuity = true; 851 852 // fall through 853 } else { 854 ALOGE("Cannot find sequence number %d in playlist " 855 "(contains %d - %d)", 856 mSeqNumber, firstSeqNumberInPlaylist, 857 firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1); 858 859 notifyError(ERROR_END_OF_STREAM); 860 return; 861 } 862 } 863 864 mNumRetries = 0; 865 866 AString uri; 867 sp<AMessage> itemMeta; 868 CHECK(mPlaylist->itemAt( 869 mSeqNumber - firstSeqNumberInPlaylist, 870 &uri, 871 &itemMeta)); 872 873 int32_t val; 874 if (itemMeta->findInt32("discontinuity", &val) && val != 0) { 875 mDiscontinuitySeq++; 876 discontinuity = true; 877 } 878 879 int64_t range_offset, range_length; 880 if (!itemMeta->findInt64("range-offset", &range_offset) 881 || !itemMeta->findInt64("range-length", &range_length)) { 882 range_offset = 0; 883 range_length = -1; 884 } 885 886 ALOGV("fetching segment %d from (%d .. %d)", 887 mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); 888 889 ALOGV("fetching '%s'", uri.c_str()); 890 891 sp<DataSource> source; 892 sp<ABuffer> buffer, tsBuffer; 893 // decrypt a junk buffer to prefetch key; since a session uses only one http connection, 894 // this avoids interleaved connections to the key and segment file. 895 { 896 sp<ABuffer> junk = new ABuffer(16); 897 junk->setRange(0, 16); 898 status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, junk, 899 true /* first */); 900 if (err != OK) { 901 notifyError(err); 902 return; 903 } 904 } 905 906 // block-wise download 907 bool startup = mStartup; 908 ssize_t bytesRead; 909 do { 910 bytesRead = mSession->fetchFile( 911 uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source); 912 913 if (bytesRead < 0) { 914 status_t err = bytesRead; 915 ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); 916 notifyError(err); 917 return; 918 } 919 920 CHECK(buffer != NULL); 921 922 size_t size = buffer->size(); 923 // Set decryption range. 924 buffer->setRange(size - bytesRead, bytesRead); 925 status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer, 926 buffer->offset() == 0 /* first */); 927 // Unset decryption range. 928 buffer->setRange(0, size); 929 930 if (err != OK) { 931 ALOGE("decryptBuffer failed w/ error %d", err); 932 933 notifyError(err); 934 return; 935 } 936 937 if (startup || discontinuity) { 938 // Signal discontinuity. 939 940 if (mPlaylist->isComplete() || mPlaylist->isEvent()) { 941 // If this was a live event this made no sense since 942 // we don't have access to all the segment before the current 943 // one. 944 mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber); 945 } 946 947 if (discontinuity) { 948 ALOGI("queueing discontinuity (explicit=%d)", discontinuity); 949 950 queueDiscontinuity( 951 ATSParser::DISCONTINUITY_FORMATCHANGE, 952 NULL /* extra */); 953 954 discontinuity = false; 955 } 956 957 startup = false; 958 } 959 960 err = OK; 961 if (bufferStartsWithTsSyncByte(buffer)) { 962 // Incremental extraction is only supported for MPEG2 transport streams. 963 if (tsBuffer == NULL) { 964 tsBuffer = new ABuffer(buffer->data(), buffer->capacity()); 965 tsBuffer->setRange(0, 0); 966 } else if (tsBuffer->capacity() != buffer->capacity()) { 967 size_t tsOff = tsBuffer->offset(), tsSize = tsBuffer->size(); 968 tsBuffer = new ABuffer(buffer->data(), buffer->capacity()); 969 tsBuffer->setRange(tsOff, tsSize); 970 } 971 tsBuffer->setRange(tsBuffer->offset(), tsBuffer->size() + bytesRead); 972 973 err = extractAndQueueAccessUnitsFromTs(tsBuffer); 974 } 975 976 if (err == -EAGAIN) { 977 // starting sequence number too low/high 978 mTSParser.clear(); 979 postMonitorQueue(); 980 return; 981 } else if (err == ERROR_OUT_OF_RANGE) { 982 // reached stopping point 983 stopAsync(/* clear = */ false); 984 return; 985 } else if (err != OK) { 986 notifyError(err); 987 return; 988 } 989 990 } while (bytesRead != 0); 991 992 if (bufferStartsWithTsSyncByte(buffer)) { 993 // If we don't see a stream in the program table after fetching a full ts segment 994 // mark it as nonexistent. 995 const size_t kNumTypes = ATSParser::NUM_SOURCE_TYPES; 996 ATSParser::SourceType srcTypes[kNumTypes] = 997 { ATSParser::VIDEO, ATSParser::AUDIO }; 998 LiveSession::StreamType streamTypes[kNumTypes] = 999 { LiveSession::STREAMTYPE_VIDEO, LiveSession::STREAMTYPE_AUDIO }; 1000 1001 for (size_t i = 0; i < kNumTypes; i++) { 1002 ATSParser::SourceType srcType = srcTypes[i]; 1003 LiveSession::StreamType streamType = streamTypes[i]; 1004 1005 sp<AnotherPacketSource> source = 1006 static_cast<AnotherPacketSource *>( 1007 mTSParser->getSource(srcType).get()); 1008 1009 if (!mTSParser->hasSource(srcType)) { 1010 ALOGW("MPEG2 Transport stream does not contain %s data.", 1011 srcType == ATSParser::VIDEO ? "video" : "audio"); 1012 1013 mStreamTypeMask &= ~streamType; 1014 mPacketSources.removeItem(streamType); 1015 } 1016 } 1017 1018 } 1019 1020 if (checkDecryptPadding(buffer) != OK) { 1021 ALOGE("Incorrect padding bytes after decryption."); 1022 notifyError(ERROR_MALFORMED); 1023 return; 1024 } 1025 1026 err = OK; 1027 if (tsBuffer != NULL) { 1028 AString method; 1029 CHECK(buffer->meta()->findString("cipher-method", &method)); 1030 if ((tsBuffer->size() > 0 && method == "NONE") 1031 || tsBuffer->size() > 16) { 1032 ALOGE("MPEG2 transport stream is not an even multiple of 188 " 1033 "bytes in length."); 1034 notifyError(ERROR_MALFORMED); 1035 return; 1036 } 1037 } 1038 1039 // bulk extract non-ts files 1040 if (tsBuffer == NULL) { 1041 err = extractAndQueueAccessUnits(buffer, itemMeta); 1042 if (err == -EAGAIN) { 1043 // starting sequence number too low/high 1044 postMonitorQueue(); 1045 return; 1046 } else if (err == ERROR_OUT_OF_RANGE) { 1047 // reached stopping point 1048 stopAsync(/* clear = */false); 1049 return; 1050 } 1051 } 1052 1053 if (err != OK) { 1054 notifyError(err); 1055 return; 1056 } 1057 1058 ++mSeqNumber; 1059 1060 postMonitorQueue(); 1061} 1062 1063int32_t PlaylistFetcher::getSeqNumberWithAnchorTime(int64_t anchorTimeUs) const { 1064 int32_t firstSeqNumberInPlaylist, lastSeqNumberInPlaylist; 1065 if (mPlaylist->meta() == NULL 1066 || !mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist)) { 1067 firstSeqNumberInPlaylist = 0; 1068 } 1069 lastSeqNumberInPlaylist = firstSeqNumberInPlaylist + mPlaylist->size() - 1; 1070 1071 int32_t index = mSeqNumber - firstSeqNumberInPlaylist - 1; 1072 while (index >= 0 && anchorTimeUs > mStartTimeUs) { 1073 sp<AMessage> itemMeta; 1074 CHECK(mPlaylist->itemAt(index, NULL /* uri */, &itemMeta)); 1075 1076 int64_t itemDurationUs; 1077 CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); 1078 1079 anchorTimeUs -= itemDurationUs; 1080 --index; 1081 } 1082 1083 int32_t newSeqNumber = firstSeqNumberInPlaylist + index + 1; 1084 if (newSeqNumber <= lastSeqNumberInPlaylist) { 1085 return newSeqNumber; 1086 } else { 1087 return lastSeqNumberInPlaylist; 1088 } 1089} 1090 1091int32_t PlaylistFetcher::getSeqNumberForDiscontinuity(size_t discontinuitySeq) const { 1092 int32_t firstSeqNumberInPlaylist; 1093 if (mPlaylist->meta() == NULL 1094 || !mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist)) { 1095 firstSeqNumberInPlaylist = 0; 1096 } 1097 1098 size_t curDiscontinuitySeq = mPlaylist->getDiscontinuitySeq(); 1099 if (discontinuitySeq < curDiscontinuitySeq) { 1100 return firstSeqNumberInPlaylist <= 0 ? 0 : (firstSeqNumberInPlaylist - 1); 1101 } 1102 1103 size_t index = 0; 1104 while (index < mPlaylist->size()) { 1105 sp<AMessage> itemMeta; 1106 CHECK(mPlaylist->itemAt( index, NULL /* uri */, &itemMeta)); 1107 1108 int64_t discontinuity; 1109 if (itemMeta->findInt64("discontinuity", &discontinuity)) { 1110 curDiscontinuitySeq++; 1111 } 1112 1113 if (curDiscontinuitySeq == discontinuitySeq) { 1114 return firstSeqNumberInPlaylist + index; 1115 } 1116 1117 ++index; 1118 } 1119 1120 return firstSeqNumberInPlaylist + mPlaylist->size(); 1121} 1122 1123int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { 1124 int32_t firstSeqNumberInPlaylist; 1125 if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( 1126 "media-sequence", &firstSeqNumberInPlaylist)) { 1127 firstSeqNumberInPlaylist = 0; 1128 } 1129 1130 size_t index = 0; 1131 int64_t segmentStartUs = 0; 1132 while (index < mPlaylist->size()) { 1133 sp<AMessage> itemMeta; 1134 CHECK(mPlaylist->itemAt( 1135 index, NULL /* uri */, &itemMeta)); 1136 1137 int64_t itemDurationUs; 1138 CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); 1139 1140 if (timeUs < segmentStartUs + itemDurationUs) { 1141 break; 1142 } 1143 1144 segmentStartUs += itemDurationUs; 1145 ++index; 1146 } 1147 1148 if (index >= mPlaylist->size()) { 1149 index = mPlaylist->size() - 1; 1150 } 1151 1152 return firstSeqNumberInPlaylist + index; 1153} 1154 1155const sp<ABuffer> &PlaylistFetcher::setAccessUnitProperties( 1156 const sp<ABuffer> &accessUnit, const sp<AnotherPacketSource> &source, bool discard) { 1157 sp<MetaData> format = source->getFormat(); 1158 if (format != NULL) { 1159 // for simplicity, store a reference to the format in each unit 1160 accessUnit->meta()->setObject("format", format); 1161 } 1162 1163 if (discard) { 1164 accessUnit->meta()->setInt32("discard", discard); 1165 } 1166 1167 accessUnit->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq); 1168 accessUnit->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber)); 1169 return accessUnit; 1170} 1171 1172status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &buffer) { 1173 if (mTSParser == NULL) { 1174 // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers. 1175 mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); 1176 } 1177 1178 if (mNextPTSTimeUs >= 0ll) { 1179 sp<AMessage> extra = new AMessage; 1180 // Since we are using absolute timestamps, signal an offset of 0 to prevent 1181 // ATSParser from skewing the timestamps of access units. 1182 extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0); 1183 1184 mTSParser->signalDiscontinuity( 1185 ATSParser::DISCONTINUITY_TIME, extra); 1186 1187 mAbsoluteTimeAnchorUs = mNextPTSTimeUs; 1188 mNextPTSTimeUs = -1ll; 1189 mFirstPTSValid = false; 1190 } 1191 1192 size_t offset = 0; 1193 while (offset + 188 <= buffer->size()) { 1194 status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); 1195 1196 if (err != OK) { 1197 return err; 1198 } 1199 1200 offset += 188; 1201 } 1202 // setRange to indicate consumed bytes. 1203 buffer->setRange(buffer->offset() + offset, buffer->size() - offset); 1204 1205 status_t err = OK; 1206 for (size_t i = mPacketSources.size(); i-- > 0;) { 1207 sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); 1208 1209 const char *key; 1210 ATSParser::SourceType type; 1211 const LiveSession::StreamType stream = mPacketSources.keyAt(i); 1212 switch (stream) { 1213 case LiveSession::STREAMTYPE_VIDEO: 1214 type = ATSParser::VIDEO; 1215 key = "timeUsVideo"; 1216 break; 1217 1218 case LiveSession::STREAMTYPE_AUDIO: 1219 type = ATSParser::AUDIO; 1220 key = "timeUsAudio"; 1221 break; 1222 1223 case LiveSession::STREAMTYPE_SUBTITLES: 1224 { 1225 ALOGE("MPEG2 Transport streams do not contain subtitles."); 1226 return ERROR_MALFORMED; 1227 break; 1228 } 1229 1230 default: 1231 TRESPASS(); 1232 } 1233 1234 sp<AnotherPacketSource> source = 1235 static_cast<AnotherPacketSource *>( 1236 mTSParser->getSource(type).get()); 1237 1238 if (source == NULL) { 1239 continue; 1240 } 1241 1242 int64_t timeUs; 1243 sp<ABuffer> accessUnit; 1244 status_t finalResult; 1245 while (source->hasBufferAvailable(&finalResult) 1246 && source->dequeueAccessUnit(&accessUnit) == OK) { 1247 1248 CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); 1249 1250 if (mStartup) { 1251 if (!mFirstPTSValid) { 1252 mFirstTimeUs = timeUs; 1253 mFirstPTSValid = true; 1254 } 1255 if (mStartTimeUsRelative) { 1256 timeUs -= mFirstTimeUs; 1257 if (timeUs < 0) { 1258 timeUs = 0; 1259 } 1260 } 1261 1262 if (timeUs < mStartTimeUs) { 1263 // buffer up to the closest preceding IDR frame 1264 ALOGV("timeUs %" PRId64 " us < mStartTimeUs %" PRId64 " us", 1265 timeUs, mStartTimeUs); 1266 const char *mime; 1267 sp<MetaData> format = source->getFormat(); 1268 bool isAvc = false; 1269 if (format != NULL && format->findCString(kKeyMIMEType, &mime) 1270 && !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) { 1271 isAvc = true; 1272 } 1273 if (isAvc && IsIDR(accessUnit)) { 1274 mVideoBuffer->clear(); 1275 } 1276 if (isAvc) { 1277 mVideoBuffer->queueAccessUnit(accessUnit); 1278 } 1279 1280 continue; 1281 } 1282 } 1283 1284 CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); 1285 if (mStartTimeUsNotify != NULL && timeUs > mStartTimeUs) { 1286 int32_t firstSeqNumberInPlaylist; 1287 if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( 1288 "media-sequence", &firstSeqNumberInPlaylist)) { 1289 firstSeqNumberInPlaylist = 0; 1290 } 1291 1292 int32_t targetDurationSecs; 1293 CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); 1294 int64_t targetDurationUs = targetDurationSecs * 1000000ll; 1295 // mStartup 1296 // mStartup is true until we have queued a packet for all the streams 1297 // we are fetching. We queue packets whose timestamps are greater than 1298 // mStartTimeUs. 1299 // mSegmentStartTimeUs >= 0 1300 // mSegmentStartTimeUs is non-negative when adapting or switching tracks 1301 // mSeqNumber > firstSeqNumberInPlaylist 1302 // don't decrement mSeqNumber if it already points to the 1st segment 1303 // timeUs - mStartTimeUs > targetDurationUs: 1304 // This and the 2 above conditions should only happen when adapting in a live 1305 // stream; the old fetcher has already fetched to mStartTimeUs; the new fetcher 1306 // would start fetching after timeUs, which should be greater than mStartTimeUs; 1307 // the old fetcher would then continue fetching data until timeUs. We don't want 1308 // timeUs to be too far ahead of mStartTimeUs because we want the old fetcher to 1309 // stop as early as possible. The definition of being "too far ahead" is 1310 // arbitrary; here we use targetDurationUs as threshold. 1311 if (mStartup && mSegmentStartTimeUs >= 0 1312 && mSeqNumber > firstSeqNumberInPlaylist 1313 && timeUs - mStartTimeUs > targetDurationUs) { 1314 // we just guessed a starting timestamp that is too high when adapting in a 1315 // live stream; re-adjust based on the actual timestamp extracted from the 1316 // media segment; if we didn't move backward after the re-adjustment 1317 // (newSeqNumber), start at least 1 segment prior. 1318 int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs); 1319 if (newSeqNumber >= mSeqNumber) { 1320 --mSeqNumber; 1321 } else { 1322 mSeqNumber = newSeqNumber; 1323 } 1324 mStartTimeUsNotify = mNotify->dup(); 1325 mStartTimeUsNotify->setInt32("what", kWhatStartedAt); 1326 return -EAGAIN; 1327 } 1328 1329 int32_t seq; 1330 if (!mStartTimeUsNotify->findInt32("discontinuitySeq", &seq)) { 1331 mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq); 1332 } 1333 int64_t startTimeUs; 1334 if (!mStartTimeUsNotify->findInt64(key, &startTimeUs)) { 1335 mStartTimeUsNotify->setInt64(key, timeUs); 1336 1337 uint32_t streamMask = 0; 1338 mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask); 1339 streamMask |= mPacketSources.keyAt(i); 1340 mStartTimeUsNotify->setInt32("streamMask", streamMask); 1341 1342 if (streamMask == mStreamTypeMask) { 1343 mStartup = false; 1344 mStartTimeUsNotify->post(); 1345 mStartTimeUsNotify.clear(); 1346 } 1347 } 1348 } 1349 1350 if (mStopParams != NULL) { 1351 // Queue discontinuity in original stream. 1352 int32_t discontinuitySeq; 1353 int64_t stopTimeUs; 1354 if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq) 1355 || discontinuitySeq > mDiscontinuitySeq 1356 || !mStopParams->findInt64(key, &stopTimeUs) 1357 || (discontinuitySeq == mDiscontinuitySeq 1358 && timeUs >= stopTimeUs)) { 1359 packetSource->queueAccessUnit(mSession->createFormatChangeBuffer()); 1360 mStreamTypeMask &= ~stream; 1361 mPacketSources.removeItemsAt(i); 1362 break; 1363 } 1364 } 1365 1366 // Note that we do NOT dequeue any discontinuities except for format change. 1367 if (stream == LiveSession::STREAMTYPE_VIDEO) { 1368 const bool discard = true; 1369 status_t status; 1370 while (mVideoBuffer->hasBufferAvailable(&status)) { 1371 sp<ABuffer> videoBuffer; 1372 mVideoBuffer->dequeueAccessUnit(&videoBuffer); 1373 setAccessUnitProperties(videoBuffer, source, discard); 1374 packetSource->queueAccessUnit(videoBuffer); 1375 } 1376 } 1377 1378 setAccessUnitProperties(accessUnit, source); 1379 packetSource->queueAccessUnit(accessUnit); 1380 } 1381 1382 if (err != OK) { 1383 break; 1384 } 1385 } 1386 1387 if (err != OK) { 1388 for (size_t i = mPacketSources.size(); i-- > 0;) { 1389 sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); 1390 packetSource->clear(); 1391 } 1392 return err; 1393 } 1394 1395 if (!mStreamTypeMask) { 1396 // Signal gap is filled between original and new stream. 1397 ALOGV("ERROR OUT OF RANGE"); 1398 return ERROR_OUT_OF_RANGE; 1399 } 1400 1401 return OK; 1402} 1403 1404/* static */ 1405bool PlaylistFetcher::bufferStartsWithWebVTTMagicSequence( 1406 const sp<ABuffer> &buffer) { 1407 size_t pos = 0; 1408 1409 // skip possible BOM 1410 if (buffer->size() >= pos + 3 && 1411 !memcmp("\xef\xbb\xbf", buffer->data() + pos, 3)) { 1412 pos += 3; 1413 } 1414 1415 // accept WEBVTT followed by SPACE, TAB or (CR) LF 1416 if (buffer->size() < pos + 6 || 1417 memcmp("WEBVTT", buffer->data() + pos, 6)) { 1418 return false; 1419 } 1420 pos += 6; 1421 1422 if (buffer->size() == pos) { 1423 return true; 1424 } 1425 1426 uint8_t sep = buffer->data()[pos]; 1427 return sep == ' ' || sep == '\t' || sep == '\n' || sep == '\r'; 1428} 1429 1430status_t PlaylistFetcher::extractAndQueueAccessUnits( 1431 const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) { 1432 if (bufferStartsWithWebVTTMagicSequence(buffer)) { 1433 if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) { 1434 ALOGE("This stream only contains subtitles."); 1435 return ERROR_MALFORMED; 1436 } 1437 1438 const sp<AnotherPacketSource> packetSource = 1439 mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); 1440 1441 int64_t durationUs; 1442 CHECK(itemMeta->findInt64("durationUs", &durationUs)); 1443 buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber)); 1444 buffer->meta()->setInt64("durationUs", durationUs); 1445 buffer->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber)); 1446 buffer->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq); 1447 buffer->meta()->setInt32("subtitleGeneration", mSubtitleGeneration); 1448 1449 packetSource->queueAccessUnit(buffer); 1450 return OK; 1451 } 1452 1453 if (mNextPTSTimeUs >= 0ll) { 1454 mFirstPTSValid = false; 1455 mAbsoluteTimeAnchorUs = mNextPTSTimeUs; 1456 mNextPTSTimeUs = -1ll; 1457 } 1458 1459 // This better be an ISO 13818-7 (AAC) or ISO 13818-1 (MPEG) audio 1460 // stream prefixed by an ID3 tag. 1461 1462 bool firstID3Tag = true; 1463 uint64_t PTS = 0; 1464 1465 for (;;) { 1466 // Make sure to skip all ID3 tags preceding the audio data. 1467 // At least one must be present to provide the PTS timestamp. 1468 1469 ID3 id3(buffer->data(), buffer->size(), true /* ignoreV1 */); 1470 if (!id3.isValid()) { 1471 if (firstID3Tag) { 1472 ALOGE("Unable to parse ID3 tag."); 1473 return ERROR_MALFORMED; 1474 } else { 1475 break; 1476 } 1477 } 1478 1479 if (firstID3Tag) { 1480 bool found = false; 1481 1482 ID3::Iterator it(id3, "PRIV"); 1483 while (!it.done()) { 1484 size_t length; 1485 const uint8_t *data = it.getData(&length); 1486 1487 static const char *kMatchName = 1488 "com.apple.streaming.transportStreamTimestamp"; 1489 static const size_t kMatchNameLen = strlen(kMatchName); 1490 1491 if (length == kMatchNameLen + 1 + 8 1492 && !strncmp((const char *)data, kMatchName, kMatchNameLen)) { 1493 found = true; 1494 PTS = U64_AT(&data[kMatchNameLen + 1]); 1495 } 1496 1497 it.next(); 1498 } 1499 1500 if (!found) { 1501 ALOGE("Unable to extract transportStreamTimestamp from ID3 tag."); 1502 return ERROR_MALFORMED; 1503 } 1504 } 1505 1506 // skip the ID3 tag 1507 buffer->setRange( 1508 buffer->offset() + id3.rawSize(), buffer->size() - id3.rawSize()); 1509 1510 firstID3Tag = false; 1511 } 1512 1513 if (mStreamTypeMask != LiveSession::STREAMTYPE_AUDIO) { 1514 ALOGW("This stream only contains audio data!"); 1515 1516 mStreamTypeMask &= LiveSession::STREAMTYPE_AUDIO; 1517 1518 if (mStreamTypeMask == 0) { 1519 return OK; 1520 } 1521 } 1522 1523 sp<AnotherPacketSource> packetSource = 1524 mPacketSources.valueFor(LiveSession::STREAMTYPE_AUDIO); 1525 1526 if (packetSource->getFormat() == NULL && buffer->size() >= 7) { 1527 ABitReader bits(buffer->data(), buffer->size()); 1528 1529 // adts_fixed_header 1530 1531 CHECK_EQ(bits.getBits(12), 0xfffu); 1532 bits.skipBits(3); // ID, layer 1533 bool protection_absent = bits.getBits(1) != 0; 1534 1535 unsigned profile = bits.getBits(2); 1536 CHECK_NE(profile, 3u); 1537 unsigned sampling_freq_index = bits.getBits(4); 1538 bits.getBits(1); // private_bit 1539 unsigned channel_configuration = bits.getBits(3); 1540 CHECK_NE(channel_configuration, 0u); 1541 bits.skipBits(2); // original_copy, home 1542 1543 sp<MetaData> meta = MakeAACCodecSpecificData( 1544 profile, sampling_freq_index, channel_configuration); 1545 1546 meta->setInt32(kKeyIsADTS, true); 1547 1548 packetSource->setFormat(meta); 1549 } 1550 1551 int64_t numSamples = 0ll; 1552 int32_t sampleRate; 1553 CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); 1554 1555 int64_t timeUs = (PTS * 100ll) / 9ll; 1556 if (!mFirstPTSValid) { 1557 mFirstPTSValid = true; 1558 mFirstTimeUs = timeUs; 1559 } 1560 1561 size_t offset = 0; 1562 while (offset < buffer->size()) { 1563 const uint8_t *adtsHeader = buffer->data() + offset; 1564 CHECK_LT(offset + 5, buffer->size()); 1565 1566 unsigned aac_frame_length = 1567 ((adtsHeader[3] & 3) << 11) 1568 | (adtsHeader[4] << 3) 1569 | (adtsHeader[5] >> 5); 1570 1571 if (aac_frame_length == 0) { 1572 const uint8_t *id3Header = adtsHeader; 1573 if (!memcmp(id3Header, "ID3", 3)) { 1574 ID3 id3(id3Header, buffer->size() - offset, true); 1575 if (id3.isValid()) { 1576 offset += id3.rawSize(); 1577 continue; 1578 }; 1579 } 1580 return ERROR_MALFORMED; 1581 } 1582 1583 CHECK_LE(offset + aac_frame_length, buffer->size()); 1584 1585 int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate; 1586 offset += aac_frame_length; 1587 1588 // Each AAC frame encodes 1024 samples. 1589 numSamples += 1024; 1590 1591 if (mStartup) { 1592 int64_t startTimeUs = unitTimeUs; 1593 if (mStartTimeUsRelative) { 1594 startTimeUs -= mFirstTimeUs; 1595 if (startTimeUs < 0) { 1596 startTimeUs = 0; 1597 } 1598 } 1599 if (startTimeUs < mStartTimeUs) { 1600 continue; 1601 } 1602 1603 if (mStartTimeUsNotify != NULL) { 1604 int32_t targetDurationSecs; 1605 CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); 1606 int64_t targetDurationUs = targetDurationSecs * 1000000ll; 1607 1608 // Duplicated logic from how we handle .ts playlists. 1609 if (mStartup && mSegmentStartTimeUs >= 0 1610 && timeUs - mStartTimeUs > targetDurationUs) { 1611 int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs); 1612 if (newSeqNumber >= mSeqNumber) { 1613 --mSeqNumber; 1614 } else { 1615 mSeqNumber = newSeqNumber; 1616 } 1617 return -EAGAIN; 1618 } 1619 1620 mStartTimeUsNotify->setInt64("timeUsAudio", timeUs); 1621 mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq); 1622 mStartTimeUsNotify->setInt32("streamMask", LiveSession::STREAMTYPE_AUDIO); 1623 mStartTimeUsNotify->post(); 1624 mStartTimeUsNotify.clear(); 1625 mStartup = false; 1626 } 1627 } 1628 1629 if (mStopParams != NULL) { 1630 // Queue discontinuity in original stream. 1631 int32_t discontinuitySeq; 1632 int64_t stopTimeUs; 1633 if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq) 1634 || discontinuitySeq > mDiscontinuitySeq 1635 || !mStopParams->findInt64("timeUsAudio", &stopTimeUs) 1636 || (discontinuitySeq == mDiscontinuitySeq && unitTimeUs >= stopTimeUs)) { 1637 packetSource->queueAccessUnit(mSession->createFormatChangeBuffer()); 1638 mStreamTypeMask = 0; 1639 mPacketSources.clear(); 1640 return ERROR_OUT_OF_RANGE; 1641 } 1642 } 1643 1644 sp<ABuffer> unit = new ABuffer(aac_frame_length); 1645 memcpy(unit->data(), adtsHeader, aac_frame_length); 1646 1647 unit->meta()->setInt64("timeUs", unitTimeUs); 1648 setAccessUnitProperties(unit, packetSource); 1649 packetSource->queueAccessUnit(unit); 1650 } 1651 1652 return OK; 1653} 1654 1655void PlaylistFetcher::updateDuration() { 1656 int64_t durationUs = 0ll; 1657 for (size_t index = 0; index < mPlaylist->size(); ++index) { 1658 sp<AMessage> itemMeta; 1659 CHECK(mPlaylist->itemAt( 1660 index, NULL /* uri */, &itemMeta)); 1661 1662 int64_t itemDurationUs; 1663 CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); 1664 1665 durationUs += itemDurationUs; 1666 } 1667 1668 sp<AMessage> msg = mNotify->dup(); 1669 msg->setInt32("what", kWhatDurationUpdate); 1670 msg->setInt64("durationUs", durationUs); 1671 msg->post(); 1672} 1673 1674int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) { 1675 int64_t durationUs, threshold; 1676 if (msg->findInt64("durationUs", &durationUs)) { 1677 return kNumSkipFrames * durationUs; 1678 } 1679 1680 sp<RefBase> obj; 1681 msg->findObject("format", &obj); 1682 MetaData *format = static_cast<MetaData *>(obj.get()); 1683 1684 const char *mime; 1685 CHECK(format->findCString(kKeyMIMEType, &mime)); 1686 bool audio = !strncasecmp(mime, "audio/", 6); 1687 if (audio) { 1688 // Assumes 1000 samples per frame. 1689 int32_t sampleRate; 1690 CHECK(format->findInt32(kKeySampleRate, &sampleRate)); 1691 return kNumSkipFrames /* frames */ * 1000 /* samples */ 1692 * (1000000 / sampleRate) /* sample duration (us) */; 1693 } else { 1694 int32_t frameRate; 1695 if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) { 1696 return kNumSkipFrames * (1000000 / frameRate); 1697 } 1698 } 1699 1700 return 500000ll; 1701} 1702 1703} // namespace android 1704