AudioTrack.cpp revision c56f3426099a3cf2d07ccff8886050c7fbce140f
1/* 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 <math.h> 23#include <sys/resource.h> 24#include <audio_utils/primitives.h> 25#include <binder/IPCThreadState.h> 26#include <media/AudioTrack.h> 27#include <utils/Log.h> 28#include <private/media/AudioTrackShared.h> 29#include <media/IAudioFlinger.h> 30 31#define WAIT_PERIOD_MS 10 32#define WAIT_STREAM_END_TIMEOUT_SEC 120 33 34 35namespace android { 36// --------------------------------------------------------------------------- 37 38// static 39status_t AudioTrack::getMinFrameCount( 40 size_t* frameCount, 41 audio_stream_type_t streamType, 42 uint32_t sampleRate) 43{ 44 if (frameCount == NULL) { 45 return BAD_VALUE; 46 } 47 48 // FIXME merge with similar code in createTrack_l(), except we're missing 49 // some information here that is available in createTrack_l(): 50 // audio_io_handle_t output 51 // audio_format_t format 52 // audio_channel_mask_t channelMask 53 // audio_output_flags_t flags 54 uint32_t afSampleRate; 55 status_t status; 56 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 57 if (status != NO_ERROR) { 58 ALOGE("Unable to query output sample rate for stream type %d; status %d", 59 streamType, status); 60 return status; 61 } 62 size_t afFrameCount; 63 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 64 if (status != NO_ERROR) { 65 ALOGE("Unable to query output frame count for stream type %d; status %d", 66 streamType, status); 67 return status; 68 } 69 uint32_t afLatency; 70 status = AudioSystem::getOutputLatency(&afLatency, streamType); 71 if (status != NO_ERROR) { 72 ALOGE("Unable to query output latency for stream type %d; status %d", 73 streamType, status); 74 return status; 75 } 76 77 // Ensure that buffer depth covers at least audio hardware latency 78 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 79 if (minBufCount < 2) { 80 minBufCount = 2; 81 } 82 83 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 84 afFrameCount * minBufCount * sampleRate / afSampleRate; 85 // The formula above should always produce a non-zero value, but return an error 86 // in the unlikely event that it does not, as that's part of the API contract. 87 if (*frameCount == 0) { 88 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 89 streamType, sampleRate); 90 return BAD_VALUE; 91 } 92 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 93 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 94 return NO_ERROR; 95} 96 97// --------------------------------------------------------------------------- 98 99AudioTrack::AudioTrack() 100 : mStatus(NO_INIT), 101 mIsTimed(false), 102 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 103 mPreviousSchedulingGroup(SP_DEFAULT), 104 mPausedPosition(0) 105{ 106} 107 108AudioTrack::AudioTrack( 109 audio_stream_type_t streamType, 110 uint32_t sampleRate, 111 audio_format_t format, 112 audio_channel_mask_t channelMask, 113 size_t frameCount, 114 audio_output_flags_t flags, 115 callback_t cbf, 116 void* user, 117 uint32_t notificationFrames, 118 int sessionId, 119 transfer_type transferType, 120 const audio_offload_info_t *offloadInfo, 121 int uid, 122 pid_t pid) 123 : mStatus(NO_INIT), 124 mIsTimed(false), 125 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 126 mPreviousSchedulingGroup(SP_DEFAULT), 127 mPausedPosition(0) 128{ 129 mStatus = set(streamType, sampleRate, format, channelMask, 130 frameCount, flags, cbf, user, notificationFrames, 131 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 132 offloadInfo, uid, pid); 133} 134 135AudioTrack::AudioTrack( 136 audio_stream_type_t streamType, 137 uint32_t sampleRate, 138 audio_format_t format, 139 audio_channel_mask_t channelMask, 140 const sp<IMemory>& sharedBuffer, 141 audio_output_flags_t flags, 142 callback_t cbf, 143 void* user, 144 uint32_t notificationFrames, 145 int sessionId, 146 transfer_type transferType, 147 const audio_offload_info_t *offloadInfo, 148 int uid, 149 pid_t pid) 150 : mStatus(NO_INIT), 151 mIsTimed(false), 152 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 153 mPreviousSchedulingGroup(SP_DEFAULT), 154 mPausedPosition(0) 155{ 156 mStatus = set(streamType, sampleRate, format, channelMask, 157 0 /*frameCount*/, flags, cbf, user, notificationFrames, 158 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 159 uid, pid); 160} 161 162AudioTrack::~AudioTrack() 163{ 164 if (mStatus == NO_ERROR) { 165 // Make sure that callback function exits in the case where 166 // it is looping on buffer full condition in obtainBuffer(). 167 // Otherwise the callback thread will never exit. 168 stop(); 169 if (mAudioTrackThread != 0) { 170 mProxy->interrupt(); 171 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 172 mAudioTrackThread->requestExitAndWait(); 173 mAudioTrackThread.clear(); 174 } 175 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 176 mAudioTrack.clear(); 177 IPCThreadState::self()->flushCommands(); 178 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 179 IPCThreadState::self()->getCallingPid(), mClientPid); 180 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 181 } 182} 183 184status_t AudioTrack::set( 185 audio_stream_type_t streamType, 186 uint32_t sampleRate, 187 audio_format_t format, 188 audio_channel_mask_t channelMask, 189 size_t frameCount, 190 audio_output_flags_t flags, 191 callback_t cbf, 192 void* user, 193 uint32_t notificationFrames, 194 const sp<IMemory>& sharedBuffer, 195 bool threadCanCallJava, 196 int sessionId, 197 transfer_type transferType, 198 const audio_offload_info_t *offloadInfo, 199 int uid, 200 pid_t pid) 201{ 202 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 203 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 204 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 205 sessionId, transferType); 206 207 switch (transferType) { 208 case TRANSFER_DEFAULT: 209 if (sharedBuffer != 0) { 210 transferType = TRANSFER_SHARED; 211 } else if (cbf == NULL || threadCanCallJava) { 212 transferType = TRANSFER_SYNC; 213 } else { 214 transferType = TRANSFER_CALLBACK; 215 } 216 break; 217 case TRANSFER_CALLBACK: 218 if (cbf == NULL || sharedBuffer != 0) { 219 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 220 return BAD_VALUE; 221 } 222 break; 223 case TRANSFER_OBTAIN: 224 case TRANSFER_SYNC: 225 if (sharedBuffer != 0) { 226 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 227 return BAD_VALUE; 228 } 229 break; 230 case TRANSFER_SHARED: 231 if (sharedBuffer == 0) { 232 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 233 return BAD_VALUE; 234 } 235 break; 236 default: 237 ALOGE("Invalid transfer type %d", transferType); 238 return BAD_VALUE; 239 } 240 mSharedBuffer = sharedBuffer; 241 mTransfer = transferType; 242 243 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 244 sharedBuffer->size()); 245 246 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 247 248 AutoMutex lock(mLock); 249 250 // invariant that mAudioTrack != 0 is true only after set() returns successfully 251 if (mAudioTrack != 0) { 252 ALOGE("Track already in use"); 253 return INVALID_OPERATION; 254 } 255 256 // handle default values first. 257 if (streamType == AUDIO_STREAM_DEFAULT) { 258 streamType = AUDIO_STREAM_MUSIC; 259 } 260 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 261 ALOGE("Invalid stream type %d", streamType); 262 return BAD_VALUE; 263 } 264 mStreamType = streamType; 265 266 status_t status; 267 if (sampleRate == 0) { 268 status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 269 if (status != NO_ERROR) { 270 ALOGE("Could not get output sample rate for stream type %d; status %d", 271 streamType, status); 272 return status; 273 } 274 } 275 mSampleRate = sampleRate; 276 277 // these below should probably come from the audioFlinger too... 278 if (format == AUDIO_FORMAT_DEFAULT) { 279 format = AUDIO_FORMAT_PCM_16_BIT; 280 } 281 282 // validate parameters 283 if (!audio_is_valid_format(format)) { 284 ALOGE("Invalid format %#x", format); 285 return BAD_VALUE; 286 } 287 mFormat = format; 288 289 if (!audio_is_output_channel(channelMask)) { 290 ALOGE("Invalid channel mask %#x", channelMask); 291 return BAD_VALUE; 292 } 293 mChannelMask = channelMask; 294 uint32_t channelCount = popcount(channelMask); 295 mChannelCount = channelCount; 296 297 // AudioFlinger does not currently support 8-bit data in shared memory 298 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 299 ALOGE("8-bit data in shared memory is not supported"); 300 return BAD_VALUE; 301 } 302 303 // force direct flag if format is not linear PCM 304 // or offload was requested 305 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 306 || !audio_is_linear_pcm(format)) { 307 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 308 ? "Offload request, forcing to Direct Output" 309 : "Not linear PCM, forcing to Direct Output"); 310 flags = (audio_output_flags_t) 311 // FIXME why can't we allow direct AND fast? 312 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 313 } 314 // only allow deep buffering for music stream type 315 if (streamType != AUDIO_STREAM_MUSIC) { 316 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 317 } 318 319 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 320 if (audio_is_linear_pcm(format)) { 321 mFrameSize = channelCount * audio_bytes_per_sample(format); 322 } else { 323 mFrameSize = sizeof(uint8_t); 324 } 325 mFrameSizeAF = mFrameSize; 326 } else { 327 ALOG_ASSERT(audio_is_linear_pcm(format)); 328 mFrameSize = channelCount * audio_bytes_per_sample(format); 329 mFrameSizeAF = channelCount * audio_bytes_per_sample( 330 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 331 // createTrack will return an error if PCM format is not supported by server, 332 // so no need to check for specific PCM formats here 333 } 334 335 // Make copy of input parameter offloadInfo so that in the future: 336 // (a) createTrack_l doesn't need it as an input parameter 337 // (b) we can support re-creation of offloaded tracks 338 if (offloadInfo != NULL) { 339 mOffloadInfoCopy = *offloadInfo; 340 mOffloadInfo = &mOffloadInfoCopy; 341 } else { 342 mOffloadInfo = NULL; 343 } 344 345 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 346 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 347 mSendLevel = 0.0f; 348 // mFrameCount is initialized in createTrack_l 349 mReqFrameCount = frameCount; 350 mNotificationFramesReq = notificationFrames; 351 mNotificationFramesAct = 0; 352 mSessionId = sessionId; 353 int callingpid = IPCThreadState::self()->getCallingPid(); 354 int mypid = getpid(); 355 if (uid == -1 || (callingpid != mypid)) { 356 mClientUid = IPCThreadState::self()->getCallingUid(); 357 } else { 358 mClientUid = uid; 359 } 360 if (pid == -1 || (callingpid != mypid)) { 361 mClientPid = callingpid; 362 } else { 363 mClientPid = pid; 364 } 365 mAuxEffectId = 0; 366 mFlags = flags; 367 mCbf = cbf; 368 369 if (cbf != NULL) { 370 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 371 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 372 } 373 374 // create the IAudioTrack 375 status = createTrack_l(0 /*epoch*/); 376 377 if (status != NO_ERROR) { 378 if (mAudioTrackThread != 0) { 379 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 380 mAudioTrackThread->requestExitAndWait(); 381 mAudioTrackThread.clear(); 382 } 383 return status; 384 } 385 386 mStatus = NO_ERROR; 387 mState = STATE_STOPPED; 388 mUserData = user; 389 mLoopPeriod = 0; 390 mMarkerPosition = 0; 391 mMarkerReached = false; 392 mNewPosition = 0; 393 mUpdatePeriod = 0; 394 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 395 mSequence = 1; 396 mObservedSequence = mSequence; 397 mInUnderrun = false; 398 399 return NO_ERROR; 400} 401 402// ------------------------------------------------------------------------- 403 404status_t AudioTrack::start() 405{ 406 AutoMutex lock(mLock); 407 408 if (mState == STATE_ACTIVE) { 409 return INVALID_OPERATION; 410 } 411 412 mInUnderrun = true; 413 414 State previousState = mState; 415 if (previousState == STATE_PAUSED_STOPPING) { 416 mState = STATE_STOPPING; 417 } else { 418 mState = STATE_ACTIVE; 419 } 420 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 421 // reset current position as seen by client to 0 422 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 423 // force refresh of remaining frames by processAudioBuffer() as last 424 // write before stop could be partial. 425 mRefreshRemaining = true; 426 } 427 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 428 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 429 430 sp<AudioTrackThread> t = mAudioTrackThread; 431 if (t != 0) { 432 if (previousState == STATE_STOPPING) { 433 mProxy->interrupt(); 434 } else { 435 t->resume(); 436 } 437 } else { 438 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 439 get_sched_policy(0, &mPreviousSchedulingGroup); 440 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 441 } 442 443 status_t status = NO_ERROR; 444 if (!(flags & CBLK_INVALID)) { 445 status = mAudioTrack->start(); 446 if (status == DEAD_OBJECT) { 447 flags |= CBLK_INVALID; 448 } 449 } 450 if (flags & CBLK_INVALID) { 451 status = restoreTrack_l("start"); 452 } 453 454 if (status != NO_ERROR) { 455 ALOGE("start() status %d", status); 456 mState = previousState; 457 if (t != 0) { 458 if (previousState != STATE_STOPPING) { 459 t->pause(); 460 } 461 } else { 462 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 463 set_sched_policy(0, mPreviousSchedulingGroup); 464 } 465 } 466 467 return status; 468} 469 470void AudioTrack::stop() 471{ 472 AutoMutex lock(mLock); 473 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 474 return; 475 } 476 477 if (isOffloaded_l()) { 478 mState = STATE_STOPPING; 479 } else { 480 mState = STATE_STOPPED; 481 } 482 483 mProxy->interrupt(); 484 mAudioTrack->stop(); 485 // the playback head position will reset to 0, so if a marker is set, we need 486 // to activate it again 487 mMarkerReached = false; 488#if 0 489 // Force flush if a shared buffer is used otherwise audioflinger 490 // will not stop before end of buffer is reached. 491 // It may be needed to make sure that we stop playback, likely in case looping is on. 492 if (mSharedBuffer != 0) { 493 flush_l(); 494 } 495#endif 496 497 sp<AudioTrackThread> t = mAudioTrackThread; 498 if (t != 0) { 499 if (!isOffloaded_l()) { 500 t->pause(); 501 } 502 } else { 503 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 504 set_sched_policy(0, mPreviousSchedulingGroup); 505 } 506} 507 508bool AudioTrack::stopped() const 509{ 510 AutoMutex lock(mLock); 511 return mState != STATE_ACTIVE; 512} 513 514void AudioTrack::flush() 515{ 516 if (mSharedBuffer != 0) { 517 return; 518 } 519 AutoMutex lock(mLock); 520 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 521 return; 522 } 523 flush_l(); 524} 525 526void AudioTrack::flush_l() 527{ 528 ALOG_ASSERT(mState != STATE_ACTIVE); 529 530 // clear playback marker and periodic update counter 531 mMarkerPosition = 0; 532 mMarkerReached = false; 533 mUpdatePeriod = 0; 534 mRefreshRemaining = true; 535 536 mState = STATE_FLUSHED; 537 if (isOffloaded_l()) { 538 mProxy->interrupt(); 539 } 540 mProxy->flush(); 541 mAudioTrack->flush(); 542} 543 544void AudioTrack::pause() 545{ 546 AutoMutex lock(mLock); 547 if (mState == STATE_ACTIVE) { 548 mState = STATE_PAUSED; 549 } else if (mState == STATE_STOPPING) { 550 mState = STATE_PAUSED_STOPPING; 551 } else { 552 return; 553 } 554 mProxy->interrupt(); 555 mAudioTrack->pause(); 556 557 if (isOffloaded_l()) { 558 if (mOutput != AUDIO_IO_HANDLE_NONE) { 559 uint32_t halFrames; 560 // OffloadThread sends HAL pause in its threadLoop.. time saved 561 // here can be slightly off 562 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 563 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 564 } 565 } 566} 567 568status_t AudioTrack::setVolume(float left, float right) 569{ 570 // This duplicates a test by AudioTrack JNI, but that is not the only caller 571 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 572 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 573 return BAD_VALUE; 574 } 575 576 AutoMutex lock(mLock); 577 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 578 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 579 580 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 581 582 if (isOffloaded_l()) { 583 mAudioTrack->signal(); 584 } 585 return NO_ERROR; 586} 587 588status_t AudioTrack::setVolume(float volume) 589{ 590 return setVolume(volume, volume); 591} 592 593status_t AudioTrack::setAuxEffectSendLevel(float level) 594{ 595 // This duplicates a test by AudioTrack JNI, but that is not the only caller 596 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 597 return BAD_VALUE; 598 } 599 600 AutoMutex lock(mLock); 601 mSendLevel = level; 602 mProxy->setSendLevel(level); 603 604 return NO_ERROR; 605} 606 607void AudioTrack::getAuxEffectSendLevel(float* level) const 608{ 609 if (level != NULL) { 610 *level = mSendLevel; 611 } 612} 613 614status_t AudioTrack::setSampleRate(uint32_t rate) 615{ 616 if (mIsTimed || isOffloaded()) { 617 return INVALID_OPERATION; 618 } 619 620 uint32_t afSamplingRate; 621 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 622 return NO_INIT; 623 } 624 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 625 if (rate == 0 || rate > afSamplingRate*2 ) { 626 return BAD_VALUE; 627 } 628 629 AutoMutex lock(mLock); 630 mSampleRate = rate; 631 mProxy->setSampleRate(rate); 632 633 return NO_ERROR; 634} 635 636uint32_t AudioTrack::getSampleRate() const 637{ 638 if (mIsTimed) { 639 return 0; 640 } 641 642 AutoMutex lock(mLock); 643 644 // sample rate can be updated during playback by the offloaded decoder so we need to 645 // query the HAL and update if needed. 646// FIXME use Proxy return channel to update the rate from server and avoid polling here 647 if (isOffloaded_l()) { 648 if (mOutput != AUDIO_IO_HANDLE_NONE) { 649 uint32_t sampleRate = 0; 650 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 651 if (status == NO_ERROR) { 652 mSampleRate = sampleRate; 653 } 654 } 655 } 656 return mSampleRate; 657} 658 659status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 660{ 661 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 662 return INVALID_OPERATION; 663 } 664 665 if (loopCount == 0) { 666 ; 667 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 668 loopEnd - loopStart >= MIN_LOOP) { 669 ; 670 } else { 671 return BAD_VALUE; 672 } 673 674 AutoMutex lock(mLock); 675 // See setPosition() regarding setting parameters such as loop points or position while active 676 if (mState == STATE_ACTIVE) { 677 return INVALID_OPERATION; 678 } 679 setLoop_l(loopStart, loopEnd, loopCount); 680 return NO_ERROR; 681} 682 683void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 684{ 685 // FIXME If setting a loop also sets position to start of loop, then 686 // this is correct. Otherwise it should be removed. 687 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 688 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 689 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 690} 691 692status_t AudioTrack::setMarkerPosition(uint32_t marker) 693{ 694 // The only purpose of setting marker position is to get a callback 695 if (mCbf == NULL || isOffloaded()) { 696 return INVALID_OPERATION; 697 } 698 699 AutoMutex lock(mLock); 700 mMarkerPosition = marker; 701 mMarkerReached = false; 702 703 return NO_ERROR; 704} 705 706status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 707{ 708 if (isOffloaded()) { 709 return INVALID_OPERATION; 710 } 711 if (marker == NULL) { 712 return BAD_VALUE; 713 } 714 715 AutoMutex lock(mLock); 716 *marker = mMarkerPosition; 717 718 return NO_ERROR; 719} 720 721status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 722{ 723 // The only purpose of setting position update period is to get a callback 724 if (mCbf == NULL || isOffloaded()) { 725 return INVALID_OPERATION; 726 } 727 728 AutoMutex lock(mLock); 729 mNewPosition = mProxy->getPosition() + updatePeriod; 730 mUpdatePeriod = updatePeriod; 731 732 return NO_ERROR; 733} 734 735status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 736{ 737 if (isOffloaded()) { 738 return INVALID_OPERATION; 739 } 740 if (updatePeriod == NULL) { 741 return BAD_VALUE; 742 } 743 744 AutoMutex lock(mLock); 745 *updatePeriod = mUpdatePeriod; 746 747 return NO_ERROR; 748} 749 750status_t AudioTrack::setPosition(uint32_t position) 751{ 752 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 753 return INVALID_OPERATION; 754 } 755 if (position > mFrameCount) { 756 return BAD_VALUE; 757 } 758 759 AutoMutex lock(mLock); 760 // Currently we require that the player is inactive before setting parameters such as position 761 // or loop points. Otherwise, there could be a race condition: the application could read the 762 // current position, compute a new position or loop parameters, and then set that position or 763 // loop parameters but it would do the "wrong" thing since the position has continued to advance 764 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 765 // to specify how it wants to handle such scenarios. 766 if (mState == STATE_ACTIVE) { 767 return INVALID_OPERATION; 768 } 769 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 770 mLoopPeriod = 0; 771 // FIXME Check whether loops and setting position are incompatible in old code. 772 // If we use setLoop for both purposes we lose the capability to set the position while looping. 773 mStaticProxy->setLoop(position, mFrameCount, 0); 774 775 return NO_ERROR; 776} 777 778status_t AudioTrack::getPosition(uint32_t *position) const 779{ 780 if (position == NULL) { 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 if (isOffloaded_l()) { 786 uint32_t dspFrames = 0; 787 788 if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) { 789 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 790 *position = mPausedPosition; 791 return NO_ERROR; 792 } 793 794 if (mOutput != AUDIO_IO_HANDLE_NONE) { 795 uint32_t halFrames; 796 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 797 } 798 *position = dspFrames; 799 } else { 800 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 801 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 802 mProxy->getPosition(); 803 } 804 return NO_ERROR; 805} 806 807status_t AudioTrack::getBufferPosition(uint32_t *position) 808{ 809 if (mSharedBuffer == 0 || mIsTimed) { 810 return INVALID_OPERATION; 811 } 812 if (position == NULL) { 813 return BAD_VALUE; 814 } 815 816 AutoMutex lock(mLock); 817 *position = mStaticProxy->getBufferPosition(); 818 return NO_ERROR; 819} 820 821status_t AudioTrack::reload() 822{ 823 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 824 return INVALID_OPERATION; 825 } 826 827 AutoMutex lock(mLock); 828 // See setPosition() regarding setting parameters such as loop points or position while active 829 if (mState == STATE_ACTIVE) { 830 return INVALID_OPERATION; 831 } 832 mNewPosition = mUpdatePeriod; 833 mLoopPeriod = 0; 834 // FIXME The new code cannot reload while keeping a loop specified. 835 // Need to check how the old code handled this, and whether it's a significant change. 836 mStaticProxy->setLoop(0, mFrameCount, 0); 837 return NO_ERROR; 838} 839 840audio_io_handle_t AudioTrack::getOutput() const 841{ 842 AutoMutex lock(mLock); 843 return mOutput; 844} 845 846status_t AudioTrack::attachAuxEffect(int effectId) 847{ 848 AutoMutex lock(mLock); 849 status_t status = mAudioTrack->attachAuxEffect(effectId); 850 if (status == NO_ERROR) { 851 mAuxEffectId = effectId; 852 } 853 return status; 854} 855 856// ------------------------------------------------------------------------- 857 858// must be called with mLock held 859status_t AudioTrack::createTrack_l(size_t epoch) 860{ 861 status_t status; 862 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 863 if (audioFlinger == 0) { 864 ALOGE("Could not get audioflinger"); 865 return NO_INIT; 866 } 867 868 audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, 869 mChannelMask, mFlags, mOffloadInfo); 870 if (output == AUDIO_IO_HANDLE_NONE) { 871 ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " 872 "channel mask %#x, flags %#x", 873 mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); 874 return BAD_VALUE; 875 } 876 { 877 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 878 // we must release it ourselves if anything goes wrong. 879 880 // Not all of these values are needed under all conditions, but it is easier to get them all 881 882 uint32_t afLatency; 883 status = AudioSystem::getLatency(output, &afLatency); 884 if (status != NO_ERROR) { 885 ALOGE("getLatency(%d) failed status %d", output, status); 886 goto release; 887 } 888 889 size_t afFrameCount; 890 status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); 891 if (status != NO_ERROR) { 892 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); 893 goto release; 894 } 895 896 uint32_t afSampleRate; 897 status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); 898 if (status != NO_ERROR) { 899 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); 900 goto release; 901 } 902 903 // Client decides whether the track is TIMED (see below), but can only express a preference 904 // for FAST. Server will perform additional tests. 905 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 906 // either of these use cases: 907 // use case 1: shared buffer 908 (mSharedBuffer != 0) || 909 // use case 2: callback transfer mode 910 (mTransfer == TRANSFER_CALLBACK)) && 911 // matching sample rate 912 (mSampleRate == afSampleRate))) { 913 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 914 // once denied, do not request again if IAudioTrack is re-created 915 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 916 } 917 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 918 919 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 920 // n = 1 fast track with single buffering; nBuffering is ignored 921 // n = 2 fast track with double buffering 922 // n = 2 normal track, no sample rate conversion 923 // n = 3 normal track, with sample rate conversion 924 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 925 // n > 3 very high latency or very small notification interval; nBuffering is ignored 926 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 927 928 mNotificationFramesAct = mNotificationFramesReq; 929 930 size_t frameCount = mReqFrameCount; 931 if (!audio_is_linear_pcm(mFormat)) { 932 933 if (mSharedBuffer != 0) { 934 // Same comment as below about ignoring frameCount parameter for set() 935 frameCount = mSharedBuffer->size(); 936 } else if (frameCount == 0) { 937 frameCount = afFrameCount; 938 } 939 if (mNotificationFramesAct != frameCount) { 940 mNotificationFramesAct = frameCount; 941 } 942 } else if (mSharedBuffer != 0) { 943 944 // Ensure that buffer alignment matches channel count 945 // 8-bit data in shared memory is not currently supported by AudioFlinger 946 size_t alignment = audio_bytes_per_sample( 947 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 948 if (alignment & 1) { 949 alignment = 1; 950 } 951 if (mChannelCount > 1) { 952 // More than 2 channels does not require stronger alignment than stereo 953 alignment <<= 1; 954 } 955 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 956 ALOGE("Invalid buffer alignment: address %p, channel count %u", 957 mSharedBuffer->pointer(), mChannelCount); 958 status = BAD_VALUE; 959 goto release; 960 } 961 962 // When initializing a shared buffer AudioTrack via constructors, 963 // there's no frameCount parameter. 964 // But when initializing a shared buffer AudioTrack via set(), 965 // there _is_ a frameCount parameter. We silently ignore it. 966 frameCount = mSharedBuffer->size() / mFrameSizeAF; 967 968 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 969 970 // FIXME move these calculations and associated checks to server 971 972 // Ensure that buffer depth covers at least audio hardware latency 973 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 974 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 975 afFrameCount, minBufCount, afSampleRate, afLatency); 976 if (minBufCount <= nBuffering) { 977 minBufCount = nBuffering; 978 } 979 980 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 981 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 982 ", afLatency=%d", 983 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 984 985 if (frameCount == 0) { 986 frameCount = minFrameCount; 987 } else if (frameCount < minFrameCount) { 988 // not ALOGW because it happens all the time when playing key clicks over A2DP 989 ALOGV("Minimum buffer size corrected from %d to %d", 990 frameCount, minFrameCount); 991 frameCount = minFrameCount; 992 } 993 // Make sure that application is notified with sufficient margin before underrun 994 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 995 mNotificationFramesAct = frameCount/nBuffering; 996 } 997 998 } else { 999 // For fast tracks, the frame count calculations and checks are done by server 1000 } 1001 1002 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1003 if (mIsTimed) { 1004 trackFlags |= IAudioFlinger::TRACK_TIMED; 1005 } 1006 1007 pid_t tid = -1; 1008 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1009 trackFlags |= IAudioFlinger::TRACK_FAST; 1010 if (mAudioTrackThread != 0) { 1011 tid = mAudioTrackThread->getTid(); 1012 } 1013 } 1014 1015 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1016 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1017 } 1018 1019 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1020 // but we will still need the original value also 1021 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1022 mSampleRate, 1023 // AudioFlinger only sees 16-bit PCM 1024 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1025 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1026 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1027 mChannelMask, 1028 &temp, 1029 &trackFlags, 1030 mSharedBuffer, 1031 output, 1032 tid, 1033 &mSessionId, 1034 mClientUid, 1035 &status); 1036 1037 if (status != NO_ERROR) { 1038 ALOGE("AudioFlinger could not create track, status: %d", status); 1039 goto release; 1040 } 1041 ALOG_ASSERT(track != 0); 1042 1043 // AudioFlinger now owns the reference to the I/O handle, 1044 // so we are no longer responsible for releasing it. 1045 1046 sp<IMemory> iMem = track->getCblk(); 1047 if (iMem == 0) { 1048 ALOGE("Could not get control block"); 1049 return NO_INIT; 1050 } 1051 void *iMemPointer = iMem->pointer(); 1052 if (iMemPointer == NULL) { 1053 ALOGE("Could not get control block pointer"); 1054 return NO_INIT; 1055 } 1056 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1057 if (mAudioTrack != 0) { 1058 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1059 mDeathNotifier.clear(); 1060 } 1061 mAudioTrack = track; 1062 1063 mCblkMemory = iMem; 1064 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1065 mCblk = cblk; 1066 // note that temp is the (possibly revised) value of frameCount 1067 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1068 // In current design, AudioTrack client checks and ensures frame count validity before 1069 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1070 // for fast track as it uses a special method of assigning frame count. 1071 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1072 } 1073 frameCount = temp; 1074 1075 mAwaitBoost = false; 1076 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1077 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1078 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1079 mAwaitBoost = true; 1080 if (mSharedBuffer == 0) { 1081 // Theoretically double-buffering is not required for fast tracks, 1082 // due to tighter scheduling. But in practice, to accommodate kernels with 1083 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1084 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1085 mNotificationFramesAct = frameCount/nBuffering; 1086 } 1087 } 1088 } else { 1089 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1090 // once denied, do not request again if IAudioTrack is re-created 1091 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1092 if (mSharedBuffer == 0) { 1093 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1094 mNotificationFramesAct = frameCount/nBuffering; 1095 } 1096 } 1097 } 1098 } 1099 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1100 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1101 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1102 } else { 1103 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1104 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1105 // FIXME This is a warning, not an error, so don't return error status 1106 //return NO_INIT; 1107 } 1108 } 1109 1110 // We retain a copy of the I/O handle, but don't own the reference 1111 mOutput = output; 1112 mRefreshRemaining = true; 1113 1114 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1115 // is the value of pointer() for the shared buffer, otherwise buffers points 1116 // immediately after the control block. This address is for the mapping within client 1117 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1118 void* buffers; 1119 if (mSharedBuffer == 0) { 1120 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1121 } else { 1122 buffers = mSharedBuffer->pointer(); 1123 } 1124 1125 mAudioTrack->attachAuxEffect(mAuxEffectId); 1126 // FIXME don't believe this lie 1127 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1128 1129 mFrameCount = frameCount; 1130 // If IAudioTrack is re-created, don't let the requested frameCount 1131 // decrease. This can confuse clients that cache frameCount(). 1132 if (frameCount > mReqFrameCount) { 1133 mReqFrameCount = frameCount; 1134 } 1135 1136 // update proxy 1137 if (mSharedBuffer == 0) { 1138 mStaticProxy.clear(); 1139 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1140 } else { 1141 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1142 mProxy = mStaticProxy; 1143 } 1144 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1145 mProxy->setSendLevel(mSendLevel); 1146 mProxy->setSampleRate(mSampleRate); 1147 mProxy->setEpoch(epoch); 1148 mProxy->setMinimum(mNotificationFramesAct); 1149 1150 mDeathNotifier = new DeathNotifier(this); 1151 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1152 1153 return NO_ERROR; 1154 } 1155 1156release: 1157 AudioSystem::releaseOutput(output); 1158 if (status == NO_ERROR) { 1159 status = NO_INIT; 1160 } 1161 return status; 1162} 1163 1164status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1165{ 1166 if (audioBuffer == NULL) { 1167 return BAD_VALUE; 1168 } 1169 if (mTransfer != TRANSFER_OBTAIN) { 1170 audioBuffer->frameCount = 0; 1171 audioBuffer->size = 0; 1172 audioBuffer->raw = NULL; 1173 return INVALID_OPERATION; 1174 } 1175 1176 const struct timespec *requested; 1177 struct timespec timeout; 1178 if (waitCount == -1) { 1179 requested = &ClientProxy::kForever; 1180 } else if (waitCount == 0) { 1181 requested = &ClientProxy::kNonBlocking; 1182 } else if (waitCount > 0) { 1183 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1184 timeout.tv_sec = ms / 1000; 1185 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1186 requested = &timeout; 1187 } else { 1188 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1189 requested = NULL; 1190 } 1191 return obtainBuffer(audioBuffer, requested); 1192} 1193 1194status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1195 struct timespec *elapsed, size_t *nonContig) 1196{ 1197 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1198 uint32_t oldSequence = 0; 1199 uint32_t newSequence; 1200 1201 Proxy::Buffer buffer; 1202 status_t status = NO_ERROR; 1203 1204 static const int32_t kMaxTries = 5; 1205 int32_t tryCounter = kMaxTries; 1206 1207 do { 1208 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1209 // keep them from going away if another thread re-creates the track during obtainBuffer() 1210 sp<AudioTrackClientProxy> proxy; 1211 sp<IMemory> iMem; 1212 1213 { // start of lock scope 1214 AutoMutex lock(mLock); 1215 1216 newSequence = mSequence; 1217 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1218 if (status == DEAD_OBJECT) { 1219 // re-create track, unless someone else has already done so 1220 if (newSequence == oldSequence) { 1221 status = restoreTrack_l("obtainBuffer"); 1222 if (status != NO_ERROR) { 1223 buffer.mFrameCount = 0; 1224 buffer.mRaw = NULL; 1225 buffer.mNonContig = 0; 1226 break; 1227 } 1228 } 1229 } 1230 oldSequence = newSequence; 1231 1232 // Keep the extra references 1233 proxy = mProxy; 1234 iMem = mCblkMemory; 1235 1236 if (mState == STATE_STOPPING) { 1237 status = -EINTR; 1238 buffer.mFrameCount = 0; 1239 buffer.mRaw = NULL; 1240 buffer.mNonContig = 0; 1241 break; 1242 } 1243 1244 // Non-blocking if track is stopped or paused 1245 if (mState != STATE_ACTIVE) { 1246 requested = &ClientProxy::kNonBlocking; 1247 } 1248 1249 } // end of lock scope 1250 1251 buffer.mFrameCount = audioBuffer->frameCount; 1252 // FIXME starts the requested timeout and elapsed over from scratch 1253 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1254 1255 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1256 1257 audioBuffer->frameCount = buffer.mFrameCount; 1258 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1259 audioBuffer->raw = buffer.mRaw; 1260 if (nonContig != NULL) { 1261 *nonContig = buffer.mNonContig; 1262 } 1263 return status; 1264} 1265 1266void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1267{ 1268 if (mTransfer == TRANSFER_SHARED) { 1269 return; 1270 } 1271 1272 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1273 if (stepCount == 0) { 1274 return; 1275 } 1276 1277 Proxy::Buffer buffer; 1278 buffer.mFrameCount = stepCount; 1279 buffer.mRaw = audioBuffer->raw; 1280 1281 AutoMutex lock(mLock); 1282 mInUnderrun = false; 1283 mProxy->releaseBuffer(&buffer); 1284 1285 // restart track if it was disabled by audioflinger due to previous underrun 1286 if (mState == STATE_ACTIVE) { 1287 audio_track_cblk_t* cblk = mCblk; 1288 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1289 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1290 // FIXME ignoring status 1291 mAudioTrack->start(); 1292 } 1293 } 1294} 1295 1296// ------------------------------------------------------------------------- 1297 1298ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1299{ 1300 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1301 return INVALID_OPERATION; 1302 } 1303 1304 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1305 // Sanity-check: user is most-likely passing an error code, and it would 1306 // make the return value ambiguous (actualSize vs error). 1307 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1308 return BAD_VALUE; 1309 } 1310 1311 size_t written = 0; 1312 Buffer audioBuffer; 1313 1314 while (userSize >= mFrameSize) { 1315 audioBuffer.frameCount = userSize / mFrameSize; 1316 1317 status_t err = obtainBuffer(&audioBuffer, 1318 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1319 if (err < 0) { 1320 if (written > 0) { 1321 break; 1322 } 1323 return ssize_t(err); 1324 } 1325 1326 size_t toWrite; 1327 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1328 // Divide capacity by 2 to take expansion into account 1329 toWrite = audioBuffer.size >> 1; 1330 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1331 } else { 1332 toWrite = audioBuffer.size; 1333 memcpy(audioBuffer.i8, buffer, toWrite); 1334 } 1335 buffer = ((const char *) buffer) + toWrite; 1336 userSize -= toWrite; 1337 written += toWrite; 1338 1339 releaseBuffer(&audioBuffer); 1340 } 1341 1342 return written; 1343} 1344 1345// ------------------------------------------------------------------------- 1346 1347TimedAudioTrack::TimedAudioTrack() { 1348 mIsTimed = true; 1349} 1350 1351status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1352{ 1353 AutoMutex lock(mLock); 1354 status_t result = UNKNOWN_ERROR; 1355 1356#if 1 1357 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1358 // while we are accessing the cblk 1359 sp<IAudioTrack> audioTrack = mAudioTrack; 1360 sp<IMemory> iMem = mCblkMemory; 1361#endif 1362 1363 // If the track is not invalid already, try to allocate a buffer. alloc 1364 // fails indicating that the server is dead, flag the track as invalid so 1365 // we can attempt to restore in just a bit. 1366 audio_track_cblk_t* cblk = mCblk; 1367 if (!(cblk->mFlags & CBLK_INVALID)) { 1368 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1369 if (result == DEAD_OBJECT) { 1370 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1371 } 1372 } 1373 1374 // If the track is invalid at this point, attempt to restore it. and try the 1375 // allocation one more time. 1376 if (cblk->mFlags & CBLK_INVALID) { 1377 result = restoreTrack_l("allocateTimedBuffer"); 1378 1379 if (result == NO_ERROR) { 1380 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1381 } 1382 } 1383 1384 return result; 1385} 1386 1387status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1388 int64_t pts) 1389{ 1390 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1391 { 1392 AutoMutex lock(mLock); 1393 audio_track_cblk_t* cblk = mCblk; 1394 // restart track if it was disabled by audioflinger due to previous underrun 1395 if (buffer->size() != 0 && status == NO_ERROR && 1396 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1397 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1398 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1399 // FIXME ignoring status 1400 mAudioTrack->start(); 1401 } 1402 } 1403 return status; 1404} 1405 1406status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1407 TargetTimeline target) 1408{ 1409 return mAudioTrack->setMediaTimeTransform(xform, target); 1410} 1411 1412// ------------------------------------------------------------------------- 1413 1414nsecs_t AudioTrack::processAudioBuffer() 1415{ 1416 // Currently the AudioTrack thread is not created if there are no callbacks. 1417 // Would it ever make sense to run the thread, even without callbacks? 1418 // If so, then replace this by checks at each use for mCbf != NULL. 1419 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1420 1421 mLock.lock(); 1422 if (mAwaitBoost) { 1423 mAwaitBoost = false; 1424 mLock.unlock(); 1425 static const int32_t kMaxTries = 5; 1426 int32_t tryCounter = kMaxTries; 1427 uint32_t pollUs = 10000; 1428 do { 1429 int policy = sched_getscheduler(0); 1430 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1431 break; 1432 } 1433 usleep(pollUs); 1434 pollUs <<= 1; 1435 } while (tryCounter-- > 0); 1436 if (tryCounter < 0) { 1437 ALOGE("did not receive expected priority boost on time"); 1438 } 1439 // Run again immediately 1440 return 0; 1441 } 1442 1443 // Can only reference mCblk while locked 1444 int32_t flags = android_atomic_and( 1445 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1446 1447 // Check for track invalidation 1448 if (flags & CBLK_INVALID) { 1449 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1450 // AudioSystem cache. We should not exit here but after calling the callback so 1451 // that the upper layers can recreate the track 1452 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1453 status_t status = restoreTrack_l("processAudioBuffer"); 1454 mLock.unlock(); 1455 // Run again immediately, but with a new IAudioTrack 1456 return 0; 1457 } 1458 } 1459 1460 bool waitStreamEnd = mState == STATE_STOPPING; 1461 bool active = mState == STATE_ACTIVE; 1462 1463 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1464 bool newUnderrun = false; 1465 if (flags & CBLK_UNDERRUN) { 1466#if 0 1467 // Currently in shared buffer mode, when the server reaches the end of buffer, 1468 // the track stays active in continuous underrun state. It's up to the application 1469 // to pause or stop the track, or set the position to a new offset within buffer. 1470 // This was some experimental code to auto-pause on underrun. Keeping it here 1471 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1472 if (mTransfer == TRANSFER_SHARED) { 1473 mState = STATE_PAUSED; 1474 active = false; 1475 } 1476#endif 1477 if (!mInUnderrun) { 1478 mInUnderrun = true; 1479 newUnderrun = true; 1480 } 1481 } 1482 1483 // Get current position of server 1484 size_t position = mProxy->getPosition(); 1485 1486 // Manage marker callback 1487 bool markerReached = false; 1488 size_t markerPosition = mMarkerPosition; 1489 // FIXME fails for wraparound, need 64 bits 1490 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1491 mMarkerReached = markerReached = true; 1492 } 1493 1494 // Determine number of new position callback(s) that will be needed, while locked 1495 size_t newPosCount = 0; 1496 size_t newPosition = mNewPosition; 1497 size_t updatePeriod = mUpdatePeriod; 1498 // FIXME fails for wraparound, need 64 bits 1499 if (updatePeriod > 0 && position >= newPosition) { 1500 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1501 mNewPosition += updatePeriod * newPosCount; 1502 } 1503 1504 // Cache other fields that will be needed soon 1505 uint32_t loopPeriod = mLoopPeriod; 1506 uint32_t sampleRate = mSampleRate; 1507 uint32_t notificationFrames = mNotificationFramesAct; 1508 if (mRefreshRemaining) { 1509 mRefreshRemaining = false; 1510 mRemainingFrames = notificationFrames; 1511 mRetryOnPartialBuffer = false; 1512 } 1513 size_t misalignment = mProxy->getMisalignment(); 1514 uint32_t sequence = mSequence; 1515 sp<AudioTrackClientProxy> proxy = mProxy; 1516 1517 // These fields don't need to be cached, because they are assigned only by set(): 1518 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1519 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1520 1521 mLock.unlock(); 1522 1523 if (waitStreamEnd) { 1524 struct timespec timeout; 1525 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1526 timeout.tv_nsec = 0; 1527 1528 status_t status = proxy->waitStreamEndDone(&timeout); 1529 switch (status) { 1530 case NO_ERROR: 1531 case DEAD_OBJECT: 1532 case TIMED_OUT: 1533 mCbf(EVENT_STREAM_END, mUserData, NULL); 1534 { 1535 AutoMutex lock(mLock); 1536 // The previously assigned value of waitStreamEnd is no longer valid, 1537 // since the mutex has been unlocked and either the callback handler 1538 // or another thread could have re-started the AudioTrack during that time. 1539 waitStreamEnd = mState == STATE_STOPPING; 1540 if (waitStreamEnd) { 1541 mState = STATE_STOPPED; 1542 } 1543 } 1544 if (waitStreamEnd && status != DEAD_OBJECT) { 1545 return NS_INACTIVE; 1546 } 1547 break; 1548 } 1549 return 0; 1550 } 1551 1552 // perform callbacks while unlocked 1553 if (newUnderrun) { 1554 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1555 } 1556 // FIXME we will miss loops if loop cycle was signaled several times since last call 1557 // to processAudioBuffer() 1558 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1559 mCbf(EVENT_LOOP_END, mUserData, NULL); 1560 } 1561 if (flags & CBLK_BUFFER_END) { 1562 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1563 } 1564 if (markerReached) { 1565 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1566 } 1567 while (newPosCount > 0) { 1568 size_t temp = newPosition; 1569 mCbf(EVENT_NEW_POS, mUserData, &temp); 1570 newPosition += updatePeriod; 1571 newPosCount--; 1572 } 1573 1574 if (mObservedSequence != sequence) { 1575 mObservedSequence = sequence; 1576 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1577 // for offloaded tracks, just wait for the upper layers to recreate the track 1578 if (isOffloaded()) { 1579 return NS_INACTIVE; 1580 } 1581 } 1582 1583 // if inactive, then don't run me again until re-started 1584 if (!active) { 1585 return NS_INACTIVE; 1586 } 1587 1588 // Compute the estimated time until the next timed event (position, markers, loops) 1589 // FIXME only for non-compressed audio 1590 uint32_t minFrames = ~0; 1591 if (!markerReached && position < markerPosition) { 1592 minFrames = markerPosition - position; 1593 } 1594 if (loopPeriod > 0 && loopPeriod < minFrames) { 1595 minFrames = loopPeriod; 1596 } 1597 if (updatePeriod > 0 && updatePeriod < minFrames) { 1598 minFrames = updatePeriod; 1599 } 1600 1601 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1602 static const uint32_t kPoll = 0; 1603 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1604 minFrames = kPoll * notificationFrames; 1605 } 1606 1607 // Convert frame units to time units 1608 nsecs_t ns = NS_WHENEVER; 1609 if (minFrames != (uint32_t) ~0) { 1610 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1611 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1612 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1613 } 1614 1615 // If not supplying data by EVENT_MORE_DATA, then we're done 1616 if (mTransfer != TRANSFER_CALLBACK) { 1617 return ns; 1618 } 1619 1620 struct timespec timeout; 1621 const struct timespec *requested = &ClientProxy::kForever; 1622 if (ns != NS_WHENEVER) { 1623 timeout.tv_sec = ns / 1000000000LL; 1624 timeout.tv_nsec = ns % 1000000000LL; 1625 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1626 requested = &timeout; 1627 } 1628 1629 while (mRemainingFrames > 0) { 1630 1631 Buffer audioBuffer; 1632 audioBuffer.frameCount = mRemainingFrames; 1633 size_t nonContig; 1634 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1635 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1636 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1637 requested = &ClientProxy::kNonBlocking; 1638 size_t avail = audioBuffer.frameCount + nonContig; 1639 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1640 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1641 if (err != NO_ERROR) { 1642 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1643 (isOffloaded() && (err == DEAD_OBJECT))) { 1644 return 0; 1645 } 1646 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1647 return NS_NEVER; 1648 } 1649 1650 if (mRetryOnPartialBuffer && !isOffloaded()) { 1651 mRetryOnPartialBuffer = false; 1652 if (avail < mRemainingFrames) { 1653 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1654 if (ns < 0 || myns < ns) { 1655 ns = myns; 1656 } 1657 return ns; 1658 } 1659 } 1660 1661 // Divide buffer size by 2 to take into account the expansion 1662 // due to 8 to 16 bit conversion: the callback must fill only half 1663 // of the destination buffer 1664 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1665 audioBuffer.size >>= 1; 1666 } 1667 1668 size_t reqSize = audioBuffer.size; 1669 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1670 size_t writtenSize = audioBuffer.size; 1671 1672 // Sanity check on returned size 1673 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1674 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1675 reqSize, (int) writtenSize); 1676 return NS_NEVER; 1677 } 1678 1679 if (writtenSize == 0) { 1680 // The callback is done filling buffers 1681 // Keep this thread going to handle timed events and 1682 // still try to get more data in intervals of WAIT_PERIOD_MS 1683 // but don't just loop and block the CPU, so wait 1684 return WAIT_PERIOD_MS * 1000000LL; 1685 } 1686 1687 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1688 // 8 to 16 bit conversion, note that source and destination are the same address 1689 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1690 audioBuffer.size <<= 1; 1691 } 1692 1693 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1694 audioBuffer.frameCount = releasedFrames; 1695 mRemainingFrames -= releasedFrames; 1696 if (misalignment >= releasedFrames) { 1697 misalignment -= releasedFrames; 1698 } else { 1699 misalignment = 0; 1700 } 1701 1702 releaseBuffer(&audioBuffer); 1703 1704 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1705 // if callback doesn't like to accept the full chunk 1706 if (writtenSize < reqSize) { 1707 continue; 1708 } 1709 1710 // There could be enough non-contiguous frames available to satisfy the remaining request 1711 if (mRemainingFrames <= nonContig) { 1712 continue; 1713 } 1714 1715#if 0 1716 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1717 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1718 // that total to a sum == notificationFrames. 1719 if (0 < misalignment && misalignment <= mRemainingFrames) { 1720 mRemainingFrames = misalignment; 1721 return (mRemainingFrames * 1100000000LL) / sampleRate; 1722 } 1723#endif 1724 1725 } 1726 mRemainingFrames = notificationFrames; 1727 mRetryOnPartialBuffer = true; 1728 1729 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1730 return 0; 1731} 1732 1733status_t AudioTrack::restoreTrack_l(const char *from) 1734{ 1735 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1736 isOffloaded_l() ? "Offloaded" : "PCM", from); 1737 ++mSequence; 1738 status_t result; 1739 1740 // refresh the audio configuration cache in this process to make sure we get new 1741 // output parameters in createTrack_l() 1742 AudioSystem::clearAudioConfigCache(); 1743 1744 if (isOffloaded_l()) { 1745 // FIXME re-creation of offloaded tracks is not yet implemented 1746 return DEAD_OBJECT; 1747 } 1748 1749 // if the new IAudioTrack is created, createTrack_l() will modify the 1750 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1751 // It will also delete the strong references on previous IAudioTrack and IMemory 1752 1753 // take the frames that will be lost by track recreation into account in saved position 1754 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1755 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1756 result = createTrack_l(position /*epoch*/); 1757 1758 if (result == NO_ERROR) { 1759 // continue playback from last known position, but 1760 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1761 if (mStaticProxy != NULL) { 1762 mLoopPeriod = 0; 1763 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1764 } 1765 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1766 // track destruction have been played? This is critical for SoundPool implementation 1767 // This must be broken, and needs to be tested/debugged. 1768#if 0 1769 // restore write index and set other indexes to reflect empty buffer status 1770 if (!strcmp(from, "start")) { 1771 // Make sure that a client relying on callback events indicating underrun or 1772 // the actual amount of audio frames played (e.g SoundPool) receives them. 1773 if (mSharedBuffer == 0) { 1774 // restart playback even if buffer is not completely filled. 1775 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1776 } 1777 } 1778#endif 1779 if (mState == STATE_ACTIVE) { 1780 result = mAudioTrack->start(); 1781 } 1782 } 1783 if (result != NO_ERROR) { 1784 ALOGW("restoreTrack_l() failed status %d", result); 1785 mState = STATE_STOPPED; 1786 } 1787 1788 return result; 1789} 1790 1791status_t AudioTrack::setParameters(const String8& keyValuePairs) 1792{ 1793 AutoMutex lock(mLock); 1794 return mAudioTrack->setParameters(keyValuePairs); 1795} 1796 1797status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1798{ 1799 AutoMutex lock(mLock); 1800 // FIXME not implemented for fast tracks; should use proxy and SSQ 1801 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1802 return INVALID_OPERATION; 1803 } 1804 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1805 return INVALID_OPERATION; 1806 } 1807 status_t status = mAudioTrack->getTimestamp(timestamp); 1808 if (status == NO_ERROR) { 1809 timestamp.mPosition += mProxy->getEpoch(); 1810 } 1811 return status; 1812} 1813 1814String8 AudioTrack::getParameters(const String8& keys) 1815{ 1816 audio_io_handle_t output = getOutput(); 1817 if (output != AUDIO_IO_HANDLE_NONE) { 1818 return AudioSystem::getParameters(output, keys); 1819 } else { 1820 return String8::empty(); 1821 } 1822} 1823 1824bool AudioTrack::isOffloaded() const 1825{ 1826 AutoMutex lock(mLock); 1827 return isOffloaded_l(); 1828} 1829 1830status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1831{ 1832 1833 const size_t SIZE = 256; 1834 char buffer[SIZE]; 1835 String8 result; 1836 1837 result.append(" AudioTrack::dump\n"); 1838 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1839 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 1840 result.append(buffer); 1841 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1842 mChannelCount, mFrameCount); 1843 result.append(buffer); 1844 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1845 result.append(buffer); 1846 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1847 result.append(buffer); 1848 ::write(fd, result.string(), result.size()); 1849 return NO_ERROR; 1850} 1851 1852uint32_t AudioTrack::getUnderrunFrames() const 1853{ 1854 AutoMutex lock(mLock); 1855 return mProxy->getUnderrunFrames(); 1856} 1857 1858// ========================================================================= 1859 1860void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1861{ 1862 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1863 if (audioTrack != 0) { 1864 AutoMutex lock(audioTrack->mLock); 1865 audioTrack->mProxy->binderDied(); 1866 } 1867} 1868 1869// ========================================================================= 1870 1871AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1872 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1873 mIgnoreNextPausedInt(false) 1874{ 1875} 1876 1877AudioTrack::AudioTrackThread::~AudioTrackThread() 1878{ 1879} 1880 1881bool AudioTrack::AudioTrackThread::threadLoop() 1882{ 1883 { 1884 AutoMutex _l(mMyLock); 1885 if (mPaused) { 1886 mMyCond.wait(mMyLock); 1887 // caller will check for exitPending() 1888 return true; 1889 } 1890 if (mIgnoreNextPausedInt) { 1891 mIgnoreNextPausedInt = false; 1892 mPausedInt = false; 1893 } 1894 if (mPausedInt) { 1895 if (mPausedNs > 0) { 1896 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1897 } else { 1898 mMyCond.wait(mMyLock); 1899 } 1900 mPausedInt = false; 1901 return true; 1902 } 1903 } 1904 nsecs_t ns = mReceiver.processAudioBuffer(); 1905 switch (ns) { 1906 case 0: 1907 return true; 1908 case NS_INACTIVE: 1909 pauseInternal(); 1910 return true; 1911 case NS_NEVER: 1912 return false; 1913 case NS_WHENEVER: 1914 // FIXME increase poll interval, or make event-driven 1915 ns = 1000000000LL; 1916 // fall through 1917 default: 1918 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1919 pauseInternal(ns); 1920 return true; 1921 } 1922} 1923 1924void AudioTrack::AudioTrackThread::requestExit() 1925{ 1926 // must be in this order to avoid a race condition 1927 Thread::requestExit(); 1928 resume(); 1929} 1930 1931void AudioTrack::AudioTrackThread::pause() 1932{ 1933 AutoMutex _l(mMyLock); 1934 mPaused = true; 1935} 1936 1937void AudioTrack::AudioTrackThread::resume() 1938{ 1939 AutoMutex _l(mMyLock); 1940 mIgnoreNextPausedInt = true; 1941 if (mPaused || mPausedInt) { 1942 mPaused = false; 1943 mPausedInt = false; 1944 mMyCond.signal(); 1945 } 1946} 1947 1948void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1949{ 1950 AutoMutex _l(mMyLock); 1951 mPausedInt = true; 1952 mPausedNs = ns; 1953} 1954 1955}; // namespace android 1956