AudioTrack.cpp revision c5ac4cb3a5124860ccfc7e4ff66251c55a5595ca
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 48namespace android { 49// --------------------------------------------------------------------------- 50 51// static 52status_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 81AudioTrack::AudioTrack() 82 : mStatus(NO_INIT), 83 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT) 84{ 85} 86 87AudioTrack::AudioTrack( 88 int streamType, 89 uint32_t sampleRate, 90 int format, 91 int channelMask, 92 int frameCount, 93 uint32_t flags, 94 callback_t cbf, 95 void* user, 96 int notificationFrames, 97 int sessionId) 98 : mStatus(NO_INIT), 99 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT) 100{ 101 mStatus = set(streamType, sampleRate, format, channelMask, 102 frameCount, flags, cbf, user, notificationFrames, 103 0, false, sessionId); 104} 105 106AudioTrack::AudioTrack( 107 int streamType, 108 uint32_t sampleRate, 109 int format, 110 int channelMask, 111 const sp<IMemory>& sharedBuffer, 112 uint32_t flags, 113 callback_t cbf, 114 void* user, 115 int notificationFrames, 116 int sessionId) 117 : mStatus(NO_INIT), 118 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT) 119{ 120 mStatus = set(streamType, sampleRate, format, channelMask, 121 0, flags, cbf, user, notificationFrames, 122 sharedBuffer, false, sessionId); 123} 124 125AudioTrack::~AudioTrack() 126{ 127 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 128 129 if (mStatus == NO_ERROR) { 130 // Make sure that callback function exits in the case where 131 // it is looping on buffer full condition in obtainBuffer(). 132 // Otherwise the callback thread will never exit. 133 stop(); 134 if (mAudioTrackThread != 0) { 135 mAudioTrackThread->requestExitAndWait(); 136 mAudioTrackThread.clear(); 137 } 138 mAudioTrack.clear(); 139 IPCThreadState::self()->flushCommands(); 140 AudioSystem::releaseAudioSessionId(mSessionId); 141 } 142} 143 144status_t AudioTrack::set( 145 int streamType, 146 uint32_t sampleRate, 147 int format, 148 int channelMask, 149 int frameCount, 150 uint32_t flags, 151 callback_t cbf, 152 void* user, 153 int notificationFrames, 154 const sp<IMemory>& sharedBuffer, 155 bool threadCanCallJava, 156 int sessionId) 157{ 158 159 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 160 161 AutoMutex lock(mLock); 162 if (mAudioTrack != 0) { 163 LOGE("Track already in use"); 164 return INVALID_OPERATION; 165 } 166 167 int afSampleRate; 168 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 169 return NO_INIT; 170 } 171 uint32_t afLatency; 172 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 173 return NO_INIT; 174 } 175 176 // handle default values first. 177 if (streamType == AUDIO_STREAM_DEFAULT) { 178 streamType = AUDIO_STREAM_MUSIC; 179 } 180 if (sampleRate == 0) { 181 sampleRate = afSampleRate; 182 } 183 // these below should probably come from the audioFlinger too... 184 if (format == 0) { 185 format = AUDIO_FORMAT_PCM_16_BIT; 186 } 187 if (channelMask == 0) { 188 channelMask = AUDIO_CHANNEL_OUT_STEREO; 189 } 190 191 // validate parameters 192 if (!audio_is_valid_format(format)) { 193 LOGE("Invalid format"); 194 return BAD_VALUE; 195 } 196 197 // force direct flag if format is not linear PCM 198 if (!audio_is_linear_pcm(format)) { 199 flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT; 200 } 201 202 if (!audio_is_output_channel(channelMask)) { 203 LOGE("Invalid channel mask"); 204 return BAD_VALUE; 205 } 206 uint32_t channelCount = popcount(channelMask); 207 208 audio_io_handle_t output = AudioSystem::getOutput( 209 (audio_stream_type_t)streamType, 210 sampleRate,format, channelMask, 211 (audio_policy_output_flags_t)flags); 212 213 if (output == 0) { 214 LOGE("Could not get audio output for stream type %d", streamType); 215 return BAD_VALUE; 216 } 217 218 mVolume[LEFT] = 1.0f; 219 mVolume[RIGHT] = 1.0f; 220 mSendLevel = 0; 221 mFrameCount = frameCount; 222 mNotificationFramesReq = notificationFrames; 223 mSessionId = sessionId; 224 mAuxEffectId = 0; 225 226 // create the IAudioTrack 227 status_t status = createTrack_l(streamType, 228 sampleRate, 229 (uint32_t)format, 230 (uint32_t)channelMask, 231 frameCount, 232 flags, 233 sharedBuffer, 234 output, 235 true); 236 237 if (status != NO_ERROR) { 238 return status; 239 } 240 241 if (cbf != 0) { 242 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 243 if (mAudioTrackThread == 0) { 244 LOGE("Could not create callback thread"); 245 return NO_INIT; 246 } 247 } 248 249 mStatus = NO_ERROR; 250 251 mStreamType = streamType; 252 mFormat = (uint32_t)format; 253 mChannelMask = (uint32_t)channelMask; 254 mChannelCount = channelCount; 255 mSharedBuffer = sharedBuffer; 256 mMuted = false; 257 mActive = 0; 258 mCbf = cbf; 259 mUserData = user; 260 mLoopCount = 0; 261 mMarkerPosition = 0; 262 mMarkerReached = false; 263 mNewPosition = 0; 264 mUpdatePeriod = 0; 265 mFlushed = false; 266 mFlags = flags; 267 AudioSystem::acquireAudioSessionId(mSessionId); 268 mRestoreStatus = NO_ERROR; 269 return NO_ERROR; 270} 271 272status_t AudioTrack::initCheck() const 273{ 274 return mStatus; 275} 276 277// ------------------------------------------------------------------------- 278 279uint32_t AudioTrack::latency() const 280{ 281 return mLatency; 282} 283 284int AudioTrack::streamType() const 285{ 286 return mStreamType; 287} 288 289int AudioTrack::format() const 290{ 291 return mFormat; 292} 293 294int AudioTrack::channelCount() const 295{ 296 return mChannelCount; 297} 298 299uint32_t AudioTrack::frameCount() const 300{ 301 return mCblk->frameCount; 302} 303 304int AudioTrack::frameSize() const 305{ 306 if (audio_is_linear_pcm(mFormat)) { 307 return channelCount()*audio_bytes_per_sample(mFormat); 308 } else { 309 return sizeof(uint8_t); 310 } 311} 312 313sp<IMemory>& AudioTrack::sharedBuffer() 314{ 315 return mSharedBuffer; 316} 317 318// ------------------------------------------------------------------------- 319 320void AudioTrack::start() 321{ 322 sp<AudioTrackThread> t = mAudioTrackThread; 323 status_t status = NO_ERROR; 324 325 ALOGV("start %p", this); 326 if (t != 0) { 327 if (t->exitPending()) { 328 if (t->requestExitAndWait() == WOULD_BLOCK) { 329 LOGE("AudioTrack::start called from thread"); 330 return; 331 } 332 } 333 t->mLock.lock(); 334 } 335 336 AutoMutex lock(mLock); 337 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 338 // while we are accessing the cblk 339 sp <IAudioTrack> audioTrack = mAudioTrack; 340 sp <IMemory> iMem = mCblkMemory; 341 audio_track_cblk_t* cblk = mCblk; 342 343 if (mActive == 0) { 344 mFlushed = false; 345 mActive = 1; 346 mNewPosition = cblk->server + mUpdatePeriod; 347 cblk->lock.lock(); 348 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 349 cblk->waitTimeMs = 0; 350 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 351 if (t != 0) { 352 t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO); 353 } else { 354 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 355 mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0); 356 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 357 } 358 359 ALOGV("start %p before lock cblk %p", this, mCblk); 360 if (!(cblk->flags & CBLK_INVALID_MSK)) { 361 cblk->lock.unlock(); 362 status = mAudioTrack->start(); 363 cblk->lock.lock(); 364 if (status == DEAD_OBJECT) { 365 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 366 } 367 } 368 if (cblk->flags & CBLK_INVALID_MSK) { 369 status = restoreTrack_l(cblk, true); 370 } 371 cblk->lock.unlock(); 372 if (status != NO_ERROR) { 373 ALOGV("start() failed"); 374 mActive = 0; 375 if (t != 0) { 376 t->requestExit(); 377 } else { 378 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 379 androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup); 380 } 381 } 382 } 383 384 if (t != 0) { 385 t->mLock.unlock(); 386 } 387} 388 389void AudioTrack::stop() 390{ 391 sp<AudioTrackThread> t = mAudioTrackThread; 392 393 ALOGV("stop %p", this); 394 if (t != 0) { 395 t->mLock.lock(); 396 } 397 398 AutoMutex lock(mLock); 399 if (mActive == 1) { 400 mActive = 0; 401 mCblk->cv.signal(); 402 mAudioTrack->stop(); 403 // Cancel loops (If we are in the middle of a loop, playback 404 // would not stop until loopCount reaches 0). 405 setLoop_l(0, 0, 0); 406 // the playback head position will reset to 0, so if a marker is set, we need 407 // to activate it again 408 mMarkerReached = false; 409 // Force flush if a shared buffer is used otherwise audioflinger 410 // will not stop before end of buffer is reached. 411 if (mSharedBuffer != 0) { 412 flush_l(); 413 } 414 if (t != 0) { 415 t->requestExit(); 416 } else { 417 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 418 androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup); 419 } 420 } 421 422 if (t != 0) { 423 t->mLock.unlock(); 424 } 425} 426 427bool AudioTrack::stopped() const 428{ 429 return !mActive; 430} 431 432void AudioTrack::flush() 433{ 434 AutoMutex lock(mLock); 435 flush_l(); 436} 437 438// must be called with mLock held 439void AudioTrack::flush_l() 440{ 441 ALOGV("flush"); 442 443 // clear playback marker and periodic update counter 444 mMarkerPosition = 0; 445 mMarkerReached = false; 446 mUpdatePeriod = 0; 447 448 if (!mActive) { 449 mFlushed = true; 450 mAudioTrack->flush(); 451 // Release AudioTrack callback thread in case it was waiting for new buffers 452 // in AudioTrack::obtainBuffer() 453 mCblk->cv.signal(); 454 } 455} 456 457void AudioTrack::pause() 458{ 459 ALOGV("pause"); 460 AutoMutex lock(mLock); 461 if (mActive == 1) { 462 mActive = 0; 463 mAudioTrack->pause(); 464 } 465} 466 467void AudioTrack::mute(bool e) 468{ 469 mAudioTrack->mute(e); 470 mMuted = e; 471} 472 473bool AudioTrack::muted() const 474{ 475 return mMuted; 476} 477 478status_t AudioTrack::setVolume(float left, float right) 479{ 480 if (left > 1.0f || right > 1.0f) { 481 return BAD_VALUE; 482 } 483 484 AutoMutex lock(mLock); 485 mVolume[LEFT] = left; 486 mVolume[RIGHT] = right; 487 488 // write must be atomic 489 mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000); 490 491 return NO_ERROR; 492} 493 494void AudioTrack::getVolume(float* left, float* right) 495{ 496 if (left != NULL) { 497 *left = mVolume[LEFT]; 498 } 499 if (right != NULL) { 500 *right = mVolume[RIGHT]; 501 } 502} 503 504status_t AudioTrack::setAuxEffectSendLevel(float level) 505{ 506 ALOGV("setAuxEffectSendLevel(%f)", level); 507 if (level > 1.0f) { 508 return BAD_VALUE; 509 } 510 AutoMutex lock(mLock); 511 512 mSendLevel = level; 513 514 mCblk->sendLevel = uint16_t(level * 0x1000); 515 516 return NO_ERROR; 517} 518 519void AudioTrack::getAuxEffectSendLevel(float* level) 520{ 521 if (level != NULL) { 522 *level = mSendLevel; 523 } 524} 525 526status_t AudioTrack::setSampleRate(int rate) 527{ 528 int afSamplingRate; 529 530 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 531 return NO_INIT; 532 } 533 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 534 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 535 536 AutoMutex lock(mLock); 537 mCblk->sampleRate = rate; 538 return NO_ERROR; 539} 540 541uint32_t AudioTrack::getSampleRate() 542{ 543 AutoMutex lock(mLock); 544 return mCblk->sampleRate; 545} 546 547status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 548{ 549 AutoMutex lock(mLock); 550 return setLoop_l(loopStart, loopEnd, loopCount); 551} 552 553// must be called with mLock held 554status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 555{ 556 audio_track_cblk_t* cblk = mCblk; 557 558 Mutex::Autolock _l(cblk->lock); 559 560 if (loopCount == 0) { 561 cblk->loopStart = UINT_MAX; 562 cblk->loopEnd = UINT_MAX; 563 cblk->loopCount = 0; 564 mLoopCount = 0; 565 return NO_ERROR; 566 } 567 568 if (loopStart >= loopEnd || 569 loopEnd - loopStart > cblk->frameCount || 570 cblk->server > loopStart) { 571 LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 572 return BAD_VALUE; 573 } 574 575 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 576 LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 577 loopStart, loopEnd, cblk->frameCount); 578 return BAD_VALUE; 579 } 580 581 cblk->loopStart = loopStart; 582 cblk->loopEnd = loopEnd; 583 cblk->loopCount = loopCount; 584 mLoopCount = loopCount; 585 586 return NO_ERROR; 587} 588 589status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) 590{ 591 AutoMutex lock(mLock); 592 if (loopStart != 0) { 593 *loopStart = mCblk->loopStart; 594 } 595 if (loopEnd != 0) { 596 *loopEnd = mCblk->loopEnd; 597 } 598 if (loopCount != 0) { 599 if (mCblk->loopCount < 0) { 600 *loopCount = -1; 601 } else { 602 *loopCount = mCblk->loopCount; 603 } 604 } 605 606 return NO_ERROR; 607} 608 609status_t AudioTrack::setMarkerPosition(uint32_t marker) 610{ 611 if (mCbf == 0) return INVALID_OPERATION; 612 613 mMarkerPosition = marker; 614 mMarkerReached = false; 615 616 return NO_ERROR; 617} 618 619status_t AudioTrack::getMarkerPosition(uint32_t *marker) 620{ 621 if (marker == 0) return BAD_VALUE; 622 623 *marker = mMarkerPosition; 624 625 return NO_ERROR; 626} 627 628status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 629{ 630 if (mCbf == 0) return INVALID_OPERATION; 631 632 uint32_t curPosition; 633 getPosition(&curPosition); 634 mNewPosition = curPosition + updatePeriod; 635 mUpdatePeriod = updatePeriod; 636 637 return NO_ERROR; 638} 639 640status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) 641{ 642 if (updatePeriod == 0) return BAD_VALUE; 643 644 *updatePeriod = mUpdatePeriod; 645 646 return NO_ERROR; 647} 648 649status_t AudioTrack::setPosition(uint32_t position) 650{ 651 AutoMutex lock(mLock); 652 Mutex::Autolock _l(mCblk->lock); 653 654 if (!stopped()) return INVALID_OPERATION; 655 656 if (position > mCblk->user) return BAD_VALUE; 657 658 mCblk->server = position; 659 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 660 661 return NO_ERROR; 662} 663 664status_t AudioTrack::getPosition(uint32_t *position) 665{ 666 if (position == 0) return BAD_VALUE; 667 AutoMutex lock(mLock); 668 *position = mFlushed ? 0 : mCblk->server; 669 670 return NO_ERROR; 671} 672 673status_t AudioTrack::reload() 674{ 675 AutoMutex lock(mLock); 676 677 if (!stopped()) return INVALID_OPERATION; 678 679 flush_l(); 680 681 mCblk->stepUser(mCblk->frameCount); 682 683 return NO_ERROR; 684} 685 686audio_io_handle_t AudioTrack::getOutput() 687{ 688 AutoMutex lock(mLock); 689 return getOutput_l(); 690} 691 692// must be called with mLock held 693audio_io_handle_t AudioTrack::getOutput_l() 694{ 695 return AudioSystem::getOutput((audio_stream_type_t)mStreamType, 696 mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags); 697} 698 699int AudioTrack::getSessionId() 700{ 701 return mSessionId; 702} 703 704status_t AudioTrack::attachAuxEffect(int effectId) 705{ 706 ALOGV("attachAuxEffect(%d)", effectId); 707 status_t status = mAudioTrack->attachAuxEffect(effectId); 708 if (status == NO_ERROR) { 709 mAuxEffectId = effectId; 710 } 711 return status; 712} 713 714// ------------------------------------------------------------------------- 715 716// must be called with mLock held 717status_t AudioTrack::createTrack_l( 718 int streamType, 719 uint32_t sampleRate, 720 uint32_t format, 721 uint32_t channelMask, 722 int frameCount, 723 uint32_t flags, 724 const sp<IMemory>& sharedBuffer, 725 audio_io_handle_t output, 726 bool enforceFrameCount) 727{ 728 status_t status; 729 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 730 if (audioFlinger == 0) { 731 LOGE("Could not get audioflinger"); 732 return NO_INIT; 733 } 734 735 int afSampleRate; 736 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 737 return NO_INIT; 738 } 739 int afFrameCount; 740 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 741 return NO_INIT; 742 } 743 uint32_t afLatency; 744 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 745 return NO_INIT; 746 } 747 748 mNotificationFramesAct = mNotificationFramesReq; 749 if (!audio_is_linear_pcm(format)) { 750 if (sharedBuffer != 0) { 751 frameCount = sharedBuffer->size(); 752 } 753 } else { 754 // Ensure that buffer depth covers at least audio hardware latency 755 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 756 if (minBufCount < 2) minBufCount = 2; 757 758 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 759 760 if (sharedBuffer == 0) { 761 if (frameCount == 0) { 762 frameCount = minFrameCount; 763 } 764 if (mNotificationFramesAct == 0) { 765 mNotificationFramesAct = frameCount/2; 766 } 767 // Make sure that application is notified with sufficient margin 768 // before underrun 769 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 770 mNotificationFramesAct = frameCount/2; 771 } 772 if (frameCount < minFrameCount) { 773 if (enforceFrameCount) { 774 LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); 775 return BAD_VALUE; 776 } else { 777 frameCount = minFrameCount; 778 } 779 } 780 } else { 781 // Ensure that buffer alignment matches channelcount 782 int channelCount = popcount(channelMask); 783 if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { 784 LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); 785 return BAD_VALUE; 786 } 787 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 788 } 789 } 790 791 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 792 streamType, 793 sampleRate, 794 format, 795 channelMask, 796 frameCount, 797 ((uint16_t)flags) << 16, 798 sharedBuffer, 799 output, 800 &mSessionId, 801 &status); 802 803 if (track == 0) { 804 LOGE("AudioFlinger could not create track, status: %d", status); 805 return status; 806 } 807 sp<IMemory> cblk = track->getCblk(); 808 if (cblk == 0) { 809 LOGE("Could not get control block"); 810 return NO_INIT; 811 } 812 mAudioTrack.clear(); 813 mAudioTrack = track; 814 mCblkMemory.clear(); 815 mCblkMemory = cblk; 816 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 817 android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 818 if (sharedBuffer == 0) { 819 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 820 } else { 821 mCblk->buffers = sharedBuffer->pointer(); 822 // Force buffer full condition as data is already present in shared memory 823 mCblk->stepUser(mCblk->frameCount); 824 } 825 826 mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000); 827 mCblk->sendLevel = uint16_t(mSendLevel * 0x1000); 828 mAudioTrack->attachAuxEffect(mAuxEffectId); 829 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 830 mCblk->waitTimeMs = 0; 831 mRemainingFrames = mNotificationFramesAct; 832 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 833 return NO_ERROR; 834} 835 836status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 837{ 838 AutoMutex lock(mLock); 839 int active; 840 status_t result = NO_ERROR; 841 audio_track_cblk_t* cblk = mCblk; 842 uint32_t framesReq = audioBuffer->frameCount; 843 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 844 845 audioBuffer->frameCount = 0; 846 audioBuffer->size = 0; 847 848 uint32_t framesAvail = cblk->framesAvailable(); 849 850 cblk->lock.lock(); 851 if (cblk->flags & CBLK_INVALID_MSK) { 852 goto create_new_track; 853 } 854 cblk->lock.unlock(); 855 856 if (framesAvail == 0) { 857 cblk->lock.lock(); 858 goto start_loop_here; 859 while (framesAvail == 0) { 860 active = mActive; 861 if (UNLIKELY(!active)) { 862 ALOGV("Not active and NO_MORE_BUFFERS"); 863 cblk->lock.unlock(); 864 return NO_MORE_BUFFERS; 865 } 866 if (UNLIKELY(!waitCount)) { 867 cblk->lock.unlock(); 868 return WOULD_BLOCK; 869 } 870 if (!(cblk->flags & CBLK_INVALID_MSK)) { 871 mLock.unlock(); 872 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 873 cblk->lock.unlock(); 874 mLock.lock(); 875 if (mActive == 0) { 876 return status_t(STOPPED); 877 } 878 cblk->lock.lock(); 879 } 880 881 if (cblk->flags & CBLK_INVALID_MSK) { 882 goto create_new_track; 883 } 884 if (__builtin_expect(result!=NO_ERROR, false)) { 885 cblk->waitTimeMs += waitTimeMs; 886 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 887 // timing out when a loop has been set and we have already written upto loop end 888 // is a normal condition: no need to wake AudioFlinger up. 889 if (cblk->user < cblk->loopEnd) { 890 LOGW( "obtainBuffer timed out (is the CPU pegged?) %p " 891 "user=%08x, server=%08x", this, cblk->user, cblk->server); 892 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 893 cblk->lock.unlock(); 894 result = mAudioTrack->start(); 895 cblk->lock.lock(); 896 if (result == DEAD_OBJECT) { 897 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 898create_new_track: 899 result = restoreTrack_l(cblk, false); 900 } 901 if (result != NO_ERROR) { 902 LOGW("obtainBuffer create Track error %d", result); 903 cblk->lock.unlock(); 904 return result; 905 } 906 } 907 cblk->waitTimeMs = 0; 908 } 909 910 if (--waitCount == 0) { 911 cblk->lock.unlock(); 912 return TIMED_OUT; 913 } 914 } 915 // read the server count again 916 start_loop_here: 917 framesAvail = cblk->framesAvailable_l(); 918 } 919 cblk->lock.unlock(); 920 } 921 922 // restart track if it was disabled by audioflinger due to previous underrun 923 if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) { 924 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 925 LOGW("obtainBuffer() track %p disabled, restarting", this); 926 mAudioTrack->start(); 927 } 928 929 cblk->waitTimeMs = 0; 930 931 if (framesReq > framesAvail) { 932 framesReq = framesAvail; 933 } 934 935 uint32_t u = cblk->user; 936 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 937 938 if (u + framesReq > bufferEnd) { 939 framesReq = bufferEnd - u; 940 } 941 942 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 943 audioBuffer->channelCount = mChannelCount; 944 audioBuffer->frameCount = framesReq; 945 audioBuffer->size = framesReq * cblk->frameSize; 946 if (audio_is_linear_pcm(mFormat)) { 947 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 948 } else { 949 audioBuffer->format = mFormat; 950 } 951 audioBuffer->raw = (int8_t *)cblk->buffer(u); 952 active = mActive; 953 return active ? status_t(NO_ERROR) : status_t(STOPPED); 954} 955 956void AudioTrack::releaseBuffer(Buffer* audioBuffer) 957{ 958 AutoMutex lock(mLock); 959 mCblk->stepUser(audioBuffer->frameCount); 960} 961 962// ------------------------------------------------------------------------- 963 964ssize_t AudioTrack::write(const void* buffer, size_t userSize) 965{ 966 967 if (mSharedBuffer != 0) return INVALID_OPERATION; 968 969 if (ssize_t(userSize) < 0) { 970 // sanity-check. user is most-likely passing an error code. 971 LOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 972 buffer, userSize, userSize); 973 return BAD_VALUE; 974 } 975 976 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 977 978 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 979 // while we are accessing the cblk 980 mLock.lock(); 981 sp <IAudioTrack> audioTrack = mAudioTrack; 982 sp <IMemory> iMem = mCblkMemory; 983 mLock.unlock(); 984 985 ssize_t written = 0; 986 const int8_t *src = (const int8_t *)buffer; 987 Buffer audioBuffer; 988 size_t frameSz = (size_t)frameSize(); 989 990 do { 991 audioBuffer.frameCount = userSize/frameSz; 992 993 // Calling obtainBuffer() with a negative wait count causes 994 // an (almost) infinite wait time. 995 status_t err = obtainBuffer(&audioBuffer, -1); 996 if (err < 0) { 997 // out of buffers, return #bytes written 998 if (err == status_t(NO_MORE_BUFFERS)) 999 break; 1000 return ssize_t(err); 1001 } 1002 1003 size_t toWrite; 1004 1005 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1006 // Divide capacity by 2 to take expansion into account 1007 toWrite = audioBuffer.size>>1; 1008 // 8 to 16 bit conversion 1009 int count = toWrite; 1010 int16_t *dst = (int16_t *)(audioBuffer.i8); 1011 while(count--) { 1012 *dst++ = (int16_t)(*src++^0x80) << 8; 1013 } 1014 } else { 1015 toWrite = audioBuffer.size; 1016 memcpy(audioBuffer.i8, src, toWrite); 1017 src += toWrite; 1018 } 1019 userSize -= toWrite; 1020 written += toWrite; 1021 1022 releaseBuffer(&audioBuffer); 1023 } while (userSize >= frameSz); 1024 1025 return written; 1026} 1027 1028// ------------------------------------------------------------------------- 1029 1030bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1031{ 1032 Buffer audioBuffer; 1033 uint32_t frames; 1034 size_t writtenSize; 1035 1036 mLock.lock(); 1037 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1038 // while we are accessing the cblk 1039 sp <IAudioTrack> audioTrack = mAudioTrack; 1040 sp <IMemory> iMem = mCblkMemory; 1041 audio_track_cblk_t* cblk = mCblk; 1042 mLock.unlock(); 1043 1044 // Manage underrun callback 1045 if (mActive && (cblk->framesAvailable() == cblk->frameCount)) { 1046 ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1047 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1048 mCbf(EVENT_UNDERRUN, mUserData, 0); 1049 if (cblk->server == cblk->frameCount) { 1050 mCbf(EVENT_BUFFER_END, mUserData, 0); 1051 } 1052 if (mSharedBuffer != 0) return false; 1053 } 1054 } 1055 1056 // Manage loop end callback 1057 while (mLoopCount > cblk->loopCount) { 1058 int loopCount = -1; 1059 mLoopCount--; 1060 if (mLoopCount >= 0) loopCount = mLoopCount; 1061 1062 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1063 } 1064 1065 // Manage marker callback 1066 if (!mMarkerReached && (mMarkerPosition > 0)) { 1067 if (cblk->server >= mMarkerPosition) { 1068 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1069 mMarkerReached = true; 1070 } 1071 } 1072 1073 // Manage new position callback 1074 if (mUpdatePeriod > 0) { 1075 while (cblk->server >= mNewPosition) { 1076 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1077 mNewPosition += mUpdatePeriod; 1078 } 1079 } 1080 1081 // If Shared buffer is used, no data is requested from client. 1082 if (mSharedBuffer != 0) { 1083 frames = 0; 1084 } else { 1085 frames = mRemainingFrames; 1086 } 1087 1088 int32_t waitCount = -1; 1089 if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { 1090 waitCount = 1; 1091 } 1092 1093 do { 1094 1095 audioBuffer.frameCount = frames; 1096 1097 // Calling obtainBuffer() with a wait count of 1 1098 // limits wait time to WAIT_PERIOD_MS. This prevents from being 1099 // stuck here not being able to handle timed events (position, markers, loops). 1100 status_t err = obtainBuffer(&audioBuffer, waitCount); 1101 if (err < NO_ERROR) { 1102 if (err != TIMED_OUT) { 1103 LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); 1104 return false; 1105 } 1106 break; 1107 } 1108 if (err == status_t(STOPPED)) return false; 1109 1110 // Divide buffer size by 2 to take into account the expansion 1111 // due to 8 to 16 bit conversion: the callback must fill only half 1112 // of the destination buffer 1113 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1114 audioBuffer.size >>= 1; 1115 } 1116 1117 size_t reqSize = audioBuffer.size; 1118 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1119 writtenSize = audioBuffer.size; 1120 1121 // Sanity check on returned size 1122 if (ssize_t(writtenSize) <= 0) { 1123 // The callback is done filling buffers 1124 // Keep this thread going to handle timed events and 1125 // still try to get more data in intervals of WAIT_PERIOD_MS 1126 // but don't just loop and block the CPU, so wait 1127 usleep(WAIT_PERIOD_MS*1000); 1128 break; 1129 } 1130 if (writtenSize > reqSize) writtenSize = reqSize; 1131 1132 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1133 // 8 to 16 bit conversion 1134 const int8_t *src = audioBuffer.i8 + writtenSize-1; 1135 int count = writtenSize; 1136 int16_t *dst = audioBuffer.i16 + writtenSize-1; 1137 while(count--) { 1138 *dst-- = (int16_t)(*src--^0x80) << 8; 1139 } 1140 writtenSize <<= 1; 1141 } 1142 1143 audioBuffer.size = writtenSize; 1144 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1145 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of 1146 // 16 bit. 1147 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1148 1149 frames -= audioBuffer.frameCount; 1150 1151 releaseBuffer(&audioBuffer); 1152 } 1153 while (frames); 1154 1155 if (frames == 0) { 1156 mRemainingFrames = mNotificationFramesAct; 1157 } else { 1158 mRemainingFrames = frames; 1159 } 1160 return true; 1161} 1162 1163// must be called with mLock and cblk.lock held. Callers must also hold strong references on 1164// the IAudioTrack and IMemory in case they are recreated here. 1165// If the IAudioTrack is successfully restored, the cblk pointer is updated 1166status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1167{ 1168 status_t result; 1169 1170 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1171 LOGW("dead IAudioTrack, creating a new one from %s TID %d", 1172 fromStart ? "start()" : "obtainBuffer()", gettid()); 1173 1174 // signal old cblk condition so that other threads waiting for available buffers stop 1175 // waiting now 1176 cblk->cv.broadcast(); 1177 cblk->lock.unlock(); 1178 1179 // refresh the audio configuration cache in this process to make sure we get new 1180 // output parameters in getOutput_l() and createTrack_l() 1181 AudioSystem::clearAudioConfigCache(); 1182 1183 // if the new IAudioTrack is created, createTrack_l() will modify the 1184 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1185 // It will also delete the strong references on previous IAudioTrack and IMemory 1186 result = createTrack_l(mStreamType, 1187 cblk->sampleRate, 1188 mFormat, 1189 mChannelMask, 1190 mFrameCount, 1191 mFlags, 1192 mSharedBuffer, 1193 getOutput_l(), 1194 false); 1195 1196 if (result == NO_ERROR) { 1197 uint32_t user = cblk->user; 1198 uint32_t server = cblk->server; 1199 // restore write index and set other indexes to reflect empty buffer status 1200 mCblk->user = user; 1201 mCblk->server = user; 1202 mCblk->userBase = user; 1203 mCblk->serverBase = user; 1204 // restore loop: this is not guaranteed to succeed if new frame count is not 1205 // compatible with loop length 1206 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1207 if (!fromStart) { 1208 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1209 // Make sure that a client relying on callback events indicating underrun or 1210 // the actual amount of audio frames played (e.g SoundPool) receives them. 1211 if (mSharedBuffer == 0) { 1212 uint32_t frames = 0; 1213 if (user > server) { 1214 frames = ((user - server) > mCblk->frameCount) ? 1215 mCblk->frameCount : (user - server); 1216 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1217 } 1218 // restart playback even if buffer is not completely filled. 1219 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1220 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1221 // the client 1222 mCblk->stepUser(frames); 1223 } 1224 } 1225 if (mActive) { 1226 result = mAudioTrack->start(); 1227 LOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1228 } 1229 if (fromStart && result == NO_ERROR) { 1230 mNewPosition = mCblk->server + mUpdatePeriod; 1231 } 1232 } 1233 if (result != NO_ERROR) { 1234 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1235 LOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1236 } 1237 mRestoreStatus = result; 1238 // signal old cblk condition for other threads waiting for restore completion 1239 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1240 cblk->cv.broadcast(); 1241 } else { 1242 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1243 LOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1244 mLock.unlock(); 1245 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1246 if (result == NO_ERROR) { 1247 result = mRestoreStatus; 1248 } 1249 cblk->lock.unlock(); 1250 mLock.lock(); 1251 } else { 1252 LOGW("dead IAudioTrack, already restored TID %d", gettid()); 1253 result = mRestoreStatus; 1254 cblk->lock.unlock(); 1255 } 1256 } 1257 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1258 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1259 1260 if (result == NO_ERROR) { 1261 // from now on we switch to the newly created cblk 1262 cblk = mCblk; 1263 } 1264 cblk->lock.lock(); 1265 1266 LOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1267 1268 return result; 1269} 1270 1271status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1272{ 1273 1274 const size_t SIZE = 256; 1275 char buffer[SIZE]; 1276 String8 result; 1277 1278 result.append(" AudioTrack::dump\n"); 1279 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1280 result.append(buffer); 1281 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1282 result.append(buffer); 1283 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1284 result.append(buffer); 1285 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1286 result.append(buffer); 1287 ::write(fd, result.string(), result.size()); 1288 return NO_ERROR; 1289} 1290 1291// ========================================================================= 1292 1293AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1294 : Thread(bCanCallJava), mReceiver(receiver) 1295{ 1296} 1297 1298bool AudioTrack::AudioTrackThread::threadLoop() 1299{ 1300 return mReceiver.processAudioBuffer(this); 1301} 1302 1303status_t AudioTrack::AudioTrackThread::readyToRun() 1304{ 1305 return NO_ERROR; 1306} 1307 1308void AudioTrack::AudioTrackThread::onFirstRef() 1309{ 1310} 1311 1312// ========================================================================= 1313 1314 1315audio_track_cblk_t::audio_track_cblk_t() 1316 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1317 userBase(0), serverBase(0), buffers(0), frameCount(0), 1318 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), 1319 sendLevel(0), flags(0) 1320{ 1321} 1322 1323uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1324{ 1325 uint32_t u = this->user; 1326 1327 u += frameCount; 1328 // Ensure that user is never ahead of server for AudioRecord 1329 if (flags & CBLK_DIRECTION_MSK) { 1330 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1331 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1332 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1333 } 1334 } else if (u > this->server) { 1335 LOGW("stepServer occured after track reset"); 1336 u = this->server; 1337 } 1338 1339 if (u >= userBase + this->frameCount) { 1340 userBase += this->frameCount; 1341 } 1342 1343 this->user = u; 1344 1345 // Clear flow control error condition as new data has been written/read to/from buffer. 1346 if (flags & CBLK_UNDERRUN_MSK) { 1347 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1348 } 1349 1350 return u; 1351} 1352 1353bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1354{ 1355 if (!tryLock()) { 1356 LOGW("stepServer() could not lock cblk"); 1357 return false; 1358 } 1359 1360 uint32_t s = this->server; 1361 1362 s += frameCount; 1363 if (flags & CBLK_DIRECTION_MSK) { 1364 // Mark that we have read the first buffer so that next time stepUser() is called 1365 // we switch to normal obtainBuffer() timeout period 1366 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1367 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1368 } 1369 // It is possible that we receive a flush() 1370 // while the mixer is processing a block: in this case, 1371 // stepServer() is called After the flush() has reset u & s and 1372 // we have s > u 1373 if (s > this->user) { 1374 LOGW("stepServer occured after track reset"); 1375 s = this->user; 1376 } 1377 } 1378 1379 if (s >= loopEnd) { 1380 LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1381 s = loopStart; 1382 if (--loopCount == 0) { 1383 loopEnd = UINT_MAX; 1384 loopStart = UINT_MAX; 1385 } 1386 } 1387 if (s >= serverBase + this->frameCount) { 1388 serverBase += this->frameCount; 1389 } 1390 1391 this->server = s; 1392 1393 if (!(flags & CBLK_INVALID_MSK)) { 1394 cv.signal(); 1395 } 1396 lock.unlock(); 1397 return true; 1398} 1399 1400void* audio_track_cblk_t::buffer(uint32_t offset) const 1401{ 1402 return (int8_t *)this->buffers + (offset - userBase) * this->frameSize; 1403} 1404 1405uint32_t audio_track_cblk_t::framesAvailable() 1406{ 1407 Mutex::Autolock _l(lock); 1408 return framesAvailable_l(); 1409} 1410 1411uint32_t audio_track_cblk_t::framesAvailable_l() 1412{ 1413 uint32_t u = this->user; 1414 uint32_t s = this->server; 1415 1416 if (flags & CBLK_DIRECTION_MSK) { 1417 uint32_t limit = (s < loopStart) ? s : loopStart; 1418 return limit + frameCount - u; 1419 } else { 1420 return frameCount + u - s; 1421 } 1422} 1423 1424uint32_t audio_track_cblk_t::framesReady() 1425{ 1426 uint32_t u = this->user; 1427 uint32_t s = this->server; 1428 1429 if (flags & CBLK_DIRECTION_MSK) { 1430 if (u < loopEnd) { 1431 return u - s; 1432 } else { 1433 // do not block on mutex shared with client on AudioFlinger side 1434 if (!tryLock()) { 1435 LOGW("framesReady() could not lock cblk"); 1436 return 0; 1437 } 1438 uint32_t frames = UINT_MAX; 1439 if (loopCount >= 0) { 1440 frames = (loopEnd - loopStart)*loopCount + u - s; 1441 } 1442 lock.unlock(); 1443 return frames; 1444 } 1445 } else { 1446 return s - u; 1447 } 1448} 1449 1450bool audio_track_cblk_t::tryLock() 1451{ 1452 // the code below simulates lock-with-timeout 1453 // we MUST do this to protect the AudioFlinger server 1454 // as this lock is shared with the client. 1455 status_t err; 1456 1457 err = lock.tryLock(); 1458 if (err == -EBUSY) { // just wait a bit 1459 usleep(1000); 1460 err = lock.tryLock(); 1461 } 1462 if (err != NO_ERROR) { 1463 // probably, the client just died. 1464 return false; 1465 } 1466 return true; 1467} 1468 1469// ------------------------------------------------------------------------- 1470 1471}; // namespace android 1472