1 /* //device/extlibs/pv/android/AudioTrack.cpp 2 ** 3 ** Copyright 2007, The Android Open Source Project 4 ** 5 ** Licensed under the Apache License, Version 2.0 (the "License"); 6 ** you may not use this file except in compliance with the License. 7 ** You may obtain a copy of the License at 8 ** 9 ** http://www.apache.org/licenses/LICENSE-2.0 10 ** 11 ** Unless required by applicable law or agreed to in writing, software 12 ** distributed under the License is distributed on an "AS IS" BASIS, 13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14 ** See the License for the specific language governing permissions and 15 ** limitations under the License. 16 */ 17 18 19 //#define LOG_NDEBUG 0 20 #define LOG_TAG "AudioTrack" 21 22 #include <stdint.h> 23 #include <sys/types.h> 24 #include <limits.h> 25 26 #include <sched.h> 27 #include <sys/resource.h> 28 29 #include <private/media/AudioTrackShared.h> 30 31 #include <media/AudioSystem.h> 32 #include <media/AudioTrack.h> 33 34 #include <utils/Log.h> 35 #include <binder/Parcel.h> 36 #include <binder/IPCThreadState.h> 37 #include <utils/Timers.h> 38 #include <utils/Atomic.h> 39 40 #include <cutils/bitops.h> 41 42 #include <system/audio.h> 43 #include <system/audio_policy.h> 44 45 #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) 46 #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) 47 48 namespace android { 49 // --------------------------------------------------------------------------- 50 51 // static 52 status_t AudioTrack::getMinFrameCount( 53 int* frameCount, 54 int streamType, 55 uint32_t sampleRate) 56 { 57 int afSampleRate; 58 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 59 return NO_INIT; 60 } 61 int afFrameCount; 62 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 63 return NO_INIT; 64 } 65 uint32_t afLatency; 66 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 67 return NO_INIT; 68 } 69 70 // Ensure that buffer depth covers at least audio hardware latency 71 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 72 if (minBufCount < 2) minBufCount = 2; 73 74 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 75 afFrameCount * minBufCount * sampleRate / afSampleRate; 76 return NO_ERROR; 77 } 78 79 // --------------------------------------------------------------------------- 80 81 AudioTrack::AudioTrack() 82 : mStatus(NO_INIT) 83 { 84 } 85 86 AudioTrack::AudioTrack( 87 int streamType, 88 uint32_t sampleRate, 89 int format, 90 int channelMask, 91 int frameCount, 92 uint32_t flags, 93 callback_t cbf, 94 void* user, 95 int notificationFrames, 96 int sessionId) 97 : mStatus(NO_INIT) 98 { 99 mStatus = set(streamType, sampleRate, format, channelMask, 100 frameCount, flags, cbf, user, notificationFrames, 101 0, false, sessionId); 102 } 103 104 AudioTrack::AudioTrack( 105 int streamType, 106 uint32_t sampleRate, 107 int format, 108 int channelMask, 109 const sp<IMemory>& sharedBuffer, 110 uint32_t flags, 111 callback_t cbf, 112 void* user, 113 int notificationFrames, 114 int sessionId) 115 : mStatus(NO_INIT) 116 { 117 mStatus = set(streamType, sampleRate, format, channelMask, 118 0, flags, cbf, user, notificationFrames, 119 sharedBuffer, false, sessionId); 120 } 121 122 AudioTrack::~AudioTrack() 123 { 124 LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 125 126 if (mStatus == NO_ERROR) { 127 // Make sure that callback function exits in the case where 128 // it is looping on buffer full condition in obtainBuffer(). 129 // Otherwise the callback thread will never exit. 130 stop(); 131 if (mAudioTrackThread != 0) { 132 mAudioTrackThread->requestExitAndWait(); 133 mAudioTrackThread.clear(); 134 } 135 mAudioTrack.clear(); 136 IPCThreadState::self()->flushCommands(); 137 AudioSystem::releaseAudioSessionId(mSessionId); 138 } 139 } 140 141 status_t AudioTrack::set( 142 int streamType, 143 uint32_t sampleRate, 144 int format, 145 int channelMask, 146 int frameCount, 147 uint32_t flags, 148 callback_t cbf, 149 void* user, 150 int notificationFrames, 151 const sp<IMemory>& sharedBuffer, 152 bool threadCanCallJava, 153 int sessionId) 154 { 155 156 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 157 158 AutoMutex lock(mLock); 159 if (mAudioTrack != 0) { 160 LOGE("Track already in use"); 161 return INVALID_OPERATION; 162 } 163 164 int afSampleRate; 165 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 166 return NO_INIT; 167 } 168 uint32_t afLatency; 169 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 170 return NO_INIT; 171 } 172 173 // handle default values first. 174 if (streamType == AUDIO_STREAM_DEFAULT) { 175 streamType = AUDIO_STREAM_MUSIC; 176 } 177 if (sampleRate == 0) { 178 sampleRate = afSampleRate; 179 } 180 // these below should probably come from the audioFlinger too... 181 if (format == 0) { 182 format = AUDIO_FORMAT_PCM_16_BIT; 183 } 184 if (channelMask == 0) { 185 channelMask = AUDIO_CHANNEL_OUT_STEREO; 186 } 187 188 // validate parameters 189 if (!audio_is_valid_format(format)) { 190 LOGE("Invalid format"); 191 return BAD_VALUE; 192 } 193 194 // force direct flag if format is not linear PCM 195 if (!audio_is_linear_pcm(format)) { 196 flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT; 197 } 198 199 if (!audio_is_output_channel(channelMask)) { 200 LOGE("Invalid channel mask"); 201 return BAD_VALUE; 202 } 203 uint32_t channelCount = popcount(channelMask); 204 205 audio_io_handle_t output = AudioSystem::getOutput( 206 (audio_stream_type_t)streamType, 207 sampleRate,format, channelMask, 208 (audio_policy_output_flags_t)flags); 209 210 if (output == 0) { 211 LOGE("Could not get audio output for stream type %d", streamType); 212 return BAD_VALUE; 213 } 214 215 mVolume[LEFT] = 1.0f; 216 mVolume[RIGHT] = 1.0f; 217 mSendLevel = 0; 218 mFrameCount = frameCount; 219 mNotificationFramesReq = notificationFrames; 220 mSessionId = sessionId; 221 mAuxEffectId = 0; 222 223 // create the IAudioTrack 224 status_t status = createTrack_l(streamType, 225 sampleRate, 226 (uint32_t)format, 227 (uint32_t)channelMask, 228 frameCount, 229 flags, 230 sharedBuffer, 231 output, 232 true); 233 234 if (status != NO_ERROR) { 235 return status; 236 } 237 238 if (cbf != 0) { 239 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 240 if (mAudioTrackThread == 0) { 241 LOGE("Could not create callback thread"); 242 return NO_INIT; 243 } 244 } 245 246 mStatus = NO_ERROR; 247 248 mStreamType = streamType; 249 mFormat = (uint32_t)format; 250 mChannelMask = (uint32_t)channelMask; 251 mChannelCount = channelCount; 252 mSharedBuffer = sharedBuffer; 253 mMuted = false; 254 mActive = 0; 255 mCbf = cbf; 256 mUserData = user; 257 mLoopCount = 0; 258 mMarkerPosition = 0; 259 mMarkerReached = false; 260 mNewPosition = 0; 261 mUpdatePeriod = 0; 262 mFlushed = false; 263 mFlags = flags; 264 AudioSystem::acquireAudioSessionId(mSessionId); 265 mRestoreStatus = NO_ERROR; 266 return NO_ERROR; 267 } 268 269 status_t AudioTrack::initCheck() const 270 { 271 return mStatus; 272 } 273 274 // ------------------------------------------------------------------------- 275 276 uint32_t AudioTrack::latency() const 277 { 278 return mLatency; 279 } 280 281 int AudioTrack::streamType() const 282 { 283 return mStreamType; 284 } 285 286 int AudioTrack::format() const 287 { 288 return mFormat; 289 } 290 291 int AudioTrack::channelCount() const 292 { 293 return mChannelCount; 294 } 295 296 uint32_t AudioTrack::frameCount() const 297 { 298 return mCblk->frameCount; 299 } 300 301 int AudioTrack::frameSize() const 302 { 303 if (audio_is_linear_pcm(mFormat)) { 304 return channelCount()*audio_bytes_per_sample(mFormat); 305 } else { 306 return sizeof(uint8_t); 307 } 308 } 309 310 sp<IMemory>& AudioTrack::sharedBuffer() 311 { 312 return mSharedBuffer; 313 } 314 315 // ------------------------------------------------------------------------- 316 317 void AudioTrack::start() 318 { 319 sp<AudioTrackThread> t = mAudioTrackThread; 320 status_t status = NO_ERROR; 321 322 LOGV("start %p", this); 323 if (t != 0) { 324 if (t->exitPending()) { 325 if (t->requestExitAndWait() == WOULD_BLOCK) { 326 LOGE("AudioTrack::start called from thread"); 327 return; 328 } 329 } 330 t->mLock.lock(); 331 } 332 333 AutoMutex lock(mLock); 334 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 335 // while we are accessing the cblk 336 sp <IAudioTrack> audioTrack = mAudioTrack; 337 sp <IMemory> iMem = mCblkMemory; 338 audio_track_cblk_t* cblk = mCblk; 339 340 if (mActive == 0) { 341 mFlushed = false; 342 mActive = 1; 343 mNewPosition = cblk->server + mUpdatePeriod; 344 cblk->lock.lock(); 345 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 346 cblk->waitTimeMs = 0; 347 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 348 if (t != 0) { 349 t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO); 350 } else { 351 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO); 352 } 353 354 LOGV("start %p before lock cblk %p", this, mCblk); 355 if (!(cblk->flags & CBLK_INVALID_MSK)) { 356 cblk->lock.unlock(); 357 status = mAudioTrack->start(); 358 cblk->lock.lock(); 359 if (status == DEAD_OBJECT) { 360 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 361 } 362 } 363 if (cblk->flags & CBLK_INVALID_MSK) { 364 status = restoreTrack_l(cblk, true); 365 } 366 cblk->lock.unlock(); 367 if (status != NO_ERROR) { 368 LOGV("start() failed"); 369 mActive = 0; 370 if (t != 0) { 371 t->requestExit(); 372 } else { 373 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); 374 } 375 } 376 } 377 378 if (t != 0) { 379 t->mLock.unlock(); 380 } 381 } 382 383 void AudioTrack::stop() 384 { 385 sp<AudioTrackThread> t = mAudioTrackThread; 386 387 LOGV("stop %p", this); 388 if (t != 0) { 389 t->mLock.lock(); 390 } 391 392 AutoMutex lock(mLock); 393 if (mActive == 1) { 394 mActive = 0; 395 mCblk->cv.signal(); 396 mAudioTrack->stop(); 397 // Cancel loops (If we are in the middle of a loop, playback 398 // would not stop until loopCount reaches 0). 399 setLoop_l(0, 0, 0); 400 // the playback head position will reset to 0, so if a marker is set, we need 401 // to activate it again 402 mMarkerReached = false; 403 // Force flush if a shared buffer is used otherwise audioflinger 404 // will not stop before end of buffer is reached. 405 if (mSharedBuffer != 0) { 406 flush_l(); 407 } 408 if (t != 0) { 409 t->requestExit(); 410 } else { 411 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); 412 } 413 } 414 415 if (t != 0) { 416 t->mLock.unlock(); 417 } 418 } 419 420 bool AudioTrack::stopped() const 421 { 422 return !mActive; 423 } 424 425 void AudioTrack::flush() 426 { 427 AutoMutex lock(mLock); 428 flush_l(); 429 } 430 431 // must be called with mLock held 432 void AudioTrack::flush_l() 433 { 434 LOGV("flush"); 435 436 // clear playback marker and periodic update counter 437 mMarkerPosition = 0; 438 mMarkerReached = false; 439 mUpdatePeriod = 0; 440 441 if (!mActive) { 442 mFlushed = true; 443 mAudioTrack->flush(); 444 // Release AudioTrack callback thread in case it was waiting for new buffers 445 // in AudioTrack::obtainBuffer() 446 mCblk->cv.signal(); 447 } 448 } 449 450 void AudioTrack::pause() 451 { 452 LOGV("pause"); 453 AutoMutex lock(mLock); 454 if (mActive == 1) { 455 mActive = 0; 456 mAudioTrack->pause(); 457 } 458 } 459 460 void AudioTrack::mute(bool e) 461 { 462 mAudioTrack->mute(e); 463 mMuted = e; 464 } 465 466 bool AudioTrack::muted() const 467 { 468 return mMuted; 469 } 470 471 status_t AudioTrack::setVolume(float left, float right) 472 { 473 if (left > 1.0f || right > 1.0f) { 474 return BAD_VALUE; 475 } 476 477 AutoMutex lock(mLock); 478 mVolume[LEFT] = left; 479 mVolume[RIGHT] = right; 480 481 // write must be atomic 482 mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000); 483 484 return NO_ERROR; 485 } 486 487 void AudioTrack::getVolume(float* left, float* right) 488 { 489 if (left != NULL) { 490 *left = mVolume[LEFT]; 491 } 492 if (right != NULL) { 493 *right = mVolume[RIGHT]; 494 } 495 } 496 497 status_t AudioTrack::setAuxEffectSendLevel(float level) 498 { 499 LOGV("setAuxEffectSendLevel(%f)", level); 500 if (level > 1.0f) { 501 return BAD_VALUE; 502 } 503 AutoMutex lock(mLock); 504 505 mSendLevel = level; 506 507 mCblk->sendLevel = uint16_t(level * 0x1000); 508 509 return NO_ERROR; 510 } 511 512 void AudioTrack::getAuxEffectSendLevel(float* level) 513 { 514 if (level != NULL) { 515 *level = mSendLevel; 516 } 517 } 518 519 status_t AudioTrack::setSampleRate(int rate) 520 { 521 int afSamplingRate; 522 523 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 524 return NO_INIT; 525 } 526 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 527 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 528 529 AutoMutex lock(mLock); 530 mCblk->sampleRate = rate; 531 return NO_ERROR; 532 } 533 534 uint32_t AudioTrack::getSampleRate() 535 { 536 AutoMutex lock(mLock); 537 return mCblk->sampleRate; 538 } 539 540 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 541 { 542 AutoMutex lock(mLock); 543 return setLoop_l(loopStart, loopEnd, loopCount); 544 } 545 546 // must be called with mLock held 547 status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 548 { 549 audio_track_cblk_t* cblk = mCblk; 550 551 Mutex::Autolock _l(cblk->lock); 552 553 if (loopCount == 0) { 554 cblk->loopStart = UINT_MAX; 555 cblk->loopEnd = UINT_MAX; 556 cblk->loopCount = 0; 557 mLoopCount = 0; 558 return NO_ERROR; 559 } 560 561 if (loopStart >= loopEnd || 562 loopEnd - loopStart > cblk->frameCount || 563 cblk->server > loopStart) { 564 LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 565 return BAD_VALUE; 566 } 567 568 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 569 LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 570 loopStart, loopEnd, cblk->frameCount); 571 return BAD_VALUE; 572 } 573 574 cblk->loopStart = loopStart; 575 cblk->loopEnd = loopEnd; 576 cblk->loopCount = loopCount; 577 mLoopCount = loopCount; 578 579 return NO_ERROR; 580 } 581 582 status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) 583 { 584 AutoMutex lock(mLock); 585 if (loopStart != 0) { 586 *loopStart = mCblk->loopStart; 587 } 588 if (loopEnd != 0) { 589 *loopEnd = mCblk->loopEnd; 590 } 591 if (loopCount != 0) { 592 if (mCblk->loopCount < 0) { 593 *loopCount = -1; 594 } else { 595 *loopCount = mCblk->loopCount; 596 } 597 } 598 599 return NO_ERROR; 600 } 601 602 status_t AudioTrack::setMarkerPosition(uint32_t marker) 603 { 604 if (mCbf == 0) return INVALID_OPERATION; 605 606 mMarkerPosition = marker; 607 mMarkerReached = false; 608 609 return NO_ERROR; 610 } 611 612 status_t AudioTrack::getMarkerPosition(uint32_t *marker) 613 { 614 if (marker == 0) return BAD_VALUE; 615 616 *marker = mMarkerPosition; 617 618 return NO_ERROR; 619 } 620 621 status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 622 { 623 if (mCbf == 0) return INVALID_OPERATION; 624 625 uint32_t curPosition; 626 getPosition(&curPosition); 627 mNewPosition = curPosition + updatePeriod; 628 mUpdatePeriod = updatePeriod; 629 630 return NO_ERROR; 631 } 632 633 status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) 634 { 635 if (updatePeriod == 0) return BAD_VALUE; 636 637 *updatePeriod = mUpdatePeriod; 638 639 return NO_ERROR; 640 } 641 642 status_t AudioTrack::setPosition(uint32_t position) 643 { 644 AutoMutex lock(mLock); 645 Mutex::Autolock _l(mCblk->lock); 646 647 if (!stopped()) return INVALID_OPERATION; 648 649 if (position > mCblk->user) return BAD_VALUE; 650 651 mCblk->server = position; 652 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 653 654 return NO_ERROR; 655 } 656 657 status_t AudioTrack::getPosition(uint32_t *position) 658 { 659 if (position == 0) return BAD_VALUE; 660 AutoMutex lock(mLock); 661 *position = mFlushed ? 0 : mCblk->server; 662 663 return NO_ERROR; 664 } 665 666 status_t AudioTrack::reload() 667 { 668 AutoMutex lock(mLock); 669 670 if (!stopped()) return INVALID_OPERATION; 671 672 flush_l(); 673 674 mCblk->stepUser(mCblk->frameCount); 675 676 return NO_ERROR; 677 } 678 679 audio_io_handle_t AudioTrack::getOutput() 680 { 681 AutoMutex lock(mLock); 682 return getOutput_l(); 683 } 684 685 // must be called with mLock held 686 audio_io_handle_t AudioTrack::getOutput_l() 687 { 688 return AudioSystem::getOutput((audio_stream_type_t)mStreamType, 689 mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags); 690 } 691 692 int AudioTrack::getSessionId() 693 { 694 return mSessionId; 695 } 696 697 status_t AudioTrack::attachAuxEffect(int effectId) 698 { 699 LOGV("attachAuxEffect(%d)", effectId); 700 status_t status = mAudioTrack->attachAuxEffect(effectId); 701 if (status == NO_ERROR) { 702 mAuxEffectId = effectId; 703 } 704 return status; 705 } 706 707 // ------------------------------------------------------------------------- 708 709 // must be called with mLock held 710 status_t AudioTrack::createTrack_l( 711 int streamType, 712 uint32_t sampleRate, 713 uint32_t format, 714 uint32_t channelMask, 715 int frameCount, 716 uint32_t flags, 717 const sp<IMemory>& sharedBuffer, 718 audio_io_handle_t output, 719 bool enforceFrameCount) 720 { 721 status_t status; 722 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 723 if (audioFlinger == 0) { 724 LOGE("Could not get audioflinger"); 725 return NO_INIT; 726 } 727 728 int afSampleRate; 729 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 730 return NO_INIT; 731 } 732 int afFrameCount; 733 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 734 return NO_INIT; 735 } 736 uint32_t afLatency; 737 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 738 return NO_INIT; 739 } 740 741 mNotificationFramesAct = mNotificationFramesReq; 742 if (!audio_is_linear_pcm(format)) { 743 if (sharedBuffer != 0) { 744 frameCount = sharedBuffer->size(); 745 } 746 } else { 747 // Ensure that buffer depth covers at least audio hardware latency 748 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 749 if (minBufCount < 2) minBufCount = 2; 750 751 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 752 753 if (sharedBuffer == 0) { 754 if (frameCount == 0) { 755 frameCount = minFrameCount; 756 } 757 if (mNotificationFramesAct == 0) { 758 mNotificationFramesAct = frameCount/2; 759 } 760 // Make sure that application is notified with sufficient margin 761 // before underrun 762 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 763 mNotificationFramesAct = frameCount/2; 764 } 765 if (frameCount < minFrameCount) { 766 if (enforceFrameCount) { 767 LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); 768 return BAD_VALUE; 769 } else { 770 frameCount = minFrameCount; 771 } 772 } 773 } else { 774 // Ensure that buffer alignment matches channelcount 775 int channelCount = popcount(channelMask); 776 if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { 777 LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); 778 return BAD_VALUE; 779 } 780 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 781 } 782 } 783 784 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 785 streamType, 786 sampleRate, 787 format, 788 channelMask, 789 frameCount, 790 ((uint16_t)flags) << 16, 791 sharedBuffer, 792 output, 793 &mSessionId, 794 &status); 795 796 if (track == 0) { 797 LOGE("AudioFlinger could not create track, status: %d", status); 798 return status; 799 } 800 sp<IMemory> cblk = track->getCblk(); 801 if (cblk == 0) { 802 LOGE("Could not get control block"); 803 return NO_INIT; 804 } 805 mAudioTrack.clear(); 806 mAudioTrack = track; 807 mCblkMemory.clear(); 808 mCblkMemory = cblk; 809 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 810 android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 811 if (sharedBuffer == 0) { 812 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 813 } else { 814 mCblk->buffers = sharedBuffer->pointer(); 815 // Force buffer full condition as data is already present in shared memory 816 mCblk->stepUser(mCblk->frameCount); 817 } 818 819 mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000); 820 mCblk->sendLevel = uint16_t(mSendLevel * 0x1000); 821 mAudioTrack->attachAuxEffect(mAuxEffectId); 822 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 823 mCblk->waitTimeMs = 0; 824 mRemainingFrames = mNotificationFramesAct; 825 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 826 return NO_ERROR; 827 } 828 829 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 830 { 831 AutoMutex lock(mLock); 832 int active; 833 status_t result = NO_ERROR; 834 audio_track_cblk_t* cblk = mCblk; 835 uint32_t framesReq = audioBuffer->frameCount; 836 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 837 838 audioBuffer->frameCount = 0; 839 audioBuffer->size = 0; 840 841 uint32_t framesAvail = cblk->framesAvailable(); 842 843 cblk->lock.lock(); 844 if (cblk->flags & CBLK_INVALID_MSK) { 845 goto create_new_track; 846 } 847 cblk->lock.unlock(); 848 849 if (framesAvail == 0) { 850 cblk->lock.lock(); 851 goto start_loop_here; 852 while (framesAvail == 0) { 853 active = mActive; 854 if (UNLIKELY(!active)) { 855 LOGV("Not active and NO_MORE_BUFFERS"); 856 cblk->lock.unlock(); 857 return NO_MORE_BUFFERS; 858 } 859 if (UNLIKELY(!waitCount)) { 860 cblk->lock.unlock(); 861 return WOULD_BLOCK; 862 } 863 if (!(cblk->flags & CBLK_INVALID_MSK)) { 864 mLock.unlock(); 865 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 866 cblk->lock.unlock(); 867 mLock.lock(); 868 if (mActive == 0) { 869 return status_t(STOPPED); 870 } 871 cblk->lock.lock(); 872 } 873 874 if (cblk->flags & CBLK_INVALID_MSK) { 875 goto create_new_track; 876 } 877 if (__builtin_expect(result!=NO_ERROR, false)) { 878 cblk->waitTimeMs += waitTimeMs; 879 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 880 // timing out when a loop has been set and we have already written upto loop end 881 // is a normal condition: no need to wake AudioFlinger up. 882 if (cblk->user < cblk->loopEnd) { 883 LOGW( "obtainBuffer timed out (is the CPU pegged?) %p " 884 "user=%08x, server=%08x", this, cblk->user, cblk->server); 885 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 886 cblk->lock.unlock(); 887 result = mAudioTrack->start(); 888 cblk->lock.lock(); 889 if (result == DEAD_OBJECT) { 890 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 891 create_new_track: 892 result = restoreTrack_l(cblk, false); 893 } 894 if (result != NO_ERROR) { 895 LOGW("obtainBuffer create Track error %d", result); 896 cblk->lock.unlock(); 897 return result; 898 } 899 } 900 cblk->waitTimeMs = 0; 901 } 902 903 if (--waitCount == 0) { 904 cblk->lock.unlock(); 905 return TIMED_OUT; 906 } 907 } 908 // read the server count again 909 start_loop_here: 910 framesAvail = cblk->framesAvailable_l(); 911 } 912 cblk->lock.unlock(); 913 } 914 915 // restart track if it was disabled by audioflinger due to previous underrun 916 if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) { 917 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 918 LOGW("obtainBuffer() track %p disabled, restarting", this); 919 mAudioTrack->start(); 920 } 921 922 cblk->waitTimeMs = 0; 923 924 if (framesReq > framesAvail) { 925 framesReq = framesAvail; 926 } 927 928 uint32_t u = cblk->user; 929 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 930 931 if (u + framesReq > bufferEnd) { 932 framesReq = bufferEnd - u; 933 } 934 935 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 936 audioBuffer->channelCount = mChannelCount; 937 audioBuffer->frameCount = framesReq; 938 audioBuffer->size = framesReq * cblk->frameSize; 939 if (audio_is_linear_pcm(mFormat)) { 940 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 941 } else { 942 audioBuffer->format = mFormat; 943 } 944 audioBuffer->raw = (int8_t *)cblk->buffer(u); 945 active = mActive; 946 return active ? status_t(NO_ERROR) : status_t(STOPPED); 947 } 948 949 void AudioTrack::releaseBuffer(Buffer* audioBuffer) 950 { 951 AutoMutex lock(mLock); 952 mCblk->stepUser(audioBuffer->frameCount); 953 } 954 955 // ------------------------------------------------------------------------- 956 957 ssize_t AudioTrack::write(const void* buffer, size_t userSize) 958 { 959 960 if (mSharedBuffer != 0) return INVALID_OPERATION; 961 962 if (ssize_t(userSize) < 0) { 963 // sanity-check. user is most-likely passing an error code. 964 LOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 965 buffer, userSize, userSize); 966 return BAD_VALUE; 967 } 968 969 LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 970 971 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 972 // while we are accessing the cblk 973 mLock.lock(); 974 sp <IAudioTrack> audioTrack = mAudioTrack; 975 sp <IMemory> iMem = mCblkMemory; 976 mLock.unlock(); 977 978 ssize_t written = 0; 979 const int8_t *src = (const int8_t *)buffer; 980 Buffer audioBuffer; 981 size_t frameSz = (size_t)frameSize(); 982 983 do { 984 audioBuffer.frameCount = userSize/frameSz; 985 986 // Calling obtainBuffer() with a negative wait count causes 987 // an (almost) infinite wait time. 988 status_t err = obtainBuffer(&audioBuffer, -1); 989 if (err < 0) { 990 // out of buffers, return #bytes written 991 if (err == status_t(NO_MORE_BUFFERS)) 992 break; 993 return ssize_t(err); 994 } 995 996 size_t toWrite; 997 998 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 999 // Divide capacity by 2 to take expansion into account 1000 toWrite = audioBuffer.size>>1; 1001 // 8 to 16 bit conversion 1002 int count = toWrite; 1003 int16_t *dst = (int16_t *)(audioBuffer.i8); 1004 while(count--) { 1005 *dst++ = (int16_t)(*src++^0x80) << 8; 1006 } 1007 } else { 1008 toWrite = audioBuffer.size; 1009 memcpy(audioBuffer.i8, src, toWrite); 1010 src += toWrite; 1011 } 1012 userSize -= toWrite; 1013 written += toWrite; 1014 1015 releaseBuffer(&audioBuffer); 1016 } while (userSize >= frameSz); 1017 1018 return written; 1019 } 1020 1021 // ------------------------------------------------------------------------- 1022 1023 bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1024 { 1025 Buffer audioBuffer; 1026 uint32_t frames; 1027 size_t writtenSize; 1028 1029 mLock.lock(); 1030 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1031 // while we are accessing the cblk 1032 sp <IAudioTrack> audioTrack = mAudioTrack; 1033 sp <IMemory> iMem = mCblkMemory; 1034 audio_track_cblk_t* cblk = mCblk; 1035 mLock.unlock(); 1036 1037 // Manage underrun callback 1038 if (mActive && (cblk->framesAvailable() == cblk->frameCount)) { 1039 LOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1040 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1041 mCbf(EVENT_UNDERRUN, mUserData, 0); 1042 if (cblk->server == cblk->frameCount) { 1043 mCbf(EVENT_BUFFER_END, mUserData, 0); 1044 } 1045 if (mSharedBuffer != 0) return false; 1046 } 1047 } 1048 1049 // Manage loop end callback 1050 while (mLoopCount > cblk->loopCount) { 1051 int loopCount = -1; 1052 mLoopCount--; 1053 if (mLoopCount >= 0) loopCount = mLoopCount; 1054 1055 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1056 } 1057 1058 // Manage marker callback 1059 if (!mMarkerReached && (mMarkerPosition > 0)) { 1060 if (cblk->server >= mMarkerPosition) { 1061 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1062 mMarkerReached = true; 1063 } 1064 } 1065 1066 // Manage new position callback 1067 if (mUpdatePeriod > 0) { 1068 while (cblk->server >= mNewPosition) { 1069 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1070 mNewPosition += mUpdatePeriod; 1071 } 1072 } 1073 1074 // If Shared buffer is used, no data is requested from client. 1075 if (mSharedBuffer != 0) { 1076 frames = 0; 1077 } else { 1078 frames = mRemainingFrames; 1079 } 1080 1081 int32_t waitCount = -1; 1082 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1083 waitCount = 1; 1084 } 1085 1086 do { 1087 1088 audioBuffer.frameCount = frames; 1089 1090 // Calling obtainBuffer() with a wait count of 1 1091 // limits wait time to WAIT_PERIOD_MS. This prevents from being 1092 // stuck here not being able to handle timed events (position, markers, loops). 1093 status_t err = obtainBuffer(&audioBuffer, waitCount); 1094 if (err < NO_ERROR) { 1095 if (err != TIMED_OUT) { 1096 LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); 1097 return false; 1098 } 1099 break; 1100 } 1101 if (err == status_t(STOPPED)) return false; 1102 1103 // Divide buffer size by 2 to take into account the expansion 1104 // due to 8 to 16 bit conversion: the callback must fill only half 1105 // of the destination buffer 1106 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1107 audioBuffer.size >>= 1; 1108 } 1109 1110 size_t reqSize = audioBuffer.size; 1111 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1112 writtenSize = audioBuffer.size; 1113 1114 // Sanity check on returned size 1115 if (ssize_t(writtenSize) <= 0) { 1116 // The callback is done filling buffers 1117 // Keep this thread going to handle timed events and 1118 // still try to get more data in intervals of WAIT_PERIOD_MS 1119 // but don't just loop and block the CPU, so wait 1120 usleep(WAIT_PERIOD_MS*1000); 1121 break; 1122 } 1123 if (writtenSize > reqSize) writtenSize = reqSize; 1124 1125 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1126 // 8 to 16 bit conversion 1127 const int8_t *src = audioBuffer.i8 + writtenSize-1; 1128 int count = writtenSize; 1129 int16_t *dst = audioBuffer.i16 + writtenSize-1; 1130 while(count--) { 1131 *dst-- = (int16_t)(*src--^0x80) << 8; 1132 } 1133 writtenSize <<= 1; 1134 } 1135 1136 audioBuffer.size = writtenSize; 1137 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1138 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of 1139 // 16 bit. 1140 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1141 1142 frames -= audioBuffer.frameCount; 1143 1144 releaseBuffer(&audioBuffer); 1145 } 1146 while (frames); 1147 1148 if (frames == 0) { 1149 mRemainingFrames = mNotificationFramesAct; 1150 } else { 1151 mRemainingFrames = frames; 1152 } 1153 return true; 1154 } 1155 1156 // must be called with mLock and cblk.lock held. Callers must also hold strong references on 1157 // the IAudioTrack and IMemory in case they are recreated here. 1158 // If the IAudioTrack is successfully restored, the cblk pointer is updated 1159 status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1160 { 1161 status_t result; 1162 1163 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1164 LOGW("dead IAudioTrack, creating a new one from %s TID %d", 1165 fromStart ? "start()" : "obtainBuffer()", gettid()); 1166 1167 // signal old cblk condition so that other threads waiting for available buffers stop 1168 // waiting now 1169 cblk->cv.broadcast(); 1170 cblk->lock.unlock(); 1171 1172 // refresh the audio configuration cache in this process to make sure we get new 1173 // output parameters in getOutput_l() and createTrack_l() 1174 AudioSystem::clearAudioConfigCache(); 1175 1176 // if the new IAudioTrack is created, createTrack_l() will modify the 1177 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1178 // It will also delete the strong references on previous IAudioTrack and IMemory 1179 result = createTrack_l(mStreamType, 1180 cblk->sampleRate, 1181 mFormat, 1182 mChannelMask, 1183 mFrameCount, 1184 mFlags, 1185 mSharedBuffer, 1186 getOutput_l(), 1187 false); 1188 1189 if (result == NO_ERROR) { 1190 uint32_t user = cblk->user; 1191 uint32_t server = cblk->server; 1192 // restore write index and set other indexes to reflect empty buffer status 1193 mCblk->user = user; 1194 mCblk->server = user; 1195 mCblk->userBase = user; 1196 mCblk->serverBase = user; 1197 // restore loop: this is not guaranteed to succeed if new frame count is not 1198 // compatible with loop length 1199 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1200 if (!fromStart) { 1201 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1202 // Make sure that a client relying on callback events indicating underrun or 1203 // the actual amount of audio frames played (e.g SoundPool) receives them. 1204 if (mSharedBuffer == 0) { 1205 uint32_t frames = 0; 1206 if (user > server) { 1207 frames = ((user - server) > mCblk->frameCount) ? 1208 mCblk->frameCount : (user - server); 1209 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1210 } 1211 // restart playback even if buffer is not completely filled. 1212 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1213 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1214 // the client 1215 mCblk->stepUser(frames); 1216 } 1217 } 1218 if (mActive) { 1219 result = mAudioTrack->start(); 1220 LOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1221 } 1222 if (fromStart && result == NO_ERROR) { 1223 mNewPosition = mCblk->server + mUpdatePeriod; 1224 } 1225 } 1226 if (result != NO_ERROR) { 1227 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1228 LOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1229 } 1230 mRestoreStatus = result; 1231 // signal old cblk condition for other threads waiting for restore completion 1232 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1233 cblk->cv.broadcast(); 1234 } else { 1235 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1236 LOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1237 mLock.unlock(); 1238 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1239 if (result == NO_ERROR) { 1240 result = mRestoreStatus; 1241 } 1242 cblk->lock.unlock(); 1243 mLock.lock(); 1244 } else { 1245 LOGW("dead IAudioTrack, already restored TID %d", gettid()); 1246 result = mRestoreStatus; 1247 cblk->lock.unlock(); 1248 } 1249 } 1250 LOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1251 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1252 1253 if (result == NO_ERROR) { 1254 // from now on we switch to the newly created cblk 1255 cblk = mCblk; 1256 } 1257 cblk->lock.lock(); 1258 1259 LOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1260 1261 return result; 1262 } 1263 1264 status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1265 { 1266 1267 const size_t SIZE = 256; 1268 char buffer[SIZE]; 1269 String8 result; 1270 1271 result.append(" AudioTrack::dump\n"); 1272 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1273 result.append(buffer); 1274 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1275 result.append(buffer); 1276 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1277 result.append(buffer); 1278 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1279 result.append(buffer); 1280 ::write(fd, result.string(), result.size()); 1281 return NO_ERROR; 1282 } 1283 1284 // ========================================================================= 1285 1286 AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1287 : Thread(bCanCallJava), mReceiver(receiver) 1288 { 1289 } 1290 1291 bool AudioTrack::AudioTrackThread::threadLoop() 1292 { 1293 return mReceiver.processAudioBuffer(this); 1294 } 1295 1296 status_t AudioTrack::AudioTrackThread::readyToRun() 1297 { 1298 return NO_ERROR; 1299 } 1300 1301 void AudioTrack::AudioTrackThread::onFirstRef() 1302 { 1303 } 1304 1305 // ========================================================================= 1306 1307 1308 audio_track_cblk_t::audio_track_cblk_t() 1309 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1310 userBase(0), serverBase(0), buffers(0), frameCount(0), 1311 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), 1312 sendLevel(0), flags(0) 1313 { 1314 } 1315 1316 uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1317 { 1318 uint32_t u = this->user; 1319 1320 u += frameCount; 1321 // Ensure that user is never ahead of server for AudioRecord 1322 if (flags & CBLK_DIRECTION_MSK) { 1323 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1324 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1325 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1326 } 1327 } else if (u > this->server) { 1328 LOGW("stepServer occured after track reset"); 1329 u = this->server; 1330 } 1331 1332 if (u >= userBase + this->frameCount) { 1333 userBase += this->frameCount; 1334 } 1335 1336 this->user = u; 1337 1338 // Clear flow control error condition as new data has been written/read to/from buffer. 1339 if (flags & CBLK_UNDERRUN_MSK) { 1340 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1341 } 1342 1343 return u; 1344 } 1345 1346 bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1347 { 1348 if (!tryLock()) { 1349 LOGW("stepServer() could not lock cblk"); 1350 return false; 1351 } 1352 1353 uint32_t s = this->server; 1354 1355 s += frameCount; 1356 if (flags & CBLK_DIRECTION_MSK) { 1357 // Mark that we have read the first buffer so that next time stepUser() is called 1358 // we switch to normal obtainBuffer() timeout period 1359 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1360 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1361 } 1362 // It is possible that we receive a flush() 1363 // while the mixer is processing a block: in this case, 1364 // stepServer() is called After the flush() has reset u & s and 1365 // we have s > u 1366 if (s > this->user) { 1367 LOGW("stepServer occured after track reset"); 1368 s = this->user; 1369 } 1370 } 1371 1372 if (s >= loopEnd) { 1373 LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1374 s = loopStart; 1375 if (--loopCount == 0) { 1376 loopEnd = UINT_MAX; 1377 loopStart = UINT_MAX; 1378 } 1379 } 1380 if (s >= serverBase + this->frameCount) { 1381 serverBase += this->frameCount; 1382 } 1383 1384 this->server = s; 1385 1386 if (!(flags & CBLK_INVALID_MSK)) { 1387 cv.signal(); 1388 } 1389 lock.unlock(); 1390 return true; 1391 } 1392 1393 void* audio_track_cblk_t::buffer(uint32_t offset) const 1394 { 1395 return (int8_t *)this->buffers + (offset - userBase) * this->frameSize; 1396 } 1397 1398 uint32_t audio_track_cblk_t::framesAvailable() 1399 { 1400 Mutex::Autolock _l(lock); 1401 return framesAvailable_l(); 1402 } 1403 1404 uint32_t audio_track_cblk_t::framesAvailable_l() 1405 { 1406 uint32_t u = this->user; 1407 uint32_t s = this->server; 1408 1409 if (flags & CBLK_DIRECTION_MSK) { 1410 uint32_t limit = (s < loopStart) ? s : loopStart; 1411 return limit + frameCount - u; 1412 } else { 1413 return frameCount + u - s; 1414 } 1415 } 1416 1417 uint32_t audio_track_cblk_t::framesReady() 1418 { 1419 uint32_t u = this->user; 1420 uint32_t s = this->server; 1421 1422 if (flags & CBLK_DIRECTION_MSK) { 1423 if (u < loopEnd) { 1424 return u - s; 1425 } else { 1426 // do not block on mutex shared with client on AudioFlinger side 1427 if (!tryLock()) { 1428 LOGW("framesReady() could not lock cblk"); 1429 return 0; 1430 } 1431 uint32_t frames = UINT_MAX; 1432 if (loopCount >= 0) { 1433 frames = (loopEnd - loopStart)*loopCount + u - s; 1434 } 1435 lock.unlock(); 1436 return frames; 1437 } 1438 } else { 1439 return s - u; 1440 } 1441 } 1442 1443 bool audio_track_cblk_t::tryLock() 1444 { 1445 // the code below simulates lock-with-timeout 1446 // we MUST do this to protect the AudioFlinger server 1447 // as this lock is shared with the client. 1448 status_t err; 1449 1450 err = lock.tryLock(); 1451 if (err == -EBUSY) { // just wait a bit 1452 usleep(1000); 1453 err = lock.tryLock(); 1454 } 1455 if (err != NO_ERROR) { 1456 // probably, the client just died. 1457 return false; 1458 } 1459 return true; 1460 } 1461 1462 // ------------------------------------------------------------------------- 1463 1464 }; // namespace android 1465 1466
