AudioTrack.cpp revision c6854100cea4fcd0f20cb2ac8235c02d1849b3a1
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 <hardware/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{ 84} 85 86AudioTrack::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 104AudioTrack::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 122AudioTrack::~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 } 138} 139 140status_t AudioTrack::set( 141 int streamType, 142 uint32_t sampleRate, 143 int format, 144 int channelMask, 145 int frameCount, 146 uint32_t flags, 147 callback_t cbf, 148 void* user, 149 int notificationFrames, 150 const sp<IMemory>& sharedBuffer, 151 bool threadCanCallJava, 152 int sessionId) 153{ 154 155 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 156 157 AutoMutex lock(mLock); 158 if (mAudioTrack != 0) { 159 LOGE("Track already in use"); 160 return INVALID_OPERATION; 161 } 162 163 int afSampleRate; 164 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 165 return NO_INIT; 166 } 167 uint32_t afLatency; 168 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 169 return NO_INIT; 170 } 171 172 // handle default values first. 173 if (streamType == AUDIO_STREAM_DEFAULT) { 174 streamType = AUDIO_STREAM_MUSIC; 175 } 176 if (sampleRate == 0) { 177 sampleRate = afSampleRate; 178 } 179 // these below should probably come from the audioFlinger too... 180 if (format == 0) { 181 format = AUDIO_FORMAT_PCM_16_BIT; 182 } 183 if (channelMask == 0) { 184 channelMask = AUDIO_CHANNEL_OUT_STEREO; 185 } 186 187 // validate parameters 188 if (!audio_is_valid_format(format)) { 189 LOGE("Invalid format"); 190 return BAD_VALUE; 191 } 192 193 // force direct flag if format is not linear PCM 194 if (!audio_is_linear_pcm(format)) { 195 flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT; 196 } 197 198 if (!audio_is_output_channel(channelMask)) { 199 LOGE("Invalid channel mask"); 200 return BAD_VALUE; 201 } 202 uint32_t channelCount = popcount(channelMask); 203 204 audio_io_handle_t output = AudioSystem::getOutput( 205 (audio_stream_type_t)streamType, 206 sampleRate,format, channelMask, 207 (audio_policy_output_flags_t)flags); 208 209 if (output == 0) { 210 LOGE("Could not get audio output for stream type %d", streamType); 211 return BAD_VALUE; 212 } 213 214 mVolume[LEFT] = 1.0f; 215 mVolume[RIGHT] = 1.0f; 216 mSendLevel = 0; 217 mFrameCount = frameCount; 218 mNotificationFramesReq = notificationFrames; 219 mSessionId = sessionId; 220 mAuxEffectId = 0; 221 222 // create the IAudioTrack 223 status_t status = createTrack_l(streamType, 224 sampleRate, 225 (uint32_t)format, 226 (uint32_t)channelMask, 227 frameCount, 228 flags, 229 sharedBuffer, 230 output, 231 true); 232 233 if (status != NO_ERROR) { 234 return status; 235 } 236 237 if (cbf != 0) { 238 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 239 if (mAudioTrackThread == 0) { 240 LOGE("Could not create callback thread"); 241 return NO_INIT; 242 } 243 } 244 245 mStatus = NO_ERROR; 246 247 mStreamType = streamType; 248 mFormat = (uint32_t)format; 249 mChannelMask = (uint32_t)channelMask; 250 mChannelCount = channelCount; 251 mSharedBuffer = sharedBuffer; 252 mMuted = false; 253 mActive = 0; 254 mCbf = cbf; 255 mUserData = user; 256 mLoopCount = 0; 257 mMarkerPosition = 0; 258 mMarkerReached = false; 259 mNewPosition = 0; 260 mUpdatePeriod = 0; 261 mFlags = flags; 262 263 return NO_ERROR; 264} 265 266status_t AudioTrack::initCheck() const 267{ 268 return mStatus; 269} 270 271// ------------------------------------------------------------------------- 272 273uint32_t AudioTrack::latency() const 274{ 275 return mLatency; 276} 277 278int AudioTrack::streamType() const 279{ 280 return mStreamType; 281} 282 283int AudioTrack::format() const 284{ 285 return mFormat; 286} 287 288int AudioTrack::channelCount() const 289{ 290 return mChannelCount; 291} 292 293uint32_t AudioTrack::frameCount() const 294{ 295 return mCblk->frameCount; 296} 297 298int AudioTrack::frameSize() const 299{ 300 if (audio_is_linear_pcm(mFormat)) { 301 return channelCount()*((format() == AUDIO_FORMAT_PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); 302 } else { 303 return sizeof(uint8_t); 304 } 305} 306 307sp<IMemory>& AudioTrack::sharedBuffer() 308{ 309 return mSharedBuffer; 310} 311 312// ------------------------------------------------------------------------- 313 314void AudioTrack::start() 315{ 316 sp<AudioTrackThread> t = mAudioTrackThread; 317 status_t status; 318 319 LOGV("start %p", this); 320 if (t != 0) { 321 if (t->exitPending()) { 322 if (t->requestExitAndWait() == WOULD_BLOCK) { 323 LOGE("AudioTrack::start called from thread"); 324 return; 325 } 326 } 327 t->mLock.lock(); 328 } 329 330 AutoMutex lock(mLock); 331 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 332 // while we are accessing the cblk 333 sp <IAudioTrack> audioTrack = mAudioTrack; 334 sp <IMemory> iMem = mCblkMemory; 335 audio_track_cblk_t* cblk = mCblk; 336 337 if (mActive == 0) { 338 mActive = 1; 339 mNewPosition = cblk->server + mUpdatePeriod; 340 cblk->lock.lock(); 341 cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 342 cblk->waitTimeMs = 0; 343 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 344 if (t != 0) { 345 t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO); 346 } else { 347 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO); 348 } 349 350 LOGV("start %p before lock cblk %p", this, mCblk); 351 if (!(cblk->flags & CBLK_INVALID_MSK)) { 352 cblk->lock.unlock(); 353 status = mAudioTrack->start(); 354 cblk->lock.lock(); 355 if (status == DEAD_OBJECT) { 356 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 357 } 358 } 359 if (cblk->flags & CBLK_INVALID_MSK) { 360 status = restoreTrack_l(cblk, true); 361 } 362 cblk->lock.unlock(); 363 if (status != NO_ERROR) { 364 LOGV("start() failed"); 365 mActive = 0; 366 if (t != 0) { 367 t->requestExit(); 368 } else { 369 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); 370 } 371 } 372 } 373 374 if (t != 0) { 375 t->mLock.unlock(); 376 } 377} 378 379void AudioTrack::stop() 380{ 381 sp<AudioTrackThread> t = mAudioTrackThread; 382 383 LOGV("stop %p", this); 384 if (t != 0) { 385 t->mLock.lock(); 386 } 387 388 AutoMutex lock(mLock); 389 if (mActive == 1) { 390 mActive = 0; 391 mCblk->cv.signal(); 392 mAudioTrack->stop(); 393 // Cancel loops (If we are in the middle of a loop, playback 394 // would not stop until loopCount reaches 0). 395 setLoop_l(0, 0, 0); 396 // the playback head position will reset to 0, so if a marker is set, we need 397 // to activate it again 398 mMarkerReached = false; 399 // Force flush if a shared buffer is used otherwise audioflinger 400 // will not stop before end of buffer is reached. 401 if (mSharedBuffer != 0) { 402 flush_l(); 403 } 404 if (t != 0) { 405 t->requestExit(); 406 } else { 407 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); 408 } 409 } 410 411 if (t != 0) { 412 t->mLock.unlock(); 413 } 414} 415 416bool AudioTrack::stopped() const 417{ 418 return !mActive; 419} 420 421void AudioTrack::flush() 422{ 423 AutoMutex lock(mLock); 424 flush_l(); 425} 426 427// must be called with mLock held 428void AudioTrack::flush_l() 429{ 430 LOGV("flush"); 431 432 // clear playback marker and periodic update counter 433 mMarkerPosition = 0; 434 mMarkerReached = false; 435 mUpdatePeriod = 0; 436 437 if (!mActive) { 438 mAudioTrack->flush(); 439 // Release AudioTrack callback thread in case it was waiting for new buffers 440 // in AudioTrack::obtainBuffer() 441 mCblk->cv.signal(); 442 } 443} 444 445void AudioTrack::pause() 446{ 447 LOGV("pause"); 448 AutoMutex lock(mLock); 449 if (mActive == 1) { 450 mActive = 0; 451 mAudioTrack->pause(); 452 } 453} 454 455void AudioTrack::mute(bool e) 456{ 457 mAudioTrack->mute(e); 458 mMuted = e; 459} 460 461bool AudioTrack::muted() const 462{ 463 return mMuted; 464} 465 466status_t AudioTrack::setVolume(float left, float right) 467{ 468 if (left > 1.0f || right > 1.0f) { 469 return BAD_VALUE; 470 } 471 472 AutoMutex lock(mLock); 473 mVolume[LEFT] = left; 474 mVolume[RIGHT] = right; 475 476 // write must be atomic 477 mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000); 478 479 return NO_ERROR; 480} 481 482void AudioTrack::getVolume(float* left, float* right) 483{ 484 if (left != NULL) { 485 *left = mVolume[LEFT]; 486 } 487 if (right != NULL) { 488 *right = mVolume[RIGHT]; 489 } 490} 491 492status_t AudioTrack::setAuxEffectSendLevel(float level) 493{ 494 LOGV("setAuxEffectSendLevel(%f)", level); 495 if (level > 1.0f) { 496 return BAD_VALUE; 497 } 498 AutoMutex lock(mLock); 499 500 mSendLevel = level; 501 502 mCblk->sendLevel = uint16_t(level * 0x1000); 503 504 return NO_ERROR; 505} 506 507void AudioTrack::getAuxEffectSendLevel(float* level) 508{ 509 if (level != NULL) { 510 *level = mSendLevel; 511 } 512} 513 514status_t AudioTrack::setSampleRate(int rate) 515{ 516 int afSamplingRate; 517 518 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 519 return NO_INIT; 520 } 521 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 522 if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; 523 524 AutoMutex lock(mLock); 525 mCblk->sampleRate = rate; 526 return NO_ERROR; 527} 528 529uint32_t AudioTrack::getSampleRate() 530{ 531 AutoMutex lock(mLock); 532 return mCblk->sampleRate; 533} 534 535status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 536{ 537 AutoMutex lock(mLock); 538 return setLoop_l(loopStart, loopEnd, loopCount); 539} 540 541// must be called with mLock held 542status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 543{ 544 audio_track_cblk_t* cblk = mCblk; 545 546 Mutex::Autolock _l(cblk->lock); 547 548 if (loopCount == 0) { 549 cblk->loopStart = UINT_MAX; 550 cblk->loopEnd = UINT_MAX; 551 cblk->loopCount = 0; 552 mLoopCount = 0; 553 return NO_ERROR; 554 } 555 556 if (loopStart >= loopEnd || 557 loopEnd - loopStart > cblk->frameCount || 558 cblk->server > loopStart) { 559 LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); 560 return BAD_VALUE; 561 } 562 563 if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { 564 LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", 565 loopStart, loopEnd, cblk->frameCount); 566 return BAD_VALUE; 567 } 568 569 cblk->loopStart = loopStart; 570 cblk->loopEnd = loopEnd; 571 cblk->loopCount = loopCount; 572 mLoopCount = loopCount; 573 574 return NO_ERROR; 575} 576 577status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) 578{ 579 AutoMutex lock(mLock); 580 if (loopStart != 0) { 581 *loopStart = mCblk->loopStart; 582 } 583 if (loopEnd != 0) { 584 *loopEnd = mCblk->loopEnd; 585 } 586 if (loopCount != 0) { 587 if (mCblk->loopCount < 0) { 588 *loopCount = -1; 589 } else { 590 *loopCount = mCblk->loopCount; 591 } 592 } 593 594 return NO_ERROR; 595} 596 597status_t AudioTrack::setMarkerPosition(uint32_t marker) 598{ 599 if (mCbf == 0) return INVALID_OPERATION; 600 601 mMarkerPosition = marker; 602 mMarkerReached = false; 603 604 return NO_ERROR; 605} 606 607status_t AudioTrack::getMarkerPosition(uint32_t *marker) 608{ 609 if (marker == 0) return BAD_VALUE; 610 611 *marker = mMarkerPosition; 612 613 return NO_ERROR; 614} 615 616status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 617{ 618 if (mCbf == 0) return INVALID_OPERATION; 619 620 uint32_t curPosition; 621 getPosition(&curPosition); 622 mNewPosition = curPosition + updatePeriod; 623 mUpdatePeriod = updatePeriod; 624 625 return NO_ERROR; 626} 627 628status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) 629{ 630 if (updatePeriod == 0) return BAD_VALUE; 631 632 *updatePeriod = mUpdatePeriod; 633 634 return NO_ERROR; 635} 636 637status_t AudioTrack::setPosition(uint32_t position) 638{ 639 AutoMutex lock(mLock); 640 Mutex::Autolock _l(mCblk->lock); 641 642 if (!stopped()) return INVALID_OPERATION; 643 644 if (position > mCblk->user) return BAD_VALUE; 645 646 mCblk->server = position; 647 android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); 648 649 return NO_ERROR; 650} 651 652status_t AudioTrack::getPosition(uint32_t *position) 653{ 654 if (position == 0) return BAD_VALUE; 655 AutoMutex lock(mLock); 656 *position = mCblk->server; 657 658 return NO_ERROR; 659} 660 661status_t AudioTrack::reload() 662{ 663 AutoMutex lock(mLock); 664 665 if (!stopped()) return INVALID_OPERATION; 666 667 flush_l(); 668 669 mCblk->stepUser(mCblk->frameCount); 670 671 return NO_ERROR; 672} 673 674audio_io_handle_t AudioTrack::getOutput() 675{ 676 AutoMutex lock(mLock); 677 return getOutput_l(); 678} 679 680// must be called with mLock held 681audio_io_handle_t AudioTrack::getOutput_l() 682{ 683 return AudioSystem::getOutput((audio_stream_type_t)mStreamType, 684 mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags); 685} 686 687int AudioTrack::getSessionId() 688{ 689 return mSessionId; 690} 691 692status_t AudioTrack::attachAuxEffect(int effectId) 693{ 694 LOGV("attachAuxEffect(%d)", effectId); 695 status_t status = mAudioTrack->attachAuxEffect(effectId); 696 if (status == NO_ERROR) { 697 mAuxEffectId = effectId; 698 } 699 return status; 700} 701 702// ------------------------------------------------------------------------- 703 704// must be called with mLock held 705status_t AudioTrack::createTrack_l( 706 int streamType, 707 uint32_t sampleRate, 708 uint32_t format, 709 uint32_t channelMask, 710 int frameCount, 711 uint32_t flags, 712 const sp<IMemory>& sharedBuffer, 713 audio_io_handle_t output, 714 bool enforceFrameCount) 715{ 716 status_t status; 717 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 718 if (audioFlinger == 0) { 719 LOGE("Could not get audioflinger"); 720 return NO_INIT; 721 } 722 723 int afSampleRate; 724 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 725 return NO_INIT; 726 } 727 int afFrameCount; 728 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 729 return NO_INIT; 730 } 731 uint32_t afLatency; 732 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 733 return NO_INIT; 734 } 735 736 mNotificationFramesAct = mNotificationFramesReq; 737 if (!audio_is_linear_pcm(format)) { 738 if (sharedBuffer != 0) { 739 frameCount = sharedBuffer->size(); 740 } 741 } else { 742 // Ensure that buffer depth covers at least audio hardware latency 743 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 744 if (minBufCount < 2) minBufCount = 2; 745 746 int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 747 748 if (sharedBuffer == 0) { 749 if (frameCount == 0) { 750 frameCount = minFrameCount; 751 } 752 if (mNotificationFramesAct == 0) { 753 mNotificationFramesAct = frameCount/2; 754 } 755 // Make sure that application is notified with sufficient margin 756 // before underrun 757 if (mNotificationFramesAct > (uint32_t)frameCount/2) { 758 mNotificationFramesAct = frameCount/2; 759 } 760 if (frameCount < minFrameCount) { 761 if (enforceFrameCount) { 762 LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); 763 return BAD_VALUE; 764 } else { 765 frameCount = minFrameCount; 766 } 767 } 768 } else { 769 // Ensure that buffer alignment matches channelcount 770 int channelCount = popcount(channelMask); 771 if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { 772 LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); 773 return BAD_VALUE; 774 } 775 frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); 776 } 777 } 778 779 sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), 780 streamType, 781 sampleRate, 782 format, 783 channelMask, 784 frameCount, 785 ((uint16_t)flags) << 16, 786 sharedBuffer, 787 output, 788 &mSessionId, 789 &status); 790 791 if (track == 0) { 792 LOGE("AudioFlinger could not create track, status: %d", status); 793 return status; 794 } 795 sp<IMemory> cblk = track->getCblk(); 796 if (cblk == 0) { 797 LOGE("Could not get control block"); 798 return NO_INIT; 799 } 800 mAudioTrack.clear(); 801 mAudioTrack = track; 802 mCblkMemory.clear(); 803 mCblkMemory = cblk; 804 mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); 805 android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); 806 if (sharedBuffer == 0) { 807 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 808 } else { 809 mCblk->buffers = sharedBuffer->pointer(); 810 // Force buffer full condition as data is already present in shared memory 811 mCblk->stepUser(mCblk->frameCount); 812 } 813 814 mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000); 815 mCblk->sendLevel = uint16_t(mSendLevel * 0x1000); 816 mAudioTrack->attachAuxEffect(mAuxEffectId); 817 mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; 818 mCblk->waitTimeMs = 0; 819 mRemainingFrames = mNotificationFramesAct; 820 mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; 821 return NO_ERROR; 822} 823 824status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 825{ 826 AutoMutex lock(mLock); 827 int active; 828 status_t result; 829 audio_track_cblk_t* cblk = mCblk; 830 uint32_t framesReq = audioBuffer->frameCount; 831 uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; 832 833 audioBuffer->frameCount = 0; 834 audioBuffer->size = 0; 835 836 uint32_t framesAvail = cblk->framesAvailable(); 837 838 cblk->lock.lock(); 839 if (cblk->flags & CBLK_INVALID_MSK) { 840 goto create_new_track; 841 } 842 cblk->lock.unlock(); 843 844 if (framesAvail == 0) { 845 cblk->lock.lock(); 846 goto start_loop_here; 847 while (framesAvail == 0) { 848 active = mActive; 849 if (UNLIKELY(!active)) { 850 LOGV("Not active and NO_MORE_BUFFERS"); 851 cblk->lock.unlock(); 852 return NO_MORE_BUFFERS; 853 } 854 if (UNLIKELY(!waitCount)) { 855 cblk->lock.unlock(); 856 return WOULD_BLOCK; 857 } 858 if (!(cblk->flags & CBLK_INVALID_MSK)) { 859 mLock.unlock(); 860 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 861 cblk->lock.unlock(); 862 mLock.lock(); 863 if (mActive == 0) { 864 return status_t(STOPPED); 865 } 866 cblk->lock.lock(); 867 } 868 869 if (cblk->flags & CBLK_INVALID_MSK) { 870 goto create_new_track; 871 } 872 if (__builtin_expect(result!=NO_ERROR, false)) { 873 cblk->waitTimeMs += waitTimeMs; 874 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { 875 // timing out when a loop has been set and we have already written upto loop end 876 // is a normal condition: no need to wake AudioFlinger up. 877 if (cblk->user < cblk->loopEnd) { 878 LOGW( "obtainBuffer timed out (is the CPU pegged?) %p " 879 "user=%08x, server=%08x", this, cblk->user, cblk->server); 880 //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 881 cblk->lock.unlock(); 882 result = mAudioTrack->start(); 883 cblk->lock.lock(); 884 if (result == DEAD_OBJECT) { 885 android_atomic_or(CBLK_INVALID_ON, &cblk->flags); 886create_new_track: 887 result = restoreTrack_l(cblk, false); 888 } 889 if (result != NO_ERROR) { 890 LOGW("obtainBuffer create Track error %d", result); 891 cblk->lock.unlock(); 892 return result; 893 } 894 } 895 cblk->waitTimeMs = 0; 896 } 897 898 if (--waitCount == 0) { 899 cblk->lock.unlock(); 900 return TIMED_OUT; 901 } 902 } 903 // read the server count again 904 start_loop_here: 905 framesAvail = cblk->framesAvailable_l(); 906 } 907 cblk->lock.unlock(); 908 } 909 910 // restart track if it was disabled by audioflinger due to previous underrun 911 if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) { 912 android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); 913 LOGW("obtainBuffer() track %p disabled, restarting", this); 914 mAudioTrack->start(); 915 } 916 917 cblk->waitTimeMs = 0; 918 919 if (framesReq > framesAvail) { 920 framesReq = framesAvail; 921 } 922 923 uint32_t u = cblk->user; 924 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 925 926 if (u + framesReq > bufferEnd) { 927 framesReq = bufferEnd - u; 928 } 929 930 audioBuffer->flags = mMuted ? Buffer::MUTE : 0; 931 audioBuffer->channelCount = mChannelCount; 932 audioBuffer->frameCount = framesReq; 933 audioBuffer->size = framesReq * cblk->frameSize; 934 if (audio_is_linear_pcm(mFormat)) { 935 audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; 936 } else { 937 audioBuffer->format = mFormat; 938 } 939 audioBuffer->raw = (int8_t *)cblk->buffer(u); 940 active = mActive; 941 return active ? status_t(NO_ERROR) : status_t(STOPPED); 942} 943 944void AudioTrack::releaseBuffer(Buffer* audioBuffer) 945{ 946 AutoMutex lock(mLock); 947 mCblk->stepUser(audioBuffer->frameCount); 948} 949 950// ------------------------------------------------------------------------- 951 952ssize_t AudioTrack::write(const void* buffer, size_t userSize) 953{ 954 955 if (mSharedBuffer != 0) return INVALID_OPERATION; 956 957 if (ssize_t(userSize) < 0) { 958 // sanity-check. user is most-likely passing an error code. 959 LOGE("AudioTrack::write(buffer=%p, size=%u (%d)", 960 buffer, userSize, userSize); 961 return BAD_VALUE; 962 } 963 964 LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); 965 966 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 967 // while we are accessing the cblk 968 mLock.lock(); 969 sp <IAudioTrack> audioTrack = mAudioTrack; 970 sp <IMemory> iMem = mCblkMemory; 971 mLock.unlock(); 972 973 ssize_t written = 0; 974 const int8_t *src = (const int8_t *)buffer; 975 Buffer audioBuffer; 976 size_t frameSz = (size_t)frameSize(); 977 978 do { 979 audioBuffer.frameCount = userSize/frameSz; 980 981 // Calling obtainBuffer() with a negative wait count causes 982 // an (almost) infinite wait time. 983 status_t err = obtainBuffer(&audioBuffer, -1); 984 if (err < 0) { 985 // out of buffers, return #bytes written 986 if (err == status_t(NO_MORE_BUFFERS)) 987 break; 988 return ssize_t(err); 989 } 990 991 size_t toWrite; 992 993 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 994 // Divide capacity by 2 to take expansion into account 995 toWrite = audioBuffer.size>>1; 996 // 8 to 16 bit conversion 997 int count = toWrite; 998 int16_t *dst = (int16_t *)(audioBuffer.i8); 999 while(count--) { 1000 *dst++ = (int16_t)(*src++^0x80) << 8; 1001 } 1002 } else { 1003 toWrite = audioBuffer.size; 1004 memcpy(audioBuffer.i8, src, toWrite); 1005 src += toWrite; 1006 } 1007 userSize -= toWrite; 1008 written += toWrite; 1009 1010 releaseBuffer(&audioBuffer); 1011 } while (userSize >= frameSz); 1012 1013 return written; 1014} 1015 1016// ------------------------------------------------------------------------- 1017 1018bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1019{ 1020 Buffer audioBuffer; 1021 uint32_t frames; 1022 size_t writtenSize; 1023 1024 mLock.lock(); 1025 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1026 // while we are accessing the cblk 1027 sp <IAudioTrack> audioTrack = mAudioTrack; 1028 sp <IMemory> iMem = mCblkMemory; 1029 audio_track_cblk_t* cblk = mCblk; 1030 mLock.unlock(); 1031 1032 // Manage underrun callback 1033 if (mActive && (cblk->framesAvailable() == cblk->frameCount)) { 1034 LOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); 1035 if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { 1036 mCbf(EVENT_UNDERRUN, mUserData, 0); 1037 if (cblk->server == cblk->frameCount) { 1038 mCbf(EVENT_BUFFER_END, mUserData, 0); 1039 } 1040 if (mSharedBuffer != 0) return false; 1041 } 1042 } 1043 1044 // Manage loop end callback 1045 while (mLoopCount > cblk->loopCount) { 1046 int loopCount = -1; 1047 mLoopCount--; 1048 if (mLoopCount >= 0) loopCount = mLoopCount; 1049 1050 mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); 1051 } 1052 1053 // Manage marker callback 1054 if (!mMarkerReached && (mMarkerPosition > 0)) { 1055 if (cblk->server >= mMarkerPosition) { 1056 mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); 1057 mMarkerReached = true; 1058 } 1059 } 1060 1061 // Manage new position callback 1062 if (mUpdatePeriod > 0) { 1063 while (cblk->server >= mNewPosition) { 1064 mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); 1065 mNewPosition += mUpdatePeriod; 1066 } 1067 } 1068 1069 // If Shared buffer is used, no data is requested from client. 1070 if (mSharedBuffer != 0) { 1071 frames = 0; 1072 } else { 1073 frames = mRemainingFrames; 1074 } 1075 1076 do { 1077 1078 audioBuffer.frameCount = frames; 1079 1080 // Calling obtainBuffer() with a wait count of 1 1081 // limits wait time to WAIT_PERIOD_MS. This prevents from being 1082 // stuck here not being able to handle timed events (position, markers, loops). 1083 status_t err = obtainBuffer(&audioBuffer, 1); 1084 if (err < NO_ERROR) { 1085 if (err != TIMED_OUT) { 1086 LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); 1087 return false; 1088 } 1089 break; 1090 } 1091 if (err == status_t(STOPPED)) return false; 1092 1093 // Divide buffer size by 2 to take into account the expansion 1094 // due to 8 to 16 bit conversion: the callback must fill only half 1095 // of the destination buffer 1096 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1097 audioBuffer.size >>= 1; 1098 } 1099 1100 size_t reqSize = audioBuffer.size; 1101 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1102 writtenSize = audioBuffer.size; 1103 1104 // Sanity check on returned size 1105 if (ssize_t(writtenSize) <= 0) { 1106 // The callback is done filling buffers 1107 // Keep this thread going to handle timed events and 1108 // still try to get more data in intervals of WAIT_PERIOD_MS 1109 // but don't just loop and block the CPU, so wait 1110 usleep(WAIT_PERIOD_MS*1000); 1111 break; 1112 } 1113 if (writtenSize > reqSize) writtenSize = reqSize; 1114 1115 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { 1116 // 8 to 16 bit conversion 1117 const int8_t *src = audioBuffer.i8 + writtenSize-1; 1118 int count = writtenSize; 1119 int16_t *dst = audioBuffer.i16 + writtenSize-1; 1120 while(count--) { 1121 *dst-- = (int16_t)(*src--^0x80) << 8; 1122 } 1123 writtenSize <<= 1; 1124 } 1125 1126 audioBuffer.size = writtenSize; 1127 // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for 1128 // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of 1129 // 16 bit. 1130 audioBuffer.frameCount = writtenSize/mCblk->frameSize; 1131 1132 frames -= audioBuffer.frameCount; 1133 1134 releaseBuffer(&audioBuffer); 1135 } 1136 while (frames); 1137 1138 if (frames == 0) { 1139 mRemainingFrames = mNotificationFramesAct; 1140 } else { 1141 mRemainingFrames = frames; 1142 } 1143 return true; 1144} 1145 1146// must be called with mLock and cblk.lock held. Callers must also hold strong references on 1147// the IAudioTrack and IMemory in case they are recreated here. 1148// If the IAudioTrack is successfully restored, the cblk pointer is updated 1149status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) 1150{ 1151 status_t result; 1152 1153 if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { 1154 LOGW("dead IAudioTrack, creating a new one from %s", 1155 fromStart ? "start()" : "obtainBuffer()"); 1156 1157 // signal old cblk condition so that other threads waiting for available buffers stop 1158 // waiting now 1159 cblk->cv.broadcast(); 1160 cblk->lock.unlock(); 1161 1162 // if the new IAudioTrack is created, createTrack_l() will modify the 1163 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1164 // It will also delete the strong references on previous IAudioTrack and IMemory 1165 result = createTrack_l(mStreamType, 1166 cblk->sampleRate, 1167 mFormat, 1168 mChannelMask, 1169 mFrameCount, 1170 mFlags, 1171 mSharedBuffer, 1172 getOutput_l(), 1173 false); 1174 1175 if (result == NO_ERROR) { 1176 // restore write index and set other indexes to reflect empty buffer status 1177 mCblk->user = cblk->user; 1178 mCblk->server = cblk->user; 1179 mCblk->userBase = cblk->user; 1180 mCblk->serverBase = cblk->user; 1181 // restore loop: this is not guaranteed to succeed if new frame count is not 1182 // compatible with loop length 1183 setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); 1184 if (!fromStart) { 1185 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1186 } 1187 if (mActive) { 1188 result = mAudioTrack->start(); 1189 } 1190 if (fromStart && result == NO_ERROR) { 1191 mNewPosition = mCblk->server + mUpdatePeriod; 1192 } 1193 } 1194 if (result != NO_ERROR) { 1195 mActive = false; 1196 } 1197 1198 // signal old cblk condition for other threads waiting for restore completion 1199 android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); 1200 cblk->cv.broadcast(); 1201 } else { 1202 if (!(cblk->flags & CBLK_RESTORED_MSK)) { 1203 LOGW("dead IAudioTrack, waiting for a new one"); 1204 mLock.unlock(); 1205 result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); 1206 cblk->lock.unlock(); 1207 mLock.lock(); 1208 } else { 1209 LOGW("dead IAudioTrack, already restored"); 1210 result = NO_ERROR; 1211 cblk->lock.unlock(); 1212 } 1213 if (result != NO_ERROR || mActive == 0) { 1214 result = status_t(STOPPED); 1215 } 1216 } 1217 LOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", 1218 result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); 1219 1220 if (result == NO_ERROR) { 1221 // from now on we switch to the newly created cblk 1222 cblk = mCblk; 1223 } 1224 cblk->lock.lock(); 1225 1226 LOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result); 1227 1228 return result; 1229} 1230 1231status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1232{ 1233 1234 const size_t SIZE = 256; 1235 char buffer[SIZE]; 1236 String8 result; 1237 1238 result.append(" AudioTrack::dump\n"); 1239 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); 1240 result.append(buffer); 1241 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); 1242 result.append(buffer); 1243 snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); 1244 result.append(buffer); 1245 snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); 1246 result.append(buffer); 1247 ::write(fd, result.string(), result.size()); 1248 return NO_ERROR; 1249} 1250 1251// ========================================================================= 1252 1253AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1254 : Thread(bCanCallJava), mReceiver(receiver) 1255{ 1256} 1257 1258bool AudioTrack::AudioTrackThread::threadLoop() 1259{ 1260 return mReceiver.processAudioBuffer(this); 1261} 1262 1263status_t AudioTrack::AudioTrackThread::readyToRun() 1264{ 1265 return NO_ERROR; 1266} 1267 1268void AudioTrack::AudioTrackThread::onFirstRef() 1269{ 1270} 1271 1272// ========================================================================= 1273 1274 1275audio_track_cblk_t::audio_track_cblk_t() 1276 : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), 1277 userBase(0), serverBase(0), buffers(0), frameCount(0), 1278 loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), 1279 sendLevel(0), flags(0) 1280{ 1281} 1282 1283uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) 1284{ 1285 uint32_t u = this->user; 1286 1287 u += frameCount; 1288 // Ensure that user is never ahead of server for AudioRecord 1289 if (flags & CBLK_DIRECTION_MSK) { 1290 // If stepServer() has been called once, switch to normal obtainBuffer() timeout period 1291 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { 1292 bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; 1293 } 1294 } else if (u > this->server) { 1295 LOGW("stepServer occured after track reset"); 1296 u = this->server; 1297 } 1298 1299 if (u >= userBase + this->frameCount) { 1300 userBase += this->frameCount; 1301 } 1302 1303 this->user = u; 1304 1305 // Clear flow control error condition as new data has been written/read to/from buffer. 1306 if (flags & CBLK_UNDERRUN_MSK) { 1307 android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); 1308 } 1309 1310 return u; 1311} 1312 1313bool audio_track_cblk_t::stepServer(uint32_t frameCount) 1314{ 1315 if (!tryLock()) { 1316 LOGW("stepServer() could not lock cblk"); 1317 return false; 1318 } 1319 1320 uint32_t s = this->server; 1321 1322 s += frameCount; 1323 if (flags & CBLK_DIRECTION_MSK) { 1324 // Mark that we have read the first buffer so that next time stepUser() is called 1325 // we switch to normal obtainBuffer() timeout period 1326 if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { 1327 bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; 1328 } 1329 // It is possible that we receive a flush() 1330 // while the mixer is processing a block: in this case, 1331 // stepServer() is called After the flush() has reset u & s and 1332 // we have s > u 1333 if (s > this->user) { 1334 LOGW("stepServer occured after track reset"); 1335 s = this->user; 1336 } 1337 } 1338 1339 if (s >= loopEnd) { 1340 LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); 1341 s = loopStart; 1342 if (--loopCount == 0) { 1343 loopEnd = UINT_MAX; 1344 loopStart = UINT_MAX; 1345 } 1346 } 1347 if (s >= serverBase + this->frameCount) { 1348 serverBase += this->frameCount; 1349 } 1350 1351 this->server = s; 1352 1353 if (!(flags & CBLK_INVALID_MSK)) { 1354 cv.signal(); 1355 } 1356 lock.unlock(); 1357 return true; 1358} 1359 1360void* audio_track_cblk_t::buffer(uint32_t offset) const 1361{ 1362 return (int8_t *)this->buffers + (offset - userBase) * this->frameSize; 1363} 1364 1365uint32_t audio_track_cblk_t::framesAvailable() 1366{ 1367 Mutex::Autolock _l(lock); 1368 return framesAvailable_l(); 1369} 1370 1371uint32_t audio_track_cblk_t::framesAvailable_l() 1372{ 1373 uint32_t u = this->user; 1374 uint32_t s = this->server; 1375 1376 if (flags & CBLK_DIRECTION_MSK) { 1377 uint32_t limit = (s < loopStart) ? s : loopStart; 1378 return limit + frameCount - u; 1379 } else { 1380 return frameCount + u - s; 1381 } 1382} 1383 1384uint32_t audio_track_cblk_t::framesReady() 1385{ 1386 uint32_t u = this->user; 1387 uint32_t s = this->server; 1388 1389 if (flags & CBLK_DIRECTION_MSK) { 1390 if (u < loopEnd) { 1391 return u - s; 1392 } else { 1393 // do not block on mutex shared with client on AudioFlinger side 1394 if (!tryLock()) { 1395 LOGW("framesReady() could not lock cblk"); 1396 return 0; 1397 } 1398 uint32_t frames = UINT_MAX; 1399 if (loopCount >= 0) { 1400 frames = (loopEnd - loopStart)*loopCount + u - s; 1401 } 1402 lock.unlock(); 1403 return frames; 1404 } 1405 } else { 1406 return s - u; 1407 } 1408} 1409 1410bool audio_track_cblk_t::tryLock() 1411{ 1412 // the code below simulates lock-with-timeout 1413 // we MUST do this to protect the AudioFlinger server 1414 // as this lock is shared with the client. 1415 status_t err; 1416 1417 err = lock.tryLock(); 1418 if (err == -EBUSY) { // just wait a bit 1419 usleep(1000); 1420 err = lock.tryLock(); 1421 } 1422 if (err != NO_ERROR) { 1423 // probably, the client just died. 1424 return false; 1425 } 1426 return true; 1427} 1428 1429// ------------------------------------------------------------------------- 1430 1431}; // namespace android 1432 1433