AudioTrack.cpp revision 0e265cf36d201a7ccc0238b5c60b50f43d1dc450
1/* frameworks/base/media/libmedia/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#include <cutils/compiler.h> 42 43#include <system/audio.h> 44#include <system/audio_policy.h> 45 46#include <audio_utils/primitives.h> 47 48namespace android { 49// --------------------------------------------------------------------------- 50 51// static 52status_t AudioTrack::getMinFrameCount( 53 int* frameCount, 54 audio_stream_type_t 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 audio_stream_type_t streamType, 89 uint32_t sampleRate, 90 audio_format_t 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 int frameCount, 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((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask, 121 frameCount, flags, cbf, user, notificationFrames, 122 0, false, sessionId); 123} 124 125AudioTrack::AudioTrack( 126 audio_stream_type_t streamType, 127 uint32_t sampleRate, 128 audio_format_t format, 129 int channelMask, 130 const sp<IMemory>& sharedBuffer, 131 uint32_t flags, 132 callback_t cbf, 133 void* user, 134 int notificationFrames, 135 int sessionId) 136 : mStatus(NO_INIT), 137 mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT) 138{ 139 mStatus = set(streamType, sampleRate, format, channelMask, 140 0, flags, cbf, user, notificationFrames, 141 sharedBuffer, false, sessionId); 142} 143 144AudioTrack::~AudioTrack() 145{ 146 ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); 147 148 if (mStatus == NO_ERROR) { 149 // Make sure that callback function exits in the case where 150 // it is looping on buffer full condition in obtainBuffer(). 151 // Otherwise the callback thread will never exit. 152 stop(); 153 if (mAudioTrackThread != 0) { 154 mAudioTrackThread->requestExitAndWait(); 155 mAudioTrackThread.clear(); 156 } 157 mAudioTrack.clear(); 158 IPCThreadState::self()->flushCommands(); 159 AudioSystem::releaseAudioSessionId(mSessionId); 160 } 161} 162 163status_t AudioTrack::set( 164 audio_stream_type_t streamType, 165 uint32_t sampleRate, 166 audio_format_t format, 167 int channelMask, 168 int frameCount, 169 uint32_t flags, 170 callback_t cbf, 171 void* user, 172 int notificationFrames, 173 const sp<IMemory>& sharedBuffer, 174 bool threadCanCallJava, 175 int sessionId) 176{ 177 178 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 179 180 AutoMutex lock(mLock); 181 if (mAudioTrack != 0) { 182 ALOGE("Track already in use"); 183 return INVALID_OPERATION; 184 } 185 186 int afSampleRate; 187 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 188 return NO_INIT; 189 } 190 uint32_t afLatency; 191 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 192 return NO_INIT; 193 } 194 195 // handle default values first. 196 if (streamType == AUDIO_STREAM_DEFAULT) { 197 streamType = AUDIO_STREAM_MUSIC; 198 } 199 if (sampleRate == 0) { 200 sampleRate = afSampleRate; 201 } 202 // these below should probably come from the audioFlinger too... 203 if (format == AUDIO_FORMAT_DEFAULT) { 204 format = AUDIO_FORMAT_PCM_16_BIT; 205 } 206 if (channelMask == 0) { 207 channelMask = AUDIO_CHANNEL_OUT_STEREO; 208 } 209 210 // validate parameters 211 if (!audio_is_valid_format(format)) { 212 ALOGE("Invalid format"); 213 return BAD_VALUE; 214 } 215 216 // force direct flag if format is not linear PCM 217 if (!audio_is_linear_pcm(format)) { 218 flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT; 219 } 220 221 if (!audio_is_output_channel(channelMask)) { 222 ALOGE("Invalid channel mask"); 223 return BAD_VALUE; 224 } 225 uint32_t channelCount = popcount(channelMask); 226 227 audio_io_handle_t output = AudioSystem::getOutput( 228 streamType, 229 sampleRate, format, channelMask, 230 (audio_policy_output_flags_t)flags); 231 232 if (output == 0) { 233 ALOGE("Could not get audio output for stream type %d", streamType); 234 return BAD_VALUE; 235 } 236 237 mVolume[LEFT] = 1.0f; 238 mVolume[RIGHT] = 1.0f; 239 mSendLevel = 0.0f; 240 mFrameCount = frameCount; 241 mNotificationFramesReq = notificationFrames; 242 mSessionId = sessionId; 243 mAuxEffectId = 0; 244 245 // create the IAudioTrack 246 status_t status = createTrack_l(streamType, 247 sampleRate, 248 format, 249 (uint32_t)channelMask, 250 frameCount, 251 flags, 252 sharedBuffer, 253 output, 254 true); 255 256 if (status != NO_ERROR) { 257 return status; 258 } 259 260 if (cbf != 0) { 261 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 262 } 263 264 mStatus = NO_ERROR; 265 266 mStreamType = streamType; 267 mFormat = format; 268 mChannelMask = (uint32_t)channelMask; 269 mChannelCount = channelCount; 270 mSharedBuffer = sharedBuffer; 271 mMuted = false; 272 mActive = false; 273 mCbf = cbf; 274 mUserData = user; 275 mLoopCount = 0; 276 mMarkerPosition = 0; 277 mMarkerReached = false; 278 mNewPosition = 0; 279 mUpdatePeriod = 0; 280 mFlushed = false; 281 mFlags = flags; 282 AudioSystem::acquireAudioSessionId(mSessionId); 283 mRestoreStatus = NO_ERROR; 284 return NO_ERROR; 285} 286 287status_t AudioTrack::initCheck() const 288{ 289 return mStatus; 290} 291 292// ------------------------------------------------------------------------- 293 294uint32_t AudioTrack::latency() const 295{ 296 return mLatency; 297} 298 299audio_stream_type_t AudioTrack::streamType() const 300{ 301 return mStreamType; 302} 303 304audio_format_t AudioTrack::format() const 305{ 306 return mFormat; 307} 308 309int AudioTrack::channelCount() const 310{ 311 return mChannelCount; 312} 313 314uint32_t AudioTrack::frameCount() const 315{ 316 return mCblk->frameCount; 317} 318 319size_t AudioTrack::frameSize() const 320{ 321 if (audio_is_linear_pcm(mFormat)) { 322 return channelCount()*audio_bytes_per_sample(mFormat); 323 } else { 324 return sizeof(uint8_t); 325 } 326} 327 328sp<IMemory>& AudioTrack::sharedBuffer() 329{ 330 return mSharedBuffer; 331} 332 333// ------------------------------------------------------------------------- 334 335void AudioTrack::start() 336{ 337 sp<AudioTrackThread> t = mAudioTrackThread; 338 status_t status = NO_ERROR; 339 340 ALOGV("start %p", this); 341 if (t != 0) { 342 if (t->exitPending()) { 343 if (t->requestExitAndWait() == WOULD_BLOCK) { 344 ALOGE("AudioTrack::start called from thread"); 345 return; 346 } 347 } 348 } 349 350 AutoMutex lock(mLock); 351 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 352 // while we are accessing the cblk 353 sp <IAudioTrack> audioTrack = mAudioTrack; 354 sp <IMemory> iMem = mCblkMemory; 355 audio_track_cblk_t* cblk = mCblk; 356 357 if (!mActive) { 358 mFlushed = false; 359 mActive = true; 360 mNewPosition = cblk->server + mUpdatePeriod; 361 cblk->lock.lock(); 362 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 363 cblk->waitTimeMs = 0; 364 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 365 if (t != 0) { 366 t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO); 367 } else { 368 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 369 mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0); 370 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 371 } 372 373 ALOGV("start %p before lock cblk %p", this, mCblk); 374 if (!(cblk->flags & CBLK_INVALID_MSK)) { 375 cblk->lock.unlock(); 376 status = mAudioTrack->start(); 377 cblk->lock.lock(); 378 if (status == DEAD_OBJECT) { 379 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 380 } 381 } 382 if (cblk->flags & CBLK_INVALID_MSK) { 383 status = restoreTrack_l(cblk, true); 384 } 385 cblk->lock.unlock(); 386 if (status != NO_ERROR) { 387 ALOGV("start() failed"); 388 mActive = false; 389 if (t != 0) { 390 t->requestExit(); 391 } else { 392 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 393 androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup); 394 } 395 } 396 } 397 398} 399 400void AudioTrack::stop() 401{ 402 sp<AudioTrackThread> t = mAudioTrackThread; 403 404 ALOGV("stop %p", this); 405 406 AutoMutex lock(mLock); 407 if (mActive) { 408 mActive = false; 409 mCblk->cv.signal(); 410 mAudioTrack->stop(); 411 // Cancel loops (If we are in the middle of a loop, playback 412 // would not stop until loopCount reaches 0). 413 setLoop_l(0, 0, 0); 414 // the playback head position will reset to 0, so if a marker is set, we need 415 // to activate it again 416 mMarkerReached = false; 417 // Force flush if a shared buffer is used otherwise audioflinger 418 // will not stop before end of buffer is reached. 419 if (mSharedBuffer != 0) { 420 flush_l(); 421 } 422 if (t != 0) { 423 t->requestExit(); 424 } else { 425 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 426 androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup); 427 } 428 } 429 430} 431 432bool AudioTrack::stopped() const 433{ 434 AutoMutex lock(mLock); 435 return stopped_l(); 436} 437 438void AudioTrack::flush() 439{ 440 AutoMutex lock(mLock); 441 flush_l(); 442} 443 444// must be called with mLock held 445void AudioTrack::flush_l() 446{ 447 ALOGV("flush"); 448 449 // clear playback marker and periodic update counter 450 mMarkerPosition = 0; 451 mMarkerReached = false; 452 mUpdatePeriod = 0; 453 454 if (!mActive) { 455 mFlushed = true; 456 mAudioTrack->flush(); 457 // Release AudioTrack callback thread in case it was waiting for new buffers 458 // in AudioTrack::obtainBuffer() 459 mCblk->cv.signal(); 460 } 461} 462 463void AudioTrack::pause() 464{ 465 ALOGV("pause"); 466 AutoMutex lock(mLock); 467 if (mActive) { 468 mActive = false; 469 mAudioTrack->pause(); 470 } 471} 472 473void AudioTrack::mute(bool e) 474{ 475 mAudioTrack->mute(e); 476 mMuted = e; 477} 478 479bool AudioTrack::muted() const 480{ 481 return mMuted; 482} 483 484status_t AudioTrack::setVolume(float left, float right) 485{ 486 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 487 return BAD_VALUE; 488 } 489 490 AutoMutex lock(mLock); 491 mVolume[LEFT] = left; 492 mVolume[RIGHT] = right; 493 494 mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000); 495 496 return NO_ERROR; 497} 498 499void AudioTrack::getVolume(float* left, float* right) 500{ 501 if (left != NULL) { 502 *left = mVolume[LEFT]; 503 } 504 if (right != NULL) { 505 *right = mVolume[RIGHT]; 506 } 507} 508 509status_t AudioTrack::setAuxEffectSendLevel(float level) 510{ 511 ALOGV("setAuxEffectSendLevel(%f)", level); 512 if (level < 0.0f || level > 1.0f) { 513 return BAD_VALUE; 514 } 515 AutoMutex lock(mLock); 516 517 mSendLevel = level; 518 519 mCblk->setSendLevel(level); 520 521 return NO_ERROR; 522} 523 524void AudioTrack::getAuxEffectSendLevel(float* level) 525{ 526 if (level != NULL) { 527 *level = mSendLevel; 528 } 529} 530 531status_t AudioTrack::setSampleRate(int rate) 532{ 533 int afSamplingRate; 534 535 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 536 return NO_INIT; 537 } 538 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 539 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 540 541 AutoMutex lock(mLock); 542 mCblk->sampleRate = rate; 543 return NO_ERROR; 544} 545 546uint32_t AudioTrack::getSampleRate() 547{ 548 AutoMutex lock(mLock); 549 return mCblk->sampleRate; 550} 551 552status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 553{ 554 AutoMutex lock(mLock); 555 return setLoop_l(loopStart, loopEnd, loopCount); 556} 557 558// must be called with mLock held 559status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 560{ 561 audio_track_cblk_t* cblk = mCblk; 562 563 Mutex::Autolock _l(cblk->lock); 564 565 if (loopCount == 0) { 566 cblk->loopStart = UINT_MAX; 567 cblk->loopEnd = UINT_MAX; 568 cblk->loopCount = 0; 569 mLoopCount = 0; 570 return NO_ERROR; 571 } 572 573 if (loopStart >= loopEnd || 574 loopEnd - loopStart > cblk->frameCount || 575 cblk->server > loopStart) { 576 ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 577 return BAD_VALUE; 578 } 579 580 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 581 ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 582 loopStart, loopEnd, cblk->frameCount); 583 return BAD_VALUE; 584 } 585 586 cblk->loopStart = loopStart; 587 cblk->loopEnd = loopEnd; 588 cblk->loopCount = loopCount; 589 mLoopCount = loopCount; 590 591 return NO_ERROR; 592} 593 594status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) 595{ 596 AutoMutex lock(mLock); 597 if (loopStart != 0) { 598 *loopStart = mCblk->loopStart; 599 } 600 if (loopEnd != 0) { 601 *loopEnd = mCblk->loopEnd; 602 } 603 if (loopCount != 0) { 604 if (mCblk->loopCount < 0) { 605 *loopCount = -1; 606 } else { 607 *loopCount = mCblk->loopCount; 608 } 609 } 610 611 return NO_ERROR; 612} 613 614status_t AudioTrack::setMarkerPosition(uint32_t marker) 615{ 616 if (mCbf == 0) return INVALID_OPERATION; 617 618 mMarkerPosition = marker; 619 mMarkerReached = false; 620 621 return NO_ERROR; 622} 623 624status_t AudioTrack::getMarkerPosition(uint32_t *marker) 625{ 626 if (marker == 0) return BAD_VALUE; 627 628 *marker = mMarkerPosition; 629 630 return NO_ERROR; 631} 632 633status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 634{ 635 if (mCbf == 0) return INVALID_OPERATION; 636 637 uint32_t curPosition; 638 getPosition(&curPosition); 639 mNewPosition = curPosition + updatePeriod; 640 mUpdatePeriod = updatePeriod; 641 642 return NO_ERROR; 643} 644 645status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) 646{ 647 if (updatePeriod == 0) return BAD_VALUE; 648 649 *updatePeriod = mUpdatePeriod; 650 651 return NO_ERROR; 652} 653 654status_t AudioTrack::setPosition(uint32_t position) 655{ 656 AutoMutex lock(mLock); 657 658 if (!stopped_l()) return INVALID_OPERATION; 659 660 Mutex::Autolock _l(mCblk->lock); 661 662 if (position > mCblk->user) return BAD_VALUE; 663 664 mCblk->server = position; 665 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 666 667 return NO_ERROR; 668} 669 670status_t AudioTrack::getPosition(uint32_t *position) 671{ 672 if (position == 0) return BAD_VALUE; 673 AutoMutex lock(mLock); 674 *position = mFlushed ? 0 : mCblk->server; 675 676 return NO_ERROR; 677} 678 679status_t AudioTrack::reload() 680{ 681 AutoMutex lock(mLock); 682 683 if (!stopped_l()) return INVALID_OPERATION; 684 685 flush_l(); 686 687 mCblk->stepUser(mCblk->frameCount); 688 689 return NO_ERROR; 690} 691 692audio_io_handle_t AudioTrack::getOutput() 693{ 694 AutoMutex lock(mLock); 695 return getOutput_l(); 696} 697 698// must be called with mLock held 699audio_io_handle_t AudioTrack::getOutput_l() 700{ 701 return AudioSystem::getOutput(mStreamType, 702 mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags); 703} 704 705int AudioTrack::getSessionId() 706{ 707 return mSessionId; 708} 709 710status_t AudioTrack::attachAuxEffect(int effectId) 711{ 712 ALOGV("attachAuxEffect(%d)", effectId); 713 status_t status = mAudioTrack->attachAuxEffect(effectId); 714 if (status == NO_ERROR) { 715 mAuxEffectId = effectId; 716 } 717 return status; 718} 719 720// ------------------------------------------------------------------------- 721 722// must be called with mLock held 723status_t AudioTrack::createTrack_l( 724 audio_stream_type_t streamType, 725 uint32_t sampleRate, 726 audio_format_t format, 727 uint32_t channelMask, 728 int frameCount, 729 uint32_t flags, 730 const sp<IMemory>& sharedBuffer, 731 audio_io_handle_t output, 732 bool enforceFrameCount) 733{ 734 status_t status; 735 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 736 if (audioFlinger == 0) { 737 ALOGE("Could not get audioflinger"); 738 return NO_INIT; 739 } 740 741 int afSampleRate; 742 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 743 return NO_INIT; 744 } 745 int afFrameCount; 746 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 747 return NO_INIT; 748 } 749 uint32_t afLatency; 750 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 751 return NO_INIT; 752 } 753 754 mNotificationFramesAct = mNotificationFramesReq; 755 if (!audio_is_linear_pcm(format)) { 756 if (sharedBuffer != 0) { 757 frameCount = sharedBuffer->size(); 758 } 759 } else { 760 // Ensure that buffer depth covers at least audio hardware latency 761 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 762 if (minBufCount < 2) minBufCount = 2; 763 764 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 765 766 if (sharedBuffer == 0) { 767 if (frameCount == 0) { 768 frameCount = minFrameCount; 769 } 770 if (mNotificationFramesAct == 0) { 771 mNotificationFramesAct = frameCount/2; 772 } 773 // Make sure that application is notified with sufficient margin 774 // before underrun 775 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 776 mNotificationFramesAct = frameCount/2; 777 } 778 if (frameCount < minFrameCount) { 779 if (enforceFrameCount) { 780 ALOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); 781 return BAD_VALUE; 782 } else { 783 frameCount = minFrameCount; 784 } 785 } 786 } else { 787 // Ensure that buffer alignment matches channelcount 788 int channelCount = popcount(channelMask); 789 if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { 790 ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); 791 return BAD_VALUE; 792 } 793 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 794 } 795 } 796 797 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 798 streamType, 799 sampleRate, 800 format, 801 channelMask, 802 frameCount, 803 ((uint16_t)flags) << 16, 804 sharedBuffer, 805 output, 806 &mSessionId, 807 &status); 808 809 if (track == 0) { 810 ALOGE("AudioFlinger could not create track, status: %d", status); 811 return status; 812 } 813 sp<IMemory> cblk = track->getCblk(); 814 if (cblk == 0) { 815 ALOGE("Could not get control block"); 816 return NO_INIT; 817 } 818 mAudioTrack = track; 819 mCblkMemory = cblk; 820 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 821 android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 822 if (sharedBuffer == 0) { 823 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 824 } else { 825 mCblk->buffers = sharedBuffer->pointer(); 826 // Force buffer full condition as data is already present in shared memory 827 mCblk->stepUser(mCblk->frameCount); 828 } 829 830 mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000); 831 mCblk->setSendLevel(mSendLevel); 832 mAudioTrack->attachAuxEffect(mAuxEffectId); 833 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 834 mCblk->waitTimeMs = 0; 835 mRemainingFrames = mNotificationFramesAct; 836 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 837 return NO_ERROR; 838} 839 840status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 841{ 842 AutoMutex lock(mLock); 843 bool active; 844 status_t result = NO_ERROR; 845 audio_track_cblk_t* cblk = mCblk; 846 uint32_t framesReq = audioBuffer->frameCount; 847 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 848 849 audioBuffer->frameCount = 0; 850 audioBuffer->size = 0; 851 852 uint32_t framesAvail = cblk->framesAvailable(); 853 854 cblk->lock.lock(); 855 if (cblk->flags & CBLK_INVALID_MSK) { 856 goto create_new_track; 857 } 858 cblk->lock.unlock(); 859 860 if (framesAvail == 0) { 861 cblk->lock.lock(); 862 goto start_loop_here; 863 while (framesAvail == 0) { 864 active = mActive; 865 if (CC_UNLIKELY(!active)) { 866 ALOGV("Not active and NO_MORE_BUFFERS"); 867 cblk->lock.unlock(); 868 return NO_MORE_BUFFERS; 869 } 870 if (CC_UNLIKELY(!waitCount)) { 871 cblk->lock.unlock(); 872 return WOULD_BLOCK; 873 } 874 if (!(cblk->flags & CBLK_INVALID_MSK)) { 875 mLock.unlock(); 876 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 877 cblk->lock.unlock(); 878 mLock.lock(); 879 if (!mActive) { 880 return status_t(STOPPED); 881 } 882 cblk->lock.lock(); 883 } 884 885 if (cblk->flags & CBLK_INVALID_MSK) { 886 goto create_new_track; 887 } 888 if (CC_UNLIKELY(result != NO_ERROR)) { 889 cblk->waitTimeMs += waitTimeMs; 890 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 891 // timing out when a loop has been set and we have already written upto loop end 892 // is a normal condition: no need to wake AudioFlinger up. 893 if (cblk->user < cblk->loopEnd) { 894 ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p " 895 "user=%08x, server=%08x", this, cblk->user, cblk->server); 896 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 897 cblk->lock.unlock(); 898 result = mAudioTrack->start(); 899 cblk->lock.lock(); 900 if (result == DEAD_OBJECT) { 901 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 902create_new_track: 903 result = restoreTrack_l(cblk, false); 904 } 905 if (result != NO_ERROR) { 906 ALOGW("obtainBuffer create Track error %d", result); 907 cblk->lock.unlock(); 908 return result; 909 } 910 } 911 cblk->waitTimeMs = 0; 912 } 913 914 if (--waitCount == 0) { 915 cblk->lock.unlock(); 916 return TIMED_OUT; 917 } 918 } 919 // read the server count again 920 start_loop_here: 921 framesAvail = cblk->framesAvailable_l(); 922 } 923 cblk->lock.unlock(); 924 } 925 926 // restart track if it was disabled by audioflinger due to previous underrun 927 if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) { 928 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 929 ALOGW("obtainBuffer() track %p disabled, restarting", this); 930 mAudioTrack->start(); 931 } 932 933 cblk->waitTimeMs = 0; 934 935 if (framesReq > framesAvail) { 936 framesReq = framesAvail; 937 } 938 939 uint32_t u = cblk->user; 940 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 941 942 if (u + framesReq > bufferEnd) { 943 framesReq = bufferEnd - u; 944 } 945 946 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 947 audioBuffer->channelCount = mChannelCount; 948 audioBuffer->frameCount = framesReq; 949 audioBuffer->size = framesReq * cblk->frameSize; 950 if (audio_is_linear_pcm(mFormat)) { 951 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 952 } else { 953 audioBuffer->format = mFormat; 954 } 955 audioBuffer->raw = (int8_t *)cblk->buffer(u); 956 active = mActive; 957 return active ? status_t(NO_ERROR) : status_t(STOPPED); 958} 959 960void AudioTrack::releaseBuffer(Buffer* audioBuffer) 961{ 962 AutoMutex lock(mLock); 963 mCblk->stepUser(audioBuffer->frameCount); 964} 965 966// ------------------------------------------------------------------------- 967 968ssize_t AudioTrack::write(const void* buffer, size_t userSize) 969{ 970 971 if (mSharedBuffer != 0) return INVALID_OPERATION; 972 973 if (ssize_t(userSize) < 0) { 974 // sanity-check. user is most-likely passing an error code. 975 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 976 buffer, userSize, userSize); 977 return BAD_VALUE; 978 } 979 980 ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 981 982 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 983 // while we are accessing the cblk 984 mLock.lock(); 985 sp <IAudioTrack> audioTrack = mAudioTrack; 986 sp <IMemory> iMem = mCblkMemory; 987 mLock.unlock(); 988 989 ssize_t written = 0; 990 const int8_t *src = (const int8_t *)buffer; 991 Buffer audioBuffer; 992 size_t frameSz = frameSize(); 993 994 do { 995 audioBuffer.frameCount = userSize/frameSz; 996 997 // Calling obtainBuffer() with a negative wait count causes 998 // an (almost) infinite wait time. 999 status_t err = obtainBuffer(&audioBuffer, -1); 1000 if (err < 0) { 1001 // out of buffers, return #bytes written 1002 if (err == status_t(NO_MORE_BUFFERS)) 1003 break; 1004 return ssize_t(err); 1005 } 1006 1007 size_t toWrite; 1008 1009 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1010 // Divide capacity by 2 to take expansion into account 1011 toWrite = audioBuffer.size>>1; 1012 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); 1013 } else { 1014 toWrite = audioBuffer.size; 1015 memcpy(audioBuffer.i8, src, toWrite); 1016 src += toWrite; 1017 } 1018 userSize -= toWrite; 1019 written += toWrite; 1020 1021 releaseBuffer(&audioBuffer); 1022 } while (userSize >= frameSz); 1023 1024 return written; 1025} 1026 1027// ------------------------------------------------------------------------- 1028 1029bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1030{ 1031 Buffer audioBuffer; 1032 uint32_t frames; 1033 size_t writtenSize; 1034 1035 mLock.lock(); 1036 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1037 // while we are accessing the cblk 1038 sp <IAudioTrack> audioTrack = mAudioTrack; 1039 sp <IMemory> iMem = mCblkMemory; 1040 audio_track_cblk_t* cblk = mCblk; 1041 bool active = mActive; 1042 mLock.unlock(); 1043 1044 // Manage underrun callback 1045 if (active && (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 ALOGE_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, note that source and destination are the same address 1134 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1135 writtenSize <<= 1; 1136 } 1137 1138 audioBuffer.size = writtenSize; 1139 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1140 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of 1141 // 16 bit. 1142 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1143 1144 frames -= audioBuffer.frameCount; 1145 1146 releaseBuffer(&audioBuffer); 1147 } 1148 while (frames); 1149 1150 if (frames == 0) { 1151 mRemainingFrames = mNotificationFramesAct; 1152 } else { 1153 mRemainingFrames = frames; 1154 } 1155 return true; 1156} 1157 1158// must be called with mLock and cblk.lock held. Callers must also hold strong references on 1159// the IAudioTrack and IMemory in case they are recreated here. 1160// If the IAudioTrack is successfully restored, the cblk pointer is updated 1161status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1162{ 1163 status_t result; 1164 1165 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1166 ALOGW("dead IAudioTrack, creating a new one from %s TID %d", 1167 fromStart ? "start()" : "obtainBuffer()", gettid()); 1168 1169 // signal old cblk condition so that other threads waiting for available buffers stop 1170 // waiting now 1171 cblk->cv.broadcast(); 1172 cblk->lock.unlock(); 1173 1174 // refresh the audio configuration cache in this process to make sure we get new 1175 // output parameters in getOutput_l() and createTrack_l() 1176 AudioSystem::clearAudioConfigCache(); 1177 1178 // if the new IAudioTrack is created, createTrack_l() will modify the 1179 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1180 // It will also delete the strong references on previous IAudioTrack and IMemory 1181 result = createTrack_l(mStreamType, 1182 cblk->sampleRate, 1183 mFormat, 1184 mChannelMask, 1185 mFrameCount, 1186 mFlags, 1187 mSharedBuffer, 1188 getOutput_l(), 1189 false); 1190 1191 if (result == NO_ERROR) { 1192 uint32_t user = cblk->user; 1193 uint32_t server = cblk->server; 1194 // restore write index and set other indexes to reflect empty buffer status 1195 mCblk->user = user; 1196 mCblk->server = user; 1197 mCblk->userBase = user; 1198 mCblk->serverBase = user; 1199 // restore loop: this is not guaranteed to succeed if new frame count is not 1200 // compatible with loop length 1201 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1202 if (!fromStart) { 1203 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1204 // Make sure that a client relying on callback events indicating underrun or 1205 // the actual amount of audio frames played (e.g SoundPool) receives them. 1206 if (mSharedBuffer == 0) { 1207 uint32_t frames = 0; 1208 if (user > server) { 1209 frames = ((user - server) > mCblk->frameCount) ? 1210 mCblk->frameCount : (user - server); 1211 memset(mCblk->buffers, 0, frames * mCblk->frameSize); 1212 } 1213 // restart playback even if buffer is not completely filled. 1214 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 1215 // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to 1216 // the client 1217 mCblk->stepUser(frames); 1218 } 1219 } 1220 if (mActive) { 1221 result = mAudioTrack->start(); 1222 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); 1223 } 1224 if (fromStart && result == NO_ERROR) { 1225 mNewPosition = mCblk->server + mUpdatePeriod; 1226 } 1227 } 1228 if (result != NO_ERROR) { 1229 android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); 1230 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); 1231 } 1232 mRestoreStatus = result; 1233 // signal old cblk condition for other threads waiting for restore completion 1234 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1235 cblk->cv.broadcast(); 1236 } else { 1237 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1238 ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); 1239 mLock.unlock(); 1240 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1241 if (result == NO_ERROR) { 1242 result = mRestoreStatus; 1243 } 1244 cblk->lock.unlock(); 1245 mLock.lock(); 1246 } else { 1247 ALOGW("dead IAudioTrack, already restored TID %d", gettid()); 1248 result = mRestoreStatus; 1249 cblk->lock.unlock(); 1250 } 1251 } 1252 ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1253 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1254 1255 if (result == NO_ERROR) { 1256 // from now on we switch to the newly created cblk 1257 cblk = mCblk; 1258 } 1259 cblk->lock.lock(); 1260 1261 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); 1262 1263 return result; 1264} 1265 1266status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1267{ 1268 1269 const size_t SIZE = 256; 1270 char buffer[SIZE]; 1271 String8 result; 1272 1273 result.append(" AudioTrack::dump\n"); 1274 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1275 result.append(buffer); 1276 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1277 result.append(buffer); 1278 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1279 result.append(buffer); 1280 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1281 result.append(buffer); 1282 ::write(fd, result.string(), result.size()); 1283 return NO_ERROR; 1284} 1285 1286// ========================================================================= 1287 1288AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1289 : Thread(bCanCallJava), mReceiver(receiver) 1290{ 1291} 1292 1293bool AudioTrack::AudioTrackThread::threadLoop() 1294{ 1295 return mReceiver.processAudioBuffer(this); 1296} 1297 1298status_t AudioTrack::AudioTrackThread::readyToRun() 1299{ 1300 return NO_ERROR; 1301} 1302 1303void AudioTrack::AudioTrackThread::onFirstRef() 1304{ 1305} 1306 1307// ========================================================================= 1308 1309 1310audio_track_cblk_t::audio_track_cblk_t() 1311 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1312 userBase(0), serverBase(0), buffers(0), frameCount(0), 1313 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), 1314 mSendLevel(0), flags(0) 1315{ 1316} 1317 1318uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1319{ 1320 uint32_t u = user; 1321 1322 u += frameCount; 1323 // Ensure that user is never ahead of server for AudioRecord 1324 if (flags & CBLK_DIRECTION_MSK) { 1325 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1326 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1327 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1328 } 1329 } else if (u > server) { 1330 ALOGW("stepServer occurred after track reset"); 1331 u = server; 1332 } 1333 1334 if (u >= userBase + this->frameCount) { 1335 userBase += this->frameCount; 1336 } 1337 1338 user = u; 1339 1340 // Clear flow control error condition as new data has been written/read to/from buffer. 1341 if (flags & CBLK_UNDERRUN_MSK) { 1342 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1343 } 1344 1345 return u; 1346} 1347 1348bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1349{ 1350 if (!tryLock()) { 1351 ALOGW("stepServer() could not lock cblk"); 1352 return false; 1353 } 1354 1355 uint32_t s = server; 1356 1357 s += frameCount; 1358 if (flags & CBLK_DIRECTION_MSK) { 1359 // Mark that we have read the first buffer so that next time stepUser() is called 1360 // we switch to normal obtainBuffer() timeout period 1361 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1362 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1363 } 1364 // It is possible that we receive a flush() 1365 // while the mixer is processing a block: in this case, 1366 // stepServer() is called After the flush() has reset u & s and 1367 // we have s > u 1368 if (s > user) { 1369 ALOGW("stepServer occurred after track reset"); 1370 s = user; 1371 } 1372 } 1373 1374 if (s >= loopEnd) { 1375 ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1376 s = loopStart; 1377 if (--loopCount == 0) { 1378 loopEnd = UINT_MAX; 1379 loopStart = UINT_MAX; 1380 } 1381 } 1382 if (s >= serverBase + this->frameCount) { 1383 serverBase += this->frameCount; 1384 } 1385 1386 server = s; 1387 1388 if (!(flags & CBLK_INVALID_MSK)) { 1389 cv.signal(); 1390 } 1391 lock.unlock(); 1392 return true; 1393} 1394 1395void* audio_track_cblk_t::buffer(uint32_t offset) const 1396{ 1397 return (int8_t *)buffers + (offset - userBase) * frameSize; 1398} 1399 1400uint32_t audio_track_cblk_t::framesAvailable() 1401{ 1402 Mutex::Autolock _l(lock); 1403 return framesAvailable_l(); 1404} 1405 1406uint32_t audio_track_cblk_t::framesAvailable_l() 1407{ 1408 uint32_t u = user; 1409 uint32_t s = server; 1410 1411 if (flags & CBLK_DIRECTION_MSK) { 1412 uint32_t limit = (s < loopStart) ? s : loopStart; 1413 return limit + frameCount - u; 1414 } else { 1415 return frameCount + u - s; 1416 } 1417} 1418 1419uint32_t audio_track_cblk_t::framesReady() 1420{ 1421 uint32_t u = user; 1422 uint32_t s = server; 1423 1424 if (flags & CBLK_DIRECTION_MSK) { 1425 if (u < loopEnd) { 1426 return u - s; 1427 } else { 1428 // do not block on mutex shared with client on AudioFlinger side 1429 if (!tryLock()) { 1430 ALOGW("framesReady() could not lock cblk"); 1431 return 0; 1432 } 1433 uint32_t frames = UINT_MAX; 1434 if (loopCount >= 0) { 1435 frames = (loopEnd - loopStart)*loopCount + u - s; 1436 } 1437 lock.unlock(); 1438 return frames; 1439 } 1440 } else { 1441 return s - u; 1442 } 1443} 1444 1445bool audio_track_cblk_t::tryLock() 1446{ 1447 // the code below simulates lock-with-timeout 1448 // we MUST do this to protect the AudioFlinger server 1449 // as this lock is shared with the client. 1450 status_t err; 1451 1452 err = lock.tryLock(); 1453 if (err == -EBUSY) { // just wait a bit 1454 usleep(1000); 1455 err = lock.tryLock(); 1456 } 1457 if (err != NO_ERROR) { 1458 // probably, the client just died. 1459 return false; 1460 } 1461 return true; 1462} 1463 1464// ------------------------------------------------------------------------- 1465 1466}; // namespace android 1467