Threads.cpp revision 0cde076ddb283c84c3801a2df4cc3df99bd1577f
1/* 2** 3** Copyright 2012, 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_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 273 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 //FIXME: mStandby should be true here. Is this some kind of hack? 276 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 277 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 278 // mName will be set by concrete (non-virtual) subclass 279 mDeathRecipient(new PMDeathRecipient(this)) 280{ 281} 282 283AudioFlinger::ThreadBase::~ThreadBase() 284{ 285 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 286 for (size_t i = 0; i < mConfigEvents.size(); i++) { 287 delete mConfigEvents[i]; 288 } 289 mConfigEvents.clear(); 290 291 mParamCond.broadcast(); 292 // do not lock the mutex in destructor 293 releaseWakeLock_l(); 294 if (mPowerManager != 0) { 295 sp<IBinder> binder = mPowerManager->asBinder(); 296 binder->unlinkToDeath(mDeathRecipient); 297 } 298} 299 300status_t AudioFlinger::ThreadBase::readyToRun() 301{ 302 status_t status = initCheck(); 303 if (status == NO_ERROR) { 304 ALOGI("AudioFlinger's thread %p ready to run", this); 305 } else { 306 ALOGE("No working audio driver found."); 307 } 308 return status; 309} 310 311void AudioFlinger::ThreadBase::exit() 312{ 313 ALOGV("ThreadBase::exit"); 314 // do any cleanup required for exit to succeed 315 preExit(); 316 { 317 // This lock prevents the following race in thread (uniprocessor for illustration): 318 // if (!exitPending()) { 319 // // context switch from here to exit() 320 // // exit() calls requestExit(), what exitPending() observes 321 // // exit() calls signal(), which is dropped since no waiters 322 // // context switch back from exit() to here 323 // mWaitWorkCV.wait(...); 324 // // now thread is hung 325 // } 326 AutoMutex lock(mLock); 327 requestExit(); 328 mWaitWorkCV.broadcast(); 329 } 330 // When Thread::requestExitAndWait is made virtual and this method is renamed to 331 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 332 requestExitAndWait(); 333} 334 335status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 336{ 337 status_t status; 338 339 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 340 Mutex::Autolock _l(mLock); 341 342 mNewParameters.add(keyValuePairs); 343 mWaitWorkCV.signal(); 344 // wait condition with timeout in case the thread loop has exited 345 // before the request could be processed 346 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 347 status = mParamStatus; 348 mWaitWorkCV.signal(); 349 } else { 350 status = TIMED_OUT; 351 } 352 return status; 353} 354 355void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 356{ 357 Mutex::Autolock _l(mLock); 358 sendIoConfigEvent_l(event, param); 359} 360 361// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 362void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 363{ 364 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 365 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 366 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 367 param); 368 mWaitWorkCV.signal(); 369} 370 371// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 372void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 373{ 374 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 375 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 376 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 377 mConfigEvents.size(), pid, tid, prio); 378 mWaitWorkCV.signal(); 379} 380 381void AudioFlinger::ThreadBase::processConfigEvents() 382{ 383 Mutex::Autolock _l(mLock); 384 processConfigEvents_l(); 385} 386 387// post condition: mConfigEvents.isEmpty() 388void AudioFlinger::ThreadBase::processConfigEvents_l() 389{ 390 while (!mConfigEvents.isEmpty()) { 391 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 392 ConfigEvent *event = mConfigEvents[0]; 393 mConfigEvents.removeAt(0); 394 // release mLock before locking AudioFlinger mLock: lock order is always 395 // AudioFlinger then ThreadBase to avoid cross deadlock 396 mLock.unlock(); 397 switch (event->type()) { 398 case CFG_EVENT_PRIO: { 399 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 400 // FIXME Need to understand why this has be done asynchronously 401 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 402 true /*asynchronous*/); 403 if (err != 0) { 404 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 405 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 406 } 407 } break; 408 case CFG_EVENT_IO: { 409 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 410 { 411 Mutex::Autolock _l(mAudioFlinger->mLock); 412 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 413 } 414 } break; 415 default: 416 ALOGE("processConfigEvents() unknown event type %d", event->type()); 417 break; 418 } 419 delete event; 420 mLock.lock(); 421 } 422} 423 424void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 425{ 426 const size_t SIZE = 256; 427 char buffer[SIZE]; 428 String8 result; 429 430 bool locked = AudioFlinger::dumpTryLock(mLock); 431 if (!locked) { 432 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 433 write(fd, buffer, strlen(buffer)); 434 } 435 436 snprintf(buffer, SIZE, "io handle: %d\n", mId); 437 result.append(buffer); 438 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 439 result.append(buffer); 440 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 449 result.append(buffer); 450 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 451 result.append(buffer); 452 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 453 result.append(buffer); 454 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 455 result.append(buffer); 456 457 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 458 result.append(buffer); 459 result.append(" Index Command"); 460 for (size_t i = 0; i < mNewParameters.size(); ++i) { 461 snprintf(buffer, SIZE, "\n %02d ", i); 462 result.append(buffer); 463 result.append(mNewParameters[i]); 464 } 465 466 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 467 result.append(buffer); 468 for (size_t i = 0; i < mConfigEvents.size(); i++) { 469 mConfigEvents[i]->dump(buffer, SIZE); 470 result.append(buffer); 471 } 472 result.append("\n"); 473 474 write(fd, result.string(), result.size()); 475 476 if (locked) { 477 mLock.unlock(); 478 } 479} 480 481void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 482{ 483 const size_t SIZE = 256; 484 char buffer[SIZE]; 485 String8 result; 486 487 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 488 write(fd, buffer, strlen(buffer)); 489 490 for (size_t i = 0; i < mEffectChains.size(); ++i) { 491 sp<EffectChain> chain = mEffectChains[i]; 492 if (chain != 0) { 493 chain->dump(fd, args); 494 } 495 } 496} 497 498void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 499{ 500 Mutex::Autolock _l(mLock); 501 acquireWakeLock_l(uid); 502} 503 504String16 AudioFlinger::ThreadBase::getWakeLockTag() 505{ 506 switch (mType) { 507 case MIXER: 508 return String16("AudioMix"); 509 case DIRECT: 510 return String16("AudioDirectOut"); 511 case DUPLICATING: 512 return String16("AudioDup"); 513 case RECORD: 514 return String16("AudioIn"); 515 case OFFLOAD: 516 return String16("AudioOffload"); 517 default: 518 ALOG_ASSERT(false); 519 return String16("AudioUnknown"); 520 } 521} 522 523void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 524{ 525 getPowerManager_l(); 526 if (mPowerManager != 0) { 527 sp<IBinder> binder = new BBinder(); 528 status_t status; 529 if (uid >= 0) { 530 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 531 binder, 532 getWakeLockTag(), 533 String16("media"), 534 uid); 535 } else { 536 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 537 binder, 538 getWakeLockTag(), 539 String16("media")); 540 } 541 if (status == NO_ERROR) { 542 mWakeLockToken = binder; 543 } 544 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 545 } 546} 547 548void AudioFlinger::ThreadBase::releaseWakeLock() 549{ 550 Mutex::Autolock _l(mLock); 551 releaseWakeLock_l(); 552} 553 554void AudioFlinger::ThreadBase::releaseWakeLock_l() 555{ 556 if (mWakeLockToken != 0) { 557 ALOGV("releaseWakeLock_l() %s", mName); 558 if (mPowerManager != 0) { 559 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 560 } 561 mWakeLockToken.clear(); 562 } 563} 564 565void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 566 Mutex::Autolock _l(mLock); 567 updateWakeLockUids_l(uids); 568} 569 570void AudioFlinger::ThreadBase::getPowerManager_l() { 571 572 if (mPowerManager == 0) { 573 // use checkService() to avoid blocking if power service is not up yet 574 sp<IBinder> binder = 575 defaultServiceManager()->checkService(String16("power")); 576 if (binder == 0) { 577 ALOGW("Thread %s cannot connect to the power manager service", mName); 578 } else { 579 mPowerManager = interface_cast<IPowerManager>(binder); 580 binder->linkToDeath(mDeathRecipient); 581 } 582 } 583} 584 585void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 586 587 getPowerManager_l(); 588 if (mWakeLockToken == NULL) { 589 ALOGE("no wake lock to update!"); 590 return; 591 } 592 if (mPowerManager != 0) { 593 sp<IBinder> binder = new BBinder(); 594 status_t status; 595 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 596 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 597 } 598} 599 600void AudioFlinger::ThreadBase::clearPowerManager() 601{ 602 Mutex::Autolock _l(mLock); 603 releaseWakeLock_l(); 604 mPowerManager.clear(); 605} 606 607void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 608{ 609 sp<ThreadBase> thread = mThread.promote(); 610 if (thread != 0) { 611 thread->clearPowerManager(); 612 } 613 ALOGW("power manager service died !!!"); 614} 615 616void AudioFlinger::ThreadBase::setEffectSuspended( 617 const effect_uuid_t *type, bool suspend, int sessionId) 618{ 619 Mutex::Autolock _l(mLock); 620 setEffectSuspended_l(type, suspend, sessionId); 621} 622 623void AudioFlinger::ThreadBase::setEffectSuspended_l( 624 const effect_uuid_t *type, bool suspend, int sessionId) 625{ 626 sp<EffectChain> chain = getEffectChain_l(sessionId); 627 if (chain != 0) { 628 if (type != NULL) { 629 chain->setEffectSuspended_l(type, suspend); 630 } else { 631 chain->setEffectSuspendedAll_l(suspend); 632 } 633 } 634 635 updateSuspendedSessions_l(type, suspend, sessionId); 636} 637 638void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 639{ 640 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 641 if (index < 0) { 642 return; 643 } 644 645 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 646 mSuspendedSessions.valueAt(index); 647 648 for (size_t i = 0; i < sessionEffects.size(); i++) { 649 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 650 for (int j = 0; j < desc->mRefCount; j++) { 651 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 652 chain->setEffectSuspendedAll_l(true); 653 } else { 654 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 655 desc->mType.timeLow); 656 chain->setEffectSuspended_l(&desc->mType, true); 657 } 658 } 659 } 660} 661 662void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 663 bool suspend, 664 int sessionId) 665{ 666 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 667 668 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 669 670 if (suspend) { 671 if (index >= 0) { 672 sessionEffects = mSuspendedSessions.valueAt(index); 673 } else { 674 mSuspendedSessions.add(sessionId, sessionEffects); 675 } 676 } else { 677 if (index < 0) { 678 return; 679 } 680 sessionEffects = mSuspendedSessions.valueAt(index); 681 } 682 683 684 int key = EffectChain::kKeyForSuspendAll; 685 if (type != NULL) { 686 key = type->timeLow; 687 } 688 index = sessionEffects.indexOfKey(key); 689 690 sp<SuspendedSessionDesc> desc; 691 if (suspend) { 692 if (index >= 0) { 693 desc = sessionEffects.valueAt(index); 694 } else { 695 desc = new SuspendedSessionDesc(); 696 if (type != NULL) { 697 desc->mType = *type; 698 } 699 sessionEffects.add(key, desc); 700 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 701 } 702 desc->mRefCount++; 703 } else { 704 if (index < 0) { 705 return; 706 } 707 desc = sessionEffects.valueAt(index); 708 if (--desc->mRefCount == 0) { 709 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 710 sessionEffects.removeItemsAt(index); 711 if (sessionEffects.isEmpty()) { 712 ALOGV("updateSuspendedSessions_l() restore removing session %d", 713 sessionId); 714 mSuspendedSessions.removeItem(sessionId); 715 } 716 } 717 } 718 if (!sessionEffects.isEmpty()) { 719 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 720 } 721} 722 723void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 724 bool enabled, 725 int sessionId) 726{ 727 Mutex::Autolock _l(mLock); 728 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 729} 730 731void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 732 bool enabled, 733 int sessionId) 734{ 735 if (mType != RECORD) { 736 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 737 // another session. This gives the priority to well behaved effect control panels 738 // and applications not using global effects. 739 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 740 // global effects 741 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 742 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 743 } 744 } 745 746 sp<EffectChain> chain = getEffectChain_l(sessionId); 747 if (chain != 0) { 748 chain->checkSuspendOnEffectEnabled(effect, enabled); 749 } 750} 751 752// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 753sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 754 const sp<AudioFlinger::Client>& client, 755 const sp<IEffectClient>& effectClient, 756 int32_t priority, 757 int sessionId, 758 effect_descriptor_t *desc, 759 int *enabled, 760 status_t *status) 761{ 762 sp<EffectModule> effect; 763 sp<EffectHandle> handle; 764 status_t lStatus; 765 sp<EffectChain> chain; 766 bool chainCreated = false; 767 bool effectCreated = false; 768 bool effectRegistered = false; 769 770 lStatus = initCheck(); 771 if (lStatus != NO_ERROR) { 772 ALOGW("createEffect_l() Audio driver not initialized."); 773 goto Exit; 774 } 775 776 // Allow global effects only on offloaded and mixer threads 777 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 778 switch (mType) { 779 case MIXER: 780 case OFFLOAD: 781 break; 782 case DIRECT: 783 case DUPLICATING: 784 case RECORD: 785 default: 786 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 787 lStatus = BAD_VALUE; 788 goto Exit; 789 } 790 } 791 792 // Only Pre processor effects are allowed on input threads and only on input threads 793 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 794 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 795 desc->name, desc->flags, mType); 796 lStatus = BAD_VALUE; 797 goto Exit; 798 } 799 800 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 801 802 { // scope for mLock 803 Mutex::Autolock _l(mLock); 804 805 // check for existing effect chain with the requested audio session 806 chain = getEffectChain_l(sessionId); 807 if (chain == 0) { 808 // create a new chain for this session 809 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 810 chain = new EffectChain(this, sessionId); 811 addEffectChain_l(chain); 812 chain->setStrategy(getStrategyForSession_l(sessionId)); 813 chainCreated = true; 814 } else { 815 effect = chain->getEffectFromDesc_l(desc); 816 } 817 818 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 819 820 if (effect == 0) { 821 int id = mAudioFlinger->nextUniqueId(); 822 // Check CPU and memory usage 823 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 824 if (lStatus != NO_ERROR) { 825 goto Exit; 826 } 827 effectRegistered = true; 828 // create a new effect module if none present in the chain 829 effect = new EffectModule(this, chain, desc, id, sessionId); 830 lStatus = effect->status(); 831 if (lStatus != NO_ERROR) { 832 goto Exit; 833 } 834 effect->setOffloaded(mType == OFFLOAD, mId); 835 836 lStatus = chain->addEffect_l(effect); 837 if (lStatus != NO_ERROR) { 838 goto Exit; 839 } 840 effectCreated = true; 841 842 effect->setDevice(mOutDevice); 843 effect->setDevice(mInDevice); 844 effect->setMode(mAudioFlinger->getMode()); 845 effect->setAudioSource(mAudioSource); 846 } 847 // create effect handle and connect it to effect module 848 handle = new EffectHandle(effect, client, effectClient, priority); 849 lStatus = handle->initCheck(); 850 if (lStatus == OK) { 851 lStatus = effect->addHandle(handle.get()); 852 } 853 if (enabled != NULL) { 854 *enabled = (int)effect->isEnabled(); 855 } 856 } 857 858Exit: 859 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 860 Mutex::Autolock _l(mLock); 861 if (effectCreated) { 862 chain->removeEffect_l(effect); 863 } 864 if (effectRegistered) { 865 AudioSystem::unregisterEffect(effect->id()); 866 } 867 if (chainCreated) { 868 removeEffectChain_l(chain); 869 } 870 handle.clear(); 871 } 872 873 *status = lStatus; 874 return handle; 875} 876 877sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 878{ 879 Mutex::Autolock _l(mLock); 880 return getEffect_l(sessionId, effectId); 881} 882 883sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 884{ 885 sp<EffectChain> chain = getEffectChain_l(sessionId); 886 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 887} 888 889// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 890// PlaybackThread::mLock held 891status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 892{ 893 // check for existing effect chain with the requested audio session 894 int sessionId = effect->sessionId(); 895 sp<EffectChain> chain = getEffectChain_l(sessionId); 896 bool chainCreated = false; 897 898 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 899 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 900 this, effect->desc().name, effect->desc().flags); 901 902 if (chain == 0) { 903 // create a new chain for this session 904 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 905 chain = new EffectChain(this, sessionId); 906 addEffectChain_l(chain); 907 chain->setStrategy(getStrategyForSession_l(sessionId)); 908 chainCreated = true; 909 } 910 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 911 912 if (chain->getEffectFromId_l(effect->id()) != 0) { 913 ALOGW("addEffect_l() %p effect %s already present in chain %p", 914 this, effect->desc().name, chain.get()); 915 return BAD_VALUE; 916 } 917 918 effect->setOffloaded(mType == OFFLOAD, mId); 919 920 status_t status = chain->addEffect_l(effect); 921 if (status != NO_ERROR) { 922 if (chainCreated) { 923 removeEffectChain_l(chain); 924 } 925 return status; 926 } 927 928 effect->setDevice(mOutDevice); 929 effect->setDevice(mInDevice); 930 effect->setMode(mAudioFlinger->getMode()); 931 effect->setAudioSource(mAudioSource); 932 return NO_ERROR; 933} 934 935void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 936 937 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 938 effect_descriptor_t desc = effect->desc(); 939 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 940 detachAuxEffect_l(effect->id()); 941 } 942 943 sp<EffectChain> chain = effect->chain().promote(); 944 if (chain != 0) { 945 // remove effect chain if removing last effect 946 if (chain->removeEffect_l(effect) == 0) { 947 removeEffectChain_l(chain); 948 } 949 } else { 950 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 951 } 952} 953 954void AudioFlinger::ThreadBase::lockEffectChains_l( 955 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 956{ 957 effectChains = mEffectChains; 958 for (size_t i = 0; i < mEffectChains.size(); i++) { 959 mEffectChains[i]->lock(); 960 } 961} 962 963void AudioFlinger::ThreadBase::unlockEffectChains( 964 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 965{ 966 for (size_t i = 0; i < effectChains.size(); i++) { 967 effectChains[i]->unlock(); 968 } 969} 970 971sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 972{ 973 Mutex::Autolock _l(mLock); 974 return getEffectChain_l(sessionId); 975} 976 977sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 978{ 979 size_t size = mEffectChains.size(); 980 for (size_t i = 0; i < size; i++) { 981 if (mEffectChains[i]->sessionId() == sessionId) { 982 return mEffectChains[i]; 983 } 984 } 985 return 0; 986} 987 988void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 989{ 990 Mutex::Autolock _l(mLock); 991 size_t size = mEffectChains.size(); 992 for (size_t i = 0; i < size; i++) { 993 mEffectChains[i]->setMode_l(mode); 994 } 995} 996 997void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 998 EffectHandle *handle, 999 bool unpinIfLast) { 1000 1001 Mutex::Autolock _l(mLock); 1002 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1003 // delete the effect module if removing last handle on it 1004 if (effect->removeHandle(handle) == 0) { 1005 if (!effect->isPinned() || unpinIfLast) { 1006 removeEffect_l(effect); 1007 AudioSystem::unregisterEffect(effect->id()); 1008 } 1009 } 1010} 1011 1012// ---------------------------------------------------------------------------- 1013// Playback 1014// ---------------------------------------------------------------------------- 1015 1016AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1017 AudioStreamOut* output, 1018 audio_io_handle_t id, 1019 audio_devices_t device, 1020 type_t type) 1021 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1022 mNormalFrameCount(0), mMixBuffer(NULL), 1023 mSuspended(0), mBytesWritten(0), 1024 mActiveTracksGeneration(0), 1025 // mStreamTypes[] initialized in constructor body 1026 mOutput(output), 1027 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1028 mMixerStatus(MIXER_IDLE), 1029 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1030 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1031 mBytesRemaining(0), 1032 mCurrentWriteLength(0), 1033 mUseAsyncWrite(false), 1034 mWriteAckSequence(0), 1035 mDrainSequence(0), 1036 mSignalPending(false), 1037 mScreenState(AudioFlinger::mScreenState), 1038 // index 0 is reserved for normal mixer's submix 1039 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1040 // mLatchD, mLatchQ, 1041 mLatchDValid(false), mLatchQValid(false) 1042{ 1043 snprintf(mName, kNameLength, "AudioOut_%X", id); 1044 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1045 1046 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1047 // it would be safer to explicitly pass initial masterVolume/masterMute as 1048 // parameter. 1049 // 1050 // If the HAL we are using has support for master volume or master mute, 1051 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1052 // and the mute set to false). 1053 mMasterVolume = audioFlinger->masterVolume_l(); 1054 mMasterMute = audioFlinger->masterMute_l(); 1055 if (mOutput && mOutput->audioHwDev) { 1056 if (mOutput->audioHwDev->canSetMasterVolume()) { 1057 mMasterVolume = 1.0; 1058 } 1059 1060 if (mOutput->audioHwDev->canSetMasterMute()) { 1061 mMasterMute = false; 1062 } 1063 } 1064 1065 readOutputParameters(); 1066 1067 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1068 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1069 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1070 stream = (audio_stream_type_t) (stream + 1)) { 1071 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1072 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1073 } 1074 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1075 // because mAudioFlinger doesn't have one to copy from 1076} 1077 1078AudioFlinger::PlaybackThread::~PlaybackThread() 1079{ 1080 mAudioFlinger->unregisterWriter(mNBLogWriter); 1081 delete[] mMixBuffer; 1082} 1083 1084void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1085{ 1086 dumpInternals(fd, args); 1087 dumpTracks(fd, args); 1088 dumpEffectChains(fd, args); 1089} 1090 1091void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1092{ 1093 const size_t SIZE = 256; 1094 char buffer[SIZE]; 1095 String8 result; 1096 1097 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1098 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1099 const stream_type_t *st = &mStreamTypes[i]; 1100 if (i > 0) { 1101 result.appendFormat(", "); 1102 } 1103 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1104 if (st->mute) { 1105 result.append("M"); 1106 } 1107 } 1108 result.append("\n"); 1109 write(fd, result.string(), result.length()); 1110 result.clear(); 1111 1112 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1113 result.append(buffer); 1114 Track::appendDumpHeader(result); 1115 for (size_t i = 0; i < mTracks.size(); ++i) { 1116 sp<Track> track = mTracks[i]; 1117 if (track != 0) { 1118 track->dump(buffer, SIZE); 1119 result.append(buffer); 1120 } 1121 } 1122 1123 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1124 result.append(buffer); 1125 Track::appendDumpHeader(result); 1126 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1127 sp<Track> track = mActiveTracks[i].promote(); 1128 if (track != 0) { 1129 track->dump(buffer, SIZE); 1130 result.append(buffer); 1131 } 1132 } 1133 write(fd, result.string(), result.size()); 1134 1135 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1136 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1137 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1138 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1139} 1140 1141void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1142{ 1143 const size_t SIZE = 256; 1144 char buffer[SIZE]; 1145 String8 result; 1146 1147 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1148 result.append(buffer); 1149 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1150 result.append(buffer); 1151 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1152 ns2ms(systemTime() - mLastWriteTime)); 1153 result.append(buffer); 1154 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1155 result.append(buffer); 1156 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1157 result.append(buffer); 1158 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1159 result.append(buffer); 1160 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1161 result.append(buffer); 1162 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1163 result.append(buffer); 1164 write(fd, result.string(), result.size()); 1165 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1166 1167 dumpBase(fd, args); 1168} 1169 1170// Thread virtuals 1171 1172void AudioFlinger::PlaybackThread::onFirstRef() 1173{ 1174 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1175} 1176 1177// ThreadBase virtuals 1178void AudioFlinger::PlaybackThread::preExit() 1179{ 1180 ALOGV(" preExit()"); 1181 // FIXME this is using hard-coded strings but in the future, this functionality will be 1182 // converted to use audio HAL extensions required to support tunneling 1183 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1184} 1185 1186// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1187sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1188 const sp<AudioFlinger::Client>& client, 1189 audio_stream_type_t streamType, 1190 uint32_t sampleRate, 1191 audio_format_t format, 1192 audio_channel_mask_t channelMask, 1193 size_t frameCount, 1194 const sp<IMemory>& sharedBuffer, 1195 int sessionId, 1196 IAudioFlinger::track_flags_t *flags, 1197 pid_t tid, 1198 int uid, 1199 status_t *status) 1200{ 1201 sp<Track> track; 1202 status_t lStatus; 1203 1204 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1205 1206 // client expresses a preference for FAST, but we get the final say 1207 if (*flags & IAudioFlinger::TRACK_FAST) { 1208 if ( 1209 // not timed 1210 (!isTimed) && 1211 // either of these use cases: 1212 ( 1213 // use case 1: shared buffer with any frame count 1214 ( 1215 (sharedBuffer != 0) 1216 ) || 1217 // use case 2: callback handler and frame count is default or at least as large as HAL 1218 ( 1219 (tid != -1) && 1220 ((frameCount == 0) || 1221 (frameCount >= mFrameCount)) 1222 ) 1223 ) && 1224 // PCM data 1225 audio_is_linear_pcm(format) && 1226 // mono or stereo 1227 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1228 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1229#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1230 // hardware sample rate 1231 (sampleRate == mSampleRate) && 1232#endif 1233 // normal mixer has an associated fast mixer 1234 hasFastMixer() && 1235 // there are sufficient fast track slots available 1236 (mFastTrackAvailMask != 0) 1237 // FIXME test that MixerThread for this fast track has a capable output HAL 1238 // FIXME add a permission test also? 1239 ) { 1240 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1241 if (frameCount == 0) { 1242 frameCount = mFrameCount * kFastTrackMultiplier; 1243 } 1244 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1245 frameCount, mFrameCount); 1246 } else { 1247 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1248 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1249 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1250 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1251 audio_is_linear_pcm(format), 1252 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1253 *flags &= ~IAudioFlinger::TRACK_FAST; 1254 // For compatibility with AudioTrack calculation, buffer depth is forced 1255 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1256 // This is probably too conservative, but legacy application code may depend on it. 1257 // If you change this calculation, also review the start threshold which is related. 1258 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1259 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1260 if (minBufCount < 2) { 1261 minBufCount = 2; 1262 } 1263 size_t minFrameCount = mNormalFrameCount * minBufCount; 1264 if (frameCount < minFrameCount) { 1265 frameCount = minFrameCount; 1266 } 1267 } 1268 } 1269 1270 if (mType == DIRECT) { 1271 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1272 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1273 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1274 "for output %p with format %d", 1275 sampleRate, format, channelMask, mOutput, mFormat); 1276 lStatus = BAD_VALUE; 1277 goto Exit; 1278 } 1279 } 1280 } else if (mType == OFFLOAD) { 1281 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1282 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1283 "for output %p with format %d", 1284 sampleRate, format, channelMask, mOutput, mFormat); 1285 lStatus = BAD_VALUE; 1286 goto Exit; 1287 } 1288 } else { 1289 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1290 ALOGE("createTrack_l() Bad parameter: format %d \"" 1291 "for output %p with format %d", 1292 format, mOutput, mFormat); 1293 lStatus = BAD_VALUE; 1294 goto Exit; 1295 } 1296 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1297 if (sampleRate > mSampleRate*2) { 1298 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1299 lStatus = BAD_VALUE; 1300 goto Exit; 1301 } 1302 } 1303 1304 lStatus = initCheck(); 1305 if (lStatus != NO_ERROR) { 1306 ALOGE("Audio driver not initialized."); 1307 goto Exit; 1308 } 1309 1310 { // scope for mLock 1311 Mutex::Autolock _l(mLock); 1312 1313 // all tracks in same audio session must share the same routing strategy otherwise 1314 // conflicts will happen when tracks are moved from one output to another by audio policy 1315 // manager 1316 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1317 for (size_t i = 0; i < mTracks.size(); ++i) { 1318 sp<Track> t = mTracks[i]; 1319 if (t != 0 && !t->isOutputTrack()) { 1320 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1321 if (sessionId == t->sessionId() && strategy != actual) { 1322 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1323 strategy, actual); 1324 lStatus = BAD_VALUE; 1325 goto Exit; 1326 } 1327 } 1328 } 1329 1330 if (!isTimed) { 1331 track = new Track(this, client, streamType, sampleRate, format, 1332 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1333 } else { 1334 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1335 channelMask, frameCount, sharedBuffer, sessionId, uid); 1336 } 1337 1338 // new Track always returns non-NULL, 1339 // but TimedTrack::create() is a factory that could fail by returning NULL 1340 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1341 if (lStatus != NO_ERROR) { 1342 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1343 track.clear(); 1344 goto Exit; 1345 } 1346 1347 mTracks.add(track); 1348 1349 sp<EffectChain> chain = getEffectChain_l(sessionId); 1350 if (chain != 0) { 1351 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1352 track->setMainBuffer(chain->inBuffer()); 1353 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1354 chain->incTrackCnt(); 1355 } 1356 1357 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1358 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1359 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1360 // so ask activity manager to do this on our behalf 1361 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1362 } 1363 } 1364 1365 lStatus = NO_ERROR; 1366 1367Exit: 1368 *status = lStatus; 1369 return track; 1370} 1371 1372uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1373{ 1374 return latency; 1375} 1376 1377uint32_t AudioFlinger::PlaybackThread::latency() const 1378{ 1379 Mutex::Autolock _l(mLock); 1380 return latency_l(); 1381} 1382uint32_t AudioFlinger::PlaybackThread::latency_l() const 1383{ 1384 if (initCheck() == NO_ERROR) { 1385 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1386 } else { 1387 return 0; 1388 } 1389} 1390 1391void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1392{ 1393 Mutex::Autolock _l(mLock); 1394 // Don't apply master volume in SW if our HAL can do it for us. 1395 if (mOutput && mOutput->audioHwDev && 1396 mOutput->audioHwDev->canSetMasterVolume()) { 1397 mMasterVolume = 1.0; 1398 } else { 1399 mMasterVolume = value; 1400 } 1401} 1402 1403void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1404{ 1405 Mutex::Autolock _l(mLock); 1406 // Don't apply master mute in SW if our HAL can do it for us. 1407 if (mOutput && mOutput->audioHwDev && 1408 mOutput->audioHwDev->canSetMasterMute()) { 1409 mMasterMute = false; 1410 } else { 1411 mMasterMute = muted; 1412 } 1413} 1414 1415void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1416{ 1417 Mutex::Autolock _l(mLock); 1418 mStreamTypes[stream].volume = value; 1419 broadcast_l(); 1420} 1421 1422void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1423{ 1424 Mutex::Autolock _l(mLock); 1425 mStreamTypes[stream].mute = muted; 1426 broadcast_l(); 1427} 1428 1429float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1430{ 1431 Mutex::Autolock _l(mLock); 1432 return mStreamTypes[stream].volume; 1433} 1434 1435// addTrack_l() must be called with ThreadBase::mLock held 1436status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1437{ 1438 status_t status = ALREADY_EXISTS; 1439 1440 // set retry count for buffer fill 1441 track->mRetryCount = kMaxTrackStartupRetries; 1442 if (mActiveTracks.indexOf(track) < 0) { 1443 // the track is newly added, make sure it fills up all its 1444 // buffers before playing. This is to ensure the client will 1445 // effectively get the latency it requested. 1446 if (!track->isOutputTrack()) { 1447 TrackBase::track_state state = track->mState; 1448 mLock.unlock(); 1449 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1450 mLock.lock(); 1451 // abort track was stopped/paused while we released the lock 1452 if (state != track->mState) { 1453 if (status == NO_ERROR) { 1454 mLock.unlock(); 1455 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1456 mLock.lock(); 1457 } 1458 return INVALID_OPERATION; 1459 } 1460 // abort if start is rejected by audio policy manager 1461 if (status != NO_ERROR) { 1462 return PERMISSION_DENIED; 1463 } 1464#ifdef ADD_BATTERY_DATA 1465 // to track the speaker usage 1466 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1467#endif 1468 } 1469 1470 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1471 track->mResetDone = false; 1472 track->mPresentationCompleteFrames = 0; 1473 mActiveTracks.add(track); 1474 mWakeLockUids.add(track->uid()); 1475 mActiveTracksGeneration++; 1476 mLatestActiveTrack = track; 1477 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1478 if (chain != 0) { 1479 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1480 track->sessionId()); 1481 chain->incActiveTrackCnt(); 1482 } 1483 1484 status = NO_ERROR; 1485 } 1486 1487 ALOGV("signal playback thread"); 1488 broadcast_l(); 1489 1490 return status; 1491} 1492 1493bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1494{ 1495 track->terminate(); 1496 // active tracks are removed by threadLoop() 1497 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1498 track->mState = TrackBase::STOPPED; 1499 if (!trackActive) { 1500 removeTrack_l(track); 1501 } else if (track->isFastTrack() || track->isOffloaded()) { 1502 track->mState = TrackBase::STOPPING_1; 1503 } 1504 1505 return trackActive; 1506} 1507 1508void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1509{ 1510 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1511 mTracks.remove(track); 1512 deleteTrackName_l(track->name()); 1513 // redundant as track is about to be destroyed, for dumpsys only 1514 track->mName = -1; 1515 if (track->isFastTrack()) { 1516 int index = track->mFastIndex; 1517 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1518 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1519 mFastTrackAvailMask |= 1 << index; 1520 // redundant as track is about to be destroyed, for dumpsys only 1521 track->mFastIndex = -1; 1522 } 1523 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1524 if (chain != 0) { 1525 chain->decTrackCnt(); 1526 } 1527} 1528 1529void AudioFlinger::PlaybackThread::broadcast_l() 1530{ 1531 // Thread could be blocked waiting for async 1532 // so signal it to handle state changes immediately 1533 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1534 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1535 mSignalPending = true; 1536 mWaitWorkCV.broadcast(); 1537} 1538 1539String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1540{ 1541 Mutex::Autolock _l(mLock); 1542 if (initCheck() != NO_ERROR) { 1543 return String8(); 1544 } 1545 1546 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1547 const String8 out_s8(s); 1548 free(s); 1549 return out_s8; 1550} 1551 1552// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1553void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1554 AudioSystem::OutputDescriptor desc; 1555 void *param2 = NULL; 1556 1557 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1558 param); 1559 1560 switch (event) { 1561 case AudioSystem::OUTPUT_OPENED: 1562 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1563 desc.channelMask = mChannelMask; 1564 desc.samplingRate = mSampleRate; 1565 desc.format = mFormat; 1566 desc.frameCount = mNormalFrameCount; // FIXME see 1567 // AudioFlinger::frameCount(audio_io_handle_t) 1568 desc.latency = latency(); 1569 param2 = &desc; 1570 break; 1571 1572 case AudioSystem::STREAM_CONFIG_CHANGED: 1573 param2 = ¶m; 1574 case AudioSystem::OUTPUT_CLOSED: 1575 default: 1576 break; 1577 } 1578 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1579} 1580 1581void AudioFlinger::PlaybackThread::writeCallback() 1582{ 1583 ALOG_ASSERT(mCallbackThread != 0); 1584 mCallbackThread->resetWriteBlocked(); 1585} 1586 1587void AudioFlinger::PlaybackThread::drainCallback() 1588{ 1589 ALOG_ASSERT(mCallbackThread != 0); 1590 mCallbackThread->resetDraining(); 1591} 1592 1593void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1594{ 1595 Mutex::Autolock _l(mLock); 1596 // reject out of sequence requests 1597 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1598 mWriteAckSequence &= ~1; 1599 mWaitWorkCV.signal(); 1600 } 1601} 1602 1603void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1604{ 1605 Mutex::Autolock _l(mLock); 1606 // reject out of sequence requests 1607 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1608 mDrainSequence &= ~1; 1609 mWaitWorkCV.signal(); 1610 } 1611} 1612 1613// static 1614int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1615 void *param, 1616 void *cookie) 1617{ 1618 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1619 ALOGV("asyncCallback() event %d", event); 1620 switch (event) { 1621 case STREAM_CBK_EVENT_WRITE_READY: 1622 me->writeCallback(); 1623 break; 1624 case STREAM_CBK_EVENT_DRAIN_READY: 1625 me->drainCallback(); 1626 break; 1627 default: 1628 ALOGW("asyncCallback() unknown event %d", event); 1629 break; 1630 } 1631 return 0; 1632} 1633 1634void AudioFlinger::PlaybackThread::readOutputParameters() 1635{ 1636 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1637 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1638 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1639 if (!audio_is_output_channel(mChannelMask)) { 1640 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1641 } 1642 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1643 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1644 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1645 } 1646 mChannelCount = popcount(mChannelMask); 1647 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1648 if (!audio_is_valid_format(mFormat)) { 1649 LOG_FATAL("HAL format %d not valid for output", mFormat); 1650 } 1651 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1652 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1653 mFormat); 1654 } 1655 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1656 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1657 mFrameCount = mBufferSize / mFrameSize; 1658 if (mFrameCount & 15) { 1659 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1660 mFrameCount); 1661 } 1662 1663 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1664 (mOutput->stream->set_callback != NULL)) { 1665 if (mOutput->stream->set_callback(mOutput->stream, 1666 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1667 mUseAsyncWrite = true; 1668 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1669 } 1670 } 1671 1672 // Calculate size of normal mix buffer relative to the HAL output buffer size 1673 double multiplier = 1.0; 1674 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1675 kUseFastMixer == FastMixer_Dynamic)) { 1676 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1677 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1678 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1679 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1680 maxNormalFrameCount = maxNormalFrameCount & ~15; 1681 if (maxNormalFrameCount < minNormalFrameCount) { 1682 maxNormalFrameCount = minNormalFrameCount; 1683 } 1684 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1685 if (multiplier <= 1.0) { 1686 multiplier = 1.0; 1687 } else if (multiplier <= 2.0) { 1688 if (2 * mFrameCount <= maxNormalFrameCount) { 1689 multiplier = 2.0; 1690 } else { 1691 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1692 } 1693 } else { 1694 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1695 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1696 // track, but we sometimes have to do this to satisfy the maximum frame count 1697 // constraint) 1698 // FIXME this rounding up should not be done if no HAL SRC 1699 uint32_t truncMult = (uint32_t) multiplier; 1700 if ((truncMult & 1)) { 1701 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1702 ++truncMult; 1703 } 1704 } 1705 multiplier = (double) truncMult; 1706 } 1707 } 1708 mNormalFrameCount = multiplier * mFrameCount; 1709 // round up to nearest 16 frames to satisfy AudioMixer 1710 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1711 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1712 mNormalFrameCount); 1713 1714 delete[] mMixBuffer; 1715 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1716 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1717 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1718 memset(mMixBuffer, 0, normalBufferSize); 1719 1720 // force reconfiguration of effect chains and engines to take new buffer size and audio 1721 // parameters into account 1722 // Note that mLock is not held when readOutputParameters() is called from the constructor 1723 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1724 // matter. 1725 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1726 Vector< sp<EffectChain> > effectChains = mEffectChains; 1727 for (size_t i = 0; i < effectChains.size(); i ++) { 1728 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1729 } 1730} 1731 1732 1733status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1734{ 1735 if (halFrames == NULL || dspFrames == NULL) { 1736 return BAD_VALUE; 1737 } 1738 Mutex::Autolock _l(mLock); 1739 if (initCheck() != NO_ERROR) { 1740 return INVALID_OPERATION; 1741 } 1742 size_t framesWritten = mBytesWritten / mFrameSize; 1743 *halFrames = framesWritten; 1744 1745 if (isSuspended()) { 1746 // return an estimation of rendered frames when the output is suspended 1747 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1748 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1749 return NO_ERROR; 1750 } else { 1751 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1752 } 1753} 1754 1755uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1756{ 1757 Mutex::Autolock _l(mLock); 1758 uint32_t result = 0; 1759 if (getEffectChain_l(sessionId) != 0) { 1760 result = EFFECT_SESSION; 1761 } 1762 1763 for (size_t i = 0; i < mTracks.size(); ++i) { 1764 sp<Track> track = mTracks[i]; 1765 if (sessionId == track->sessionId() && !track->isInvalid()) { 1766 result |= TRACK_SESSION; 1767 break; 1768 } 1769 } 1770 1771 return result; 1772} 1773 1774uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1775{ 1776 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1777 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1778 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1779 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1780 } 1781 for (size_t i = 0; i < mTracks.size(); i++) { 1782 sp<Track> track = mTracks[i]; 1783 if (sessionId == track->sessionId() && !track->isInvalid()) { 1784 return AudioSystem::getStrategyForStream(track->streamType()); 1785 } 1786 } 1787 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1788} 1789 1790 1791AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1792{ 1793 Mutex::Autolock _l(mLock); 1794 return mOutput; 1795} 1796 1797AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1798{ 1799 Mutex::Autolock _l(mLock); 1800 AudioStreamOut *output = mOutput; 1801 mOutput = NULL; 1802 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1803 // must push a NULL and wait for ack 1804 mOutputSink.clear(); 1805 mPipeSink.clear(); 1806 mNormalSink.clear(); 1807 return output; 1808} 1809 1810// this method must always be called either with ThreadBase mLock held or inside the thread loop 1811audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1812{ 1813 if (mOutput == NULL) { 1814 return NULL; 1815 } 1816 return &mOutput->stream->common; 1817} 1818 1819uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1820{ 1821 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1822} 1823 1824status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1825{ 1826 if (!isValidSyncEvent(event)) { 1827 return BAD_VALUE; 1828 } 1829 1830 Mutex::Autolock _l(mLock); 1831 1832 for (size_t i = 0; i < mTracks.size(); ++i) { 1833 sp<Track> track = mTracks[i]; 1834 if (event->triggerSession() == track->sessionId()) { 1835 (void) track->setSyncEvent(event); 1836 return NO_ERROR; 1837 } 1838 } 1839 1840 return NAME_NOT_FOUND; 1841} 1842 1843bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1844{ 1845 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1846} 1847 1848void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1849 const Vector< sp<Track> >& tracksToRemove) 1850{ 1851 size_t count = tracksToRemove.size(); 1852 if (count > 0) { 1853 for (size_t i = 0 ; i < count ; i++) { 1854 const sp<Track>& track = tracksToRemove.itemAt(i); 1855 if (!track->isOutputTrack()) { 1856 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1857#ifdef ADD_BATTERY_DATA 1858 // to track the speaker usage 1859 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1860#endif 1861 if (track->isTerminated()) { 1862 AudioSystem::releaseOutput(mId); 1863 } 1864 } 1865 } 1866 } 1867} 1868 1869void AudioFlinger::PlaybackThread::checkSilentMode_l() 1870{ 1871 if (!mMasterMute) { 1872 char value[PROPERTY_VALUE_MAX]; 1873 if (property_get("ro.audio.silent", value, "0") > 0) { 1874 char *endptr; 1875 unsigned long ul = strtoul(value, &endptr, 0); 1876 if (*endptr == '\0' && ul != 0) { 1877 ALOGD("Silence is golden"); 1878 // The setprop command will not allow a property to be changed after 1879 // the first time it is set, so we don't have to worry about un-muting. 1880 setMasterMute_l(true); 1881 } 1882 } 1883 } 1884} 1885 1886// shared by MIXER and DIRECT, overridden by DUPLICATING 1887ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1888{ 1889 // FIXME rewrite to reduce number of system calls 1890 mLastWriteTime = systemTime(); 1891 mInWrite = true; 1892 ssize_t bytesWritten; 1893 1894 // If an NBAIO sink is present, use it to write the normal mixer's submix 1895 if (mNormalSink != 0) { 1896#define mBitShift 2 // FIXME 1897 size_t count = mBytesRemaining >> mBitShift; 1898 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1899 ATRACE_BEGIN("write"); 1900 // update the setpoint when AudioFlinger::mScreenState changes 1901 uint32_t screenState = AudioFlinger::mScreenState; 1902 if (screenState != mScreenState) { 1903 mScreenState = screenState; 1904 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1905 if (pipe != NULL) { 1906 pipe->setAvgFrames((mScreenState & 1) ? 1907 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1908 } 1909 } 1910 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1911 ATRACE_END(); 1912 if (framesWritten > 0) { 1913 bytesWritten = framesWritten << mBitShift; 1914 } else { 1915 bytesWritten = framesWritten; 1916 } 1917 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1918 if (status == NO_ERROR) { 1919 size_t totalFramesWritten = mNormalSink->framesWritten(); 1920 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1921 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1922 mLatchDValid = true; 1923 } 1924 } 1925 // otherwise use the HAL / AudioStreamOut directly 1926 } else { 1927 // Direct output and offload threads 1928 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1929 if (mUseAsyncWrite) { 1930 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1931 mWriteAckSequence += 2; 1932 mWriteAckSequence |= 1; 1933 ALOG_ASSERT(mCallbackThread != 0); 1934 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1935 } 1936 // FIXME We should have an implementation of timestamps for direct output threads. 1937 // They are used e.g for multichannel PCM playback over HDMI. 1938 bytesWritten = mOutput->stream->write(mOutput->stream, 1939 (char *)mMixBuffer + offset, mBytesRemaining); 1940 if (mUseAsyncWrite && 1941 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1942 // do not wait for async callback in case of error of full write 1943 mWriteAckSequence &= ~1; 1944 ALOG_ASSERT(mCallbackThread != 0); 1945 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1946 } 1947 } 1948 1949 mNumWrites++; 1950 mInWrite = false; 1951 mStandby = false; 1952 return bytesWritten; 1953} 1954 1955void AudioFlinger::PlaybackThread::threadLoop_drain() 1956{ 1957 if (mOutput->stream->drain) { 1958 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1959 if (mUseAsyncWrite) { 1960 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1961 mDrainSequence |= 1; 1962 ALOG_ASSERT(mCallbackThread != 0); 1963 mCallbackThread->setDraining(mDrainSequence); 1964 } 1965 mOutput->stream->drain(mOutput->stream, 1966 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1967 : AUDIO_DRAIN_ALL); 1968 } 1969} 1970 1971void AudioFlinger::PlaybackThread::threadLoop_exit() 1972{ 1973 // Default implementation has nothing to do 1974} 1975 1976/* 1977The derived values that are cached: 1978 - mixBufferSize from frame count * frame size 1979 - activeSleepTime from activeSleepTimeUs() 1980 - idleSleepTime from idleSleepTimeUs() 1981 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1982 - maxPeriod from frame count and sample rate (MIXER only) 1983 1984The parameters that affect these derived values are: 1985 - frame count 1986 - frame size 1987 - sample rate 1988 - device type: A2DP or not 1989 - device latency 1990 - format: PCM or not 1991 - active sleep time 1992 - idle sleep time 1993*/ 1994 1995void AudioFlinger::PlaybackThread::cacheParameters_l() 1996{ 1997 mixBufferSize = mNormalFrameCount * mFrameSize; 1998 activeSleepTime = activeSleepTimeUs(); 1999 idleSleepTime = idleSleepTimeUs(); 2000} 2001 2002void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2003{ 2004 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2005 this, streamType, mTracks.size()); 2006 Mutex::Autolock _l(mLock); 2007 2008 size_t size = mTracks.size(); 2009 for (size_t i = 0; i < size; i++) { 2010 sp<Track> t = mTracks[i]; 2011 if (t->streamType() == streamType) { 2012 t->invalidate(); 2013 } 2014 } 2015} 2016 2017status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2018{ 2019 int session = chain->sessionId(); 2020 int16_t *buffer = mMixBuffer; 2021 bool ownsBuffer = false; 2022 2023 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2024 if (session > 0) { 2025 // Only one effect chain can be present in direct output thread and it uses 2026 // the mix buffer as input 2027 if (mType != DIRECT) { 2028 size_t numSamples = mNormalFrameCount * mChannelCount; 2029 buffer = new int16_t[numSamples]; 2030 memset(buffer, 0, numSamples * sizeof(int16_t)); 2031 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2032 ownsBuffer = true; 2033 } 2034 2035 // Attach all tracks with same session ID to this chain. 2036 for (size_t i = 0; i < mTracks.size(); ++i) { 2037 sp<Track> track = mTracks[i]; 2038 if (session == track->sessionId()) { 2039 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2040 buffer); 2041 track->setMainBuffer(buffer); 2042 chain->incTrackCnt(); 2043 } 2044 } 2045 2046 // indicate all active tracks in the chain 2047 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2048 sp<Track> track = mActiveTracks[i].promote(); 2049 if (track == 0) { 2050 continue; 2051 } 2052 if (session == track->sessionId()) { 2053 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2054 chain->incActiveTrackCnt(); 2055 } 2056 } 2057 } 2058 2059 chain->setInBuffer(buffer, ownsBuffer); 2060 chain->setOutBuffer(mMixBuffer); 2061 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2062 // chains list in order to be processed last as it contains output stage effects 2063 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2064 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2065 // after track specific effects and before output stage 2066 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2067 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2068 // Effect chain for other sessions are inserted at beginning of effect 2069 // chains list to be processed before output mix effects. Relative order between other 2070 // sessions is not important 2071 size_t size = mEffectChains.size(); 2072 size_t i = 0; 2073 for (i = 0; i < size; i++) { 2074 if (mEffectChains[i]->sessionId() < session) { 2075 break; 2076 } 2077 } 2078 mEffectChains.insertAt(chain, i); 2079 checkSuspendOnAddEffectChain_l(chain); 2080 2081 return NO_ERROR; 2082} 2083 2084size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2085{ 2086 int session = chain->sessionId(); 2087 2088 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2089 2090 for (size_t i = 0; i < mEffectChains.size(); i++) { 2091 if (chain == mEffectChains[i]) { 2092 mEffectChains.removeAt(i); 2093 // detach all active tracks from the chain 2094 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2095 sp<Track> track = mActiveTracks[i].promote(); 2096 if (track == 0) { 2097 continue; 2098 } 2099 if (session == track->sessionId()) { 2100 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2101 chain.get(), session); 2102 chain->decActiveTrackCnt(); 2103 } 2104 } 2105 2106 // detach all tracks with same session ID from this chain 2107 for (size_t i = 0; i < mTracks.size(); ++i) { 2108 sp<Track> track = mTracks[i]; 2109 if (session == track->sessionId()) { 2110 track->setMainBuffer(mMixBuffer); 2111 chain->decTrackCnt(); 2112 } 2113 } 2114 break; 2115 } 2116 } 2117 return mEffectChains.size(); 2118} 2119 2120status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2121 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2122{ 2123 Mutex::Autolock _l(mLock); 2124 return attachAuxEffect_l(track, EffectId); 2125} 2126 2127status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2128 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2129{ 2130 status_t status = NO_ERROR; 2131 2132 if (EffectId == 0) { 2133 track->setAuxBuffer(0, NULL); 2134 } else { 2135 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2136 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2137 if (effect != 0) { 2138 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2139 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2140 } else { 2141 status = INVALID_OPERATION; 2142 } 2143 } else { 2144 status = BAD_VALUE; 2145 } 2146 } 2147 return status; 2148} 2149 2150void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2151{ 2152 for (size_t i = 0; i < mTracks.size(); ++i) { 2153 sp<Track> track = mTracks[i]; 2154 if (track->auxEffectId() == effectId) { 2155 attachAuxEffect_l(track, 0); 2156 } 2157 } 2158} 2159 2160bool AudioFlinger::PlaybackThread::threadLoop() 2161{ 2162 Vector< sp<Track> > tracksToRemove; 2163 2164 standbyTime = systemTime(); 2165 2166 // MIXER 2167 nsecs_t lastWarning = 0; 2168 2169 // DUPLICATING 2170 // FIXME could this be made local to while loop? 2171 writeFrames = 0; 2172 2173 int lastGeneration = 0; 2174 2175 cacheParameters_l(); 2176 sleepTime = idleSleepTime; 2177 2178 if (mType == MIXER) { 2179 sleepTimeShift = 0; 2180 } 2181 2182 CpuStats cpuStats; 2183 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2184 2185 acquireWakeLock(); 2186 2187 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2188 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2189 // and then that string will be logged at the next convenient opportunity. 2190 const char *logString = NULL; 2191 2192 checkSilentMode_l(); 2193 2194 while (!exitPending()) 2195 { 2196 cpuStats.sample(myName); 2197 2198 Vector< sp<EffectChain> > effectChains; 2199 2200 processConfigEvents(); 2201 2202 { // scope for mLock 2203 2204 Mutex::Autolock _l(mLock); 2205 2206 if (logString != NULL) { 2207 mNBLogWriter->logTimestamp(); 2208 mNBLogWriter->log(logString); 2209 logString = NULL; 2210 } 2211 2212 if (mLatchDValid) { 2213 mLatchQ = mLatchD; 2214 mLatchDValid = false; 2215 mLatchQValid = true; 2216 } 2217 2218 if (checkForNewParameters_l()) { 2219 cacheParameters_l(); 2220 } 2221 2222 saveOutputTracks(); 2223 if (mSignalPending) { 2224 // A signal was raised while we were unlocked 2225 mSignalPending = false; 2226 } else if (waitingAsyncCallback_l()) { 2227 if (exitPending()) { 2228 break; 2229 } 2230 releaseWakeLock_l(); 2231 mWakeLockUids.clear(); 2232 mActiveTracksGeneration++; 2233 ALOGV("wait async completion"); 2234 mWaitWorkCV.wait(mLock); 2235 ALOGV("async completion/wake"); 2236 acquireWakeLock_l(); 2237 standbyTime = systemTime() + standbyDelay; 2238 sleepTime = 0; 2239 2240 continue; 2241 } 2242 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2243 isSuspended()) { 2244 // put audio hardware into standby after short delay 2245 if (shouldStandby_l()) { 2246 2247 threadLoop_standby(); 2248 2249 mStandby = true; 2250 } 2251 2252 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2253 // we're about to wait, flush the binder command buffer 2254 IPCThreadState::self()->flushCommands(); 2255 2256 clearOutputTracks(); 2257 2258 if (exitPending()) { 2259 break; 2260 } 2261 2262 releaseWakeLock_l(); 2263 mWakeLockUids.clear(); 2264 mActiveTracksGeneration++; 2265 // wait until we have something to do... 2266 ALOGV("%s going to sleep", myName.string()); 2267 mWaitWorkCV.wait(mLock); 2268 ALOGV("%s waking up", myName.string()); 2269 acquireWakeLock_l(); 2270 2271 mMixerStatus = MIXER_IDLE; 2272 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2273 mBytesWritten = 0; 2274 mBytesRemaining = 0; 2275 checkSilentMode_l(); 2276 2277 standbyTime = systemTime() + standbyDelay; 2278 sleepTime = idleSleepTime; 2279 if (mType == MIXER) { 2280 sleepTimeShift = 0; 2281 } 2282 2283 continue; 2284 } 2285 } 2286 // mMixerStatusIgnoringFastTracks is also updated internally 2287 mMixerStatus = prepareTracks_l(&tracksToRemove); 2288 2289 // compare with previously applied list 2290 if (lastGeneration != mActiveTracksGeneration) { 2291 // update wakelock 2292 updateWakeLockUids_l(mWakeLockUids); 2293 lastGeneration = mActiveTracksGeneration; 2294 } 2295 2296 // prevent any changes in effect chain list and in each effect chain 2297 // during mixing and effect process as the audio buffers could be deleted 2298 // or modified if an effect is created or deleted 2299 lockEffectChains_l(effectChains); 2300 } // mLock scope ends 2301 2302 if (mBytesRemaining == 0) { 2303 mCurrentWriteLength = 0; 2304 if (mMixerStatus == MIXER_TRACKS_READY) { 2305 // threadLoop_mix() sets mCurrentWriteLength 2306 threadLoop_mix(); 2307 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2308 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2309 // threadLoop_sleepTime sets sleepTime to 0 if data 2310 // must be written to HAL 2311 threadLoop_sleepTime(); 2312 if (sleepTime == 0) { 2313 mCurrentWriteLength = mixBufferSize; 2314 } 2315 } 2316 mBytesRemaining = mCurrentWriteLength; 2317 if (isSuspended()) { 2318 sleepTime = suspendSleepTimeUs(); 2319 // simulate write to HAL when suspended 2320 mBytesWritten += mixBufferSize; 2321 mBytesRemaining = 0; 2322 } 2323 2324 // only process effects if we're going to write 2325 if (sleepTime == 0 && mType != OFFLOAD) { 2326 for (size_t i = 0; i < effectChains.size(); i ++) { 2327 effectChains[i]->process_l(); 2328 } 2329 } 2330 } 2331 // Process effect chains for offloaded thread even if no audio 2332 // was read from audio track: process only updates effect state 2333 // and thus does have to be synchronized with audio writes but may have 2334 // to be called while waiting for async write callback 2335 if (mType == OFFLOAD) { 2336 for (size_t i = 0; i < effectChains.size(); i ++) { 2337 effectChains[i]->process_l(); 2338 } 2339 } 2340 2341 // enable changes in effect chain 2342 unlockEffectChains(effectChains); 2343 2344 if (!waitingAsyncCallback()) { 2345 // sleepTime == 0 means we must write to audio hardware 2346 if (sleepTime == 0) { 2347 if (mBytesRemaining) { 2348 ssize_t ret = threadLoop_write(); 2349 if (ret < 0) { 2350 mBytesRemaining = 0; 2351 } else { 2352 mBytesWritten += ret; 2353 mBytesRemaining -= ret; 2354 } 2355 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2356 (mMixerStatus == MIXER_DRAIN_ALL)) { 2357 threadLoop_drain(); 2358 } 2359if (mType == MIXER) { 2360 // write blocked detection 2361 nsecs_t now = systemTime(); 2362 nsecs_t delta = now - mLastWriteTime; 2363 if (!mStandby && delta > maxPeriod) { 2364 mNumDelayedWrites++; 2365 if ((now - lastWarning) > kWarningThrottleNs) { 2366 ATRACE_NAME("underrun"); 2367 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2368 ns2ms(delta), mNumDelayedWrites, this); 2369 lastWarning = now; 2370 } 2371 } 2372} 2373 2374 } else { 2375 usleep(sleepTime); 2376 } 2377 } 2378 2379 // Finally let go of removed track(s), without the lock held 2380 // since we can't guarantee the destructors won't acquire that 2381 // same lock. This will also mutate and push a new fast mixer state. 2382 threadLoop_removeTracks(tracksToRemove); 2383 tracksToRemove.clear(); 2384 2385 // FIXME I don't understand the need for this here; 2386 // it was in the original code but maybe the 2387 // assignment in saveOutputTracks() makes this unnecessary? 2388 clearOutputTracks(); 2389 2390 // Effect chains will be actually deleted here if they were removed from 2391 // mEffectChains list during mixing or effects processing 2392 effectChains.clear(); 2393 2394 // FIXME Note that the above .clear() is no longer necessary since effectChains 2395 // is now local to this block, but will keep it for now (at least until merge done). 2396 } 2397 2398 threadLoop_exit(); 2399 2400 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2401 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2402 // put output stream into standby mode 2403 if (!mStandby) { 2404 mOutput->stream->common.standby(&mOutput->stream->common); 2405 } 2406 } 2407 2408 releaseWakeLock(); 2409 mWakeLockUids.clear(); 2410 mActiveTracksGeneration++; 2411 2412 ALOGV("Thread %p type %d exiting", this, mType); 2413 return false; 2414} 2415 2416// removeTracks_l() must be called with ThreadBase::mLock held 2417void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2418{ 2419 size_t count = tracksToRemove.size(); 2420 if (count > 0) { 2421 for (size_t i=0 ; i<count ; i++) { 2422 const sp<Track>& track = tracksToRemove.itemAt(i); 2423 mActiveTracks.remove(track); 2424 mWakeLockUids.remove(track->uid()); 2425 mActiveTracksGeneration++; 2426 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2427 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2428 if (chain != 0) { 2429 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2430 track->sessionId()); 2431 chain->decActiveTrackCnt(); 2432 } 2433 if (track->isTerminated()) { 2434 removeTrack_l(track); 2435 } 2436 } 2437 } 2438 2439} 2440 2441status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2442{ 2443 if (mNormalSink != 0) { 2444 return mNormalSink->getTimestamp(timestamp); 2445 } 2446 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2447 uint64_t position64; 2448 int ret = mOutput->stream->get_presentation_position( 2449 mOutput->stream, &position64, ×tamp.mTime); 2450 if (ret == 0) { 2451 timestamp.mPosition = (uint32_t)position64; 2452 return NO_ERROR; 2453 } 2454 } 2455 return INVALID_OPERATION; 2456} 2457// ---------------------------------------------------------------------------- 2458 2459AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2460 audio_io_handle_t id, audio_devices_t device, type_t type) 2461 : PlaybackThread(audioFlinger, output, id, device, type), 2462 // mAudioMixer below 2463 // mFastMixer below 2464 mFastMixerFutex(0) 2465 // mOutputSink below 2466 // mPipeSink below 2467 // mNormalSink below 2468{ 2469 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2470 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2471 "mFrameCount=%d, mNormalFrameCount=%d", 2472 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2473 mNormalFrameCount); 2474 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2475 2476 // FIXME - Current mixer implementation only supports stereo output 2477 if (mChannelCount != FCC_2) { 2478 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2479 } 2480 2481 // create an NBAIO sink for the HAL output stream, and negotiate 2482 mOutputSink = new AudioStreamOutSink(output->stream); 2483 size_t numCounterOffers = 0; 2484 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2485 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2486 ALOG_ASSERT(index == 0); 2487 2488 // initialize fast mixer depending on configuration 2489 bool initFastMixer; 2490 switch (kUseFastMixer) { 2491 case FastMixer_Never: 2492 initFastMixer = false; 2493 break; 2494 case FastMixer_Always: 2495 initFastMixer = true; 2496 break; 2497 case FastMixer_Static: 2498 case FastMixer_Dynamic: 2499 initFastMixer = mFrameCount < mNormalFrameCount; 2500 break; 2501 } 2502 if (initFastMixer) { 2503 2504 // create a MonoPipe to connect our submix to FastMixer 2505 NBAIO_Format format = mOutputSink->format(); 2506 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2507 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2508 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2509 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2510 const NBAIO_Format offers[1] = {format}; 2511 size_t numCounterOffers = 0; 2512 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2513 ALOG_ASSERT(index == 0); 2514 monoPipe->setAvgFrames((mScreenState & 1) ? 2515 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2516 mPipeSink = monoPipe; 2517 2518#ifdef TEE_SINK 2519 if (mTeeSinkOutputEnabled) { 2520 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2521 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2522 numCounterOffers = 0; 2523 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2524 ALOG_ASSERT(index == 0); 2525 mTeeSink = teeSink; 2526 PipeReader *teeSource = new PipeReader(*teeSink); 2527 numCounterOffers = 0; 2528 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2529 ALOG_ASSERT(index == 0); 2530 mTeeSource = teeSource; 2531 } 2532#endif 2533 2534 // create fast mixer and configure it initially with just one fast track for our submix 2535 mFastMixer = new FastMixer(); 2536 FastMixerStateQueue *sq = mFastMixer->sq(); 2537#ifdef STATE_QUEUE_DUMP 2538 sq->setObserverDump(&mStateQueueObserverDump); 2539 sq->setMutatorDump(&mStateQueueMutatorDump); 2540#endif 2541 FastMixerState *state = sq->begin(); 2542 FastTrack *fastTrack = &state->mFastTracks[0]; 2543 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2544 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2545 fastTrack->mVolumeProvider = NULL; 2546 fastTrack->mGeneration++; 2547 state->mFastTracksGen++; 2548 state->mTrackMask = 1; 2549 // fast mixer will use the HAL output sink 2550 state->mOutputSink = mOutputSink.get(); 2551 state->mOutputSinkGen++; 2552 state->mFrameCount = mFrameCount; 2553 state->mCommand = FastMixerState::COLD_IDLE; 2554 // already done in constructor initialization list 2555 //mFastMixerFutex = 0; 2556 state->mColdFutexAddr = &mFastMixerFutex; 2557 state->mColdGen++; 2558 state->mDumpState = &mFastMixerDumpState; 2559#ifdef TEE_SINK 2560 state->mTeeSink = mTeeSink.get(); 2561#endif 2562 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2563 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2564 sq->end(); 2565 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2566 2567 // start the fast mixer 2568 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2569 pid_t tid = mFastMixer->getTid(); 2570 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2571 if (err != 0) { 2572 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2573 kPriorityFastMixer, getpid_cached, tid, err); 2574 } 2575 2576#ifdef AUDIO_WATCHDOG 2577 // create and start the watchdog 2578 mAudioWatchdog = new AudioWatchdog(); 2579 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2580 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2581 tid = mAudioWatchdog->getTid(); 2582 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2583 if (err != 0) { 2584 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2585 kPriorityFastMixer, getpid_cached, tid, err); 2586 } 2587#endif 2588 2589 } else { 2590 mFastMixer = NULL; 2591 } 2592 2593 switch (kUseFastMixer) { 2594 case FastMixer_Never: 2595 case FastMixer_Dynamic: 2596 mNormalSink = mOutputSink; 2597 break; 2598 case FastMixer_Always: 2599 mNormalSink = mPipeSink; 2600 break; 2601 case FastMixer_Static: 2602 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2603 break; 2604 } 2605} 2606 2607AudioFlinger::MixerThread::~MixerThread() 2608{ 2609 if (mFastMixer != NULL) { 2610 FastMixerStateQueue *sq = mFastMixer->sq(); 2611 FastMixerState *state = sq->begin(); 2612 if (state->mCommand == FastMixerState::COLD_IDLE) { 2613 int32_t old = android_atomic_inc(&mFastMixerFutex); 2614 if (old == -1) { 2615 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2616 } 2617 } 2618 state->mCommand = FastMixerState::EXIT; 2619 sq->end(); 2620 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2621 mFastMixer->join(); 2622 // Though the fast mixer thread has exited, it's state queue is still valid. 2623 // We'll use that extract the final state which contains one remaining fast track 2624 // corresponding to our sub-mix. 2625 state = sq->begin(); 2626 ALOG_ASSERT(state->mTrackMask == 1); 2627 FastTrack *fastTrack = &state->mFastTracks[0]; 2628 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2629 delete fastTrack->mBufferProvider; 2630 sq->end(false /*didModify*/); 2631 delete mFastMixer; 2632#ifdef AUDIO_WATCHDOG 2633 if (mAudioWatchdog != 0) { 2634 mAudioWatchdog->requestExit(); 2635 mAudioWatchdog->requestExitAndWait(); 2636 mAudioWatchdog.clear(); 2637 } 2638#endif 2639 } 2640 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2641 delete mAudioMixer; 2642} 2643 2644 2645uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2646{ 2647 if (mFastMixer != NULL) { 2648 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2649 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2650 } 2651 return latency; 2652} 2653 2654 2655void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2656{ 2657 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2658} 2659 2660ssize_t AudioFlinger::MixerThread::threadLoop_write() 2661{ 2662 // FIXME we should only do one push per cycle; confirm this is true 2663 // Start the fast mixer if it's not already running 2664 if (mFastMixer != NULL) { 2665 FastMixerStateQueue *sq = mFastMixer->sq(); 2666 FastMixerState *state = sq->begin(); 2667 if (state->mCommand != FastMixerState::MIX_WRITE && 2668 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2669 if (state->mCommand == FastMixerState::COLD_IDLE) { 2670 int32_t old = android_atomic_inc(&mFastMixerFutex); 2671 if (old == -1) { 2672 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2673 } 2674#ifdef AUDIO_WATCHDOG 2675 if (mAudioWatchdog != 0) { 2676 mAudioWatchdog->resume(); 2677 } 2678#endif 2679 } 2680 state->mCommand = FastMixerState::MIX_WRITE; 2681 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2682 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2683 sq->end(); 2684 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2685 if (kUseFastMixer == FastMixer_Dynamic) { 2686 mNormalSink = mPipeSink; 2687 } 2688 } else { 2689 sq->end(false /*didModify*/); 2690 } 2691 } 2692 return PlaybackThread::threadLoop_write(); 2693} 2694 2695void AudioFlinger::MixerThread::threadLoop_standby() 2696{ 2697 // Idle the fast mixer if it's currently running 2698 if (mFastMixer != NULL) { 2699 FastMixerStateQueue *sq = mFastMixer->sq(); 2700 FastMixerState *state = sq->begin(); 2701 if (!(state->mCommand & FastMixerState::IDLE)) { 2702 state->mCommand = FastMixerState::COLD_IDLE; 2703 state->mColdFutexAddr = &mFastMixerFutex; 2704 state->mColdGen++; 2705 mFastMixerFutex = 0; 2706 sq->end(); 2707 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2708 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2709 if (kUseFastMixer == FastMixer_Dynamic) { 2710 mNormalSink = mOutputSink; 2711 } 2712#ifdef AUDIO_WATCHDOG 2713 if (mAudioWatchdog != 0) { 2714 mAudioWatchdog->pause(); 2715 } 2716#endif 2717 } else { 2718 sq->end(false /*didModify*/); 2719 } 2720 } 2721 PlaybackThread::threadLoop_standby(); 2722} 2723 2724// Empty implementation for standard mixer 2725// Overridden for offloaded playback 2726void AudioFlinger::PlaybackThread::flushOutput_l() 2727{ 2728} 2729 2730bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2731{ 2732 return false; 2733} 2734 2735bool AudioFlinger::PlaybackThread::shouldStandby_l() 2736{ 2737 return !mStandby; 2738} 2739 2740bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2741{ 2742 Mutex::Autolock _l(mLock); 2743 return waitingAsyncCallback_l(); 2744} 2745 2746// shared by MIXER and DIRECT, overridden by DUPLICATING 2747void AudioFlinger::PlaybackThread::threadLoop_standby() 2748{ 2749 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2750 mOutput->stream->common.standby(&mOutput->stream->common); 2751 if (mUseAsyncWrite != 0) { 2752 // discard any pending drain or write ack by incrementing sequence 2753 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2754 mDrainSequence = (mDrainSequence + 2) & ~1; 2755 ALOG_ASSERT(mCallbackThread != 0); 2756 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2757 mCallbackThread->setDraining(mDrainSequence); 2758 } 2759} 2760 2761void AudioFlinger::MixerThread::threadLoop_mix() 2762{ 2763 // obtain the presentation timestamp of the next output buffer 2764 int64_t pts; 2765 status_t status = INVALID_OPERATION; 2766 2767 if (mNormalSink != 0) { 2768 status = mNormalSink->getNextWriteTimestamp(&pts); 2769 } else { 2770 status = mOutputSink->getNextWriteTimestamp(&pts); 2771 } 2772 2773 if (status != NO_ERROR) { 2774 pts = AudioBufferProvider::kInvalidPTS; 2775 } 2776 2777 // mix buffers... 2778 mAudioMixer->process(pts); 2779 mCurrentWriteLength = mixBufferSize; 2780 // increase sleep time progressively when application underrun condition clears. 2781 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2782 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2783 // such that we would underrun the audio HAL. 2784 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2785 sleepTimeShift--; 2786 } 2787 sleepTime = 0; 2788 standbyTime = systemTime() + standbyDelay; 2789 //TODO: delay standby when effects have a tail 2790} 2791 2792void AudioFlinger::MixerThread::threadLoop_sleepTime() 2793{ 2794 // If no tracks are ready, sleep once for the duration of an output 2795 // buffer size, then write 0s to the output 2796 if (sleepTime == 0) { 2797 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2798 sleepTime = activeSleepTime >> sleepTimeShift; 2799 if (sleepTime < kMinThreadSleepTimeUs) { 2800 sleepTime = kMinThreadSleepTimeUs; 2801 } 2802 // reduce sleep time in case of consecutive application underruns to avoid 2803 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2804 // duration we would end up writing less data than needed by the audio HAL if 2805 // the condition persists. 2806 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2807 sleepTimeShift++; 2808 } 2809 } else { 2810 sleepTime = idleSleepTime; 2811 } 2812 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2813 memset(mMixBuffer, 0, mixBufferSize); 2814 sleepTime = 0; 2815 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2816 "anticipated start"); 2817 } 2818 // TODO add standby time extension fct of effect tail 2819} 2820 2821// prepareTracks_l() must be called with ThreadBase::mLock held 2822AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2823 Vector< sp<Track> > *tracksToRemove) 2824{ 2825 2826 mixer_state mixerStatus = MIXER_IDLE; 2827 // find out which tracks need to be processed 2828 size_t count = mActiveTracks.size(); 2829 size_t mixedTracks = 0; 2830 size_t tracksWithEffect = 0; 2831 // counts only _active_ fast tracks 2832 size_t fastTracks = 0; 2833 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2834 2835 float masterVolume = mMasterVolume; 2836 bool masterMute = mMasterMute; 2837 2838 if (masterMute) { 2839 masterVolume = 0; 2840 } 2841 // Delegate master volume control to effect in output mix effect chain if needed 2842 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2843 if (chain != 0) { 2844 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2845 chain->setVolume_l(&v, &v); 2846 masterVolume = (float)((v + (1 << 23)) >> 24); 2847 chain.clear(); 2848 } 2849 2850 // prepare a new state to push 2851 FastMixerStateQueue *sq = NULL; 2852 FastMixerState *state = NULL; 2853 bool didModify = false; 2854 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2855 if (mFastMixer != NULL) { 2856 sq = mFastMixer->sq(); 2857 state = sq->begin(); 2858 } 2859 2860 for (size_t i=0 ; i<count ; i++) { 2861 const sp<Track> t = mActiveTracks[i].promote(); 2862 if (t == 0) { 2863 continue; 2864 } 2865 2866 // this const just means the local variable doesn't change 2867 Track* const track = t.get(); 2868 2869 // process fast tracks 2870 if (track->isFastTrack()) { 2871 2872 // It's theoretically possible (though unlikely) for a fast track to be created 2873 // and then removed within the same normal mix cycle. This is not a problem, as 2874 // the track never becomes active so it's fast mixer slot is never touched. 2875 // The converse, of removing an (active) track and then creating a new track 2876 // at the identical fast mixer slot within the same normal mix cycle, 2877 // is impossible because the slot isn't marked available until the end of each cycle. 2878 int j = track->mFastIndex; 2879 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2880 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2881 FastTrack *fastTrack = &state->mFastTracks[j]; 2882 2883 // Determine whether the track is currently in underrun condition, 2884 // and whether it had a recent underrun. 2885 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2886 FastTrackUnderruns underruns = ftDump->mUnderruns; 2887 uint32_t recentFull = (underruns.mBitFields.mFull - 2888 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2889 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2890 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2891 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2892 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2893 uint32_t recentUnderruns = recentPartial + recentEmpty; 2894 track->mObservedUnderruns = underruns; 2895 // don't count underruns that occur while stopping or pausing 2896 // or stopped which can occur when flush() is called while active 2897 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2898 recentUnderruns > 0) { 2899 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2900 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2901 } 2902 2903 // This is similar to the state machine for normal tracks, 2904 // with a few modifications for fast tracks. 2905 bool isActive = true; 2906 switch (track->mState) { 2907 case TrackBase::STOPPING_1: 2908 // track stays active in STOPPING_1 state until first underrun 2909 if (recentUnderruns > 0 || track->isTerminated()) { 2910 track->mState = TrackBase::STOPPING_2; 2911 } 2912 break; 2913 case TrackBase::PAUSING: 2914 // ramp down is not yet implemented 2915 track->setPaused(); 2916 break; 2917 case TrackBase::RESUMING: 2918 // ramp up is not yet implemented 2919 track->mState = TrackBase::ACTIVE; 2920 break; 2921 case TrackBase::ACTIVE: 2922 if (recentFull > 0 || recentPartial > 0) { 2923 // track has provided at least some frames recently: reset retry count 2924 track->mRetryCount = kMaxTrackRetries; 2925 } 2926 if (recentUnderruns == 0) { 2927 // no recent underruns: stay active 2928 break; 2929 } 2930 // there has recently been an underrun of some kind 2931 if (track->sharedBuffer() == 0) { 2932 // were any of the recent underruns "empty" (no frames available)? 2933 if (recentEmpty == 0) { 2934 // no, then ignore the partial underruns as they are allowed indefinitely 2935 break; 2936 } 2937 // there has recently been an "empty" underrun: decrement the retry counter 2938 if (--(track->mRetryCount) > 0) { 2939 break; 2940 } 2941 // indicate to client process that the track was disabled because of underrun; 2942 // it will then automatically call start() when data is available 2943 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2944 // remove from active list, but state remains ACTIVE [confusing but true] 2945 isActive = false; 2946 break; 2947 } 2948 // fall through 2949 case TrackBase::STOPPING_2: 2950 case TrackBase::PAUSED: 2951 case TrackBase::STOPPED: 2952 case TrackBase::FLUSHED: // flush() while active 2953 // Check for presentation complete if track is inactive 2954 // We have consumed all the buffers of this track. 2955 // This would be incomplete if we auto-paused on underrun 2956 { 2957 size_t audioHALFrames = 2958 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2959 size_t framesWritten = mBytesWritten / mFrameSize; 2960 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2961 // track stays in active list until presentation is complete 2962 break; 2963 } 2964 } 2965 if (track->isStopping_2()) { 2966 track->mState = TrackBase::STOPPED; 2967 } 2968 if (track->isStopped()) { 2969 // Can't reset directly, as fast mixer is still polling this track 2970 // track->reset(); 2971 // So instead mark this track as needing to be reset after push with ack 2972 resetMask |= 1 << i; 2973 } 2974 isActive = false; 2975 break; 2976 case TrackBase::IDLE: 2977 default: 2978 LOG_FATAL("unexpected track state %d", track->mState); 2979 } 2980 2981 if (isActive) { 2982 // was it previously inactive? 2983 if (!(state->mTrackMask & (1 << j))) { 2984 ExtendedAudioBufferProvider *eabp = track; 2985 VolumeProvider *vp = track; 2986 fastTrack->mBufferProvider = eabp; 2987 fastTrack->mVolumeProvider = vp; 2988 fastTrack->mSampleRate = track->mSampleRate; 2989 fastTrack->mChannelMask = track->mChannelMask; 2990 fastTrack->mGeneration++; 2991 state->mTrackMask |= 1 << j; 2992 didModify = true; 2993 // no acknowledgement required for newly active tracks 2994 } 2995 // cache the combined master volume and stream type volume for fast mixer; this 2996 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2997 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2998 ++fastTracks; 2999 } else { 3000 // was it previously active? 3001 if (state->mTrackMask & (1 << j)) { 3002 fastTrack->mBufferProvider = NULL; 3003 fastTrack->mGeneration++; 3004 state->mTrackMask &= ~(1 << j); 3005 didModify = true; 3006 // If any fast tracks were removed, we must wait for acknowledgement 3007 // because we're about to decrement the last sp<> on those tracks. 3008 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3009 } else { 3010 LOG_FATAL("fast track %d should have been active", j); 3011 } 3012 tracksToRemove->add(track); 3013 // Avoids a misleading display in dumpsys 3014 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3015 } 3016 continue; 3017 } 3018 3019 { // local variable scope to avoid goto warning 3020 3021 audio_track_cblk_t* cblk = track->cblk(); 3022 3023 // The first time a track is added we wait 3024 // for all its buffers to be filled before processing it 3025 int name = track->name(); 3026 // make sure that we have enough frames to mix one full buffer. 3027 // enforce this condition only once to enable draining the buffer in case the client 3028 // app does not call stop() and relies on underrun to stop: 3029 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3030 // during last round 3031 size_t desiredFrames; 3032 uint32_t sr = track->sampleRate(); 3033 if (sr == mSampleRate) { 3034 desiredFrames = mNormalFrameCount; 3035 } else { 3036 // +1 for rounding and +1 for additional sample needed for interpolation 3037 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3038 // add frames already consumed but not yet released by the resampler 3039 // because mAudioTrackServerProxy->framesReady() will include these frames 3040 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3041 // the minimum track buffer size is normally twice the number of frames necessary 3042 // to fill one buffer and the resampler should not leave more than one buffer worth 3043 // of unreleased frames after each pass, but just in case... 3044 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3045 } 3046 uint32_t minFrames = 1; 3047 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3048 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3049 minFrames = desiredFrames; 3050 } 3051 3052 size_t framesReady = track->framesReady(); 3053 if ((framesReady >= minFrames) && track->isReady() && 3054 !track->isPaused() && !track->isTerminated()) 3055 { 3056 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3057 3058 mixedTracks++; 3059 3060 // track->mainBuffer() != mMixBuffer means there is an effect chain 3061 // connected to the track 3062 chain.clear(); 3063 if (track->mainBuffer() != mMixBuffer) { 3064 chain = getEffectChain_l(track->sessionId()); 3065 // Delegate volume control to effect in track effect chain if needed 3066 if (chain != 0) { 3067 tracksWithEffect++; 3068 } else { 3069 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3070 "session %d", 3071 name, track->sessionId()); 3072 } 3073 } 3074 3075 3076 int param = AudioMixer::VOLUME; 3077 if (track->mFillingUpStatus == Track::FS_FILLED) { 3078 // no ramp for the first volume setting 3079 track->mFillingUpStatus = Track::FS_ACTIVE; 3080 if (track->mState == TrackBase::RESUMING) { 3081 track->mState = TrackBase::ACTIVE; 3082 param = AudioMixer::RAMP_VOLUME; 3083 } 3084 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3085 // FIXME should not make a decision based on mServer 3086 } else if (cblk->mServer != 0) { 3087 // If the track is stopped before the first frame was mixed, 3088 // do not apply ramp 3089 param = AudioMixer::RAMP_VOLUME; 3090 } 3091 3092 // compute volume for this track 3093 uint32_t vl, vr, va; 3094 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3095 vl = vr = va = 0; 3096 if (track->isPausing()) { 3097 track->setPaused(); 3098 } 3099 } else { 3100 3101 // read original volumes with volume control 3102 float typeVolume = mStreamTypes[track->streamType()].volume; 3103 float v = masterVolume * typeVolume; 3104 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3105 uint32_t vlr = proxy->getVolumeLR(); 3106 vl = vlr & 0xFFFF; 3107 vr = vlr >> 16; 3108 // track volumes come from shared memory, so can't be trusted and must be clamped 3109 if (vl > MAX_GAIN_INT) { 3110 ALOGV("Track left volume out of range: %04X", vl); 3111 vl = MAX_GAIN_INT; 3112 } 3113 if (vr > MAX_GAIN_INT) { 3114 ALOGV("Track right volume out of range: %04X", vr); 3115 vr = MAX_GAIN_INT; 3116 } 3117 // now apply the master volume and stream type volume 3118 vl = (uint32_t)(v * vl) << 12; 3119 vr = (uint32_t)(v * vr) << 12; 3120 // assuming master volume and stream type volume each go up to 1.0, 3121 // vl and vr are now in 8.24 format 3122 3123 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3124 // send level comes from shared memory and so may be corrupt 3125 if (sendLevel > MAX_GAIN_INT) { 3126 ALOGV("Track send level out of range: %04X", sendLevel); 3127 sendLevel = MAX_GAIN_INT; 3128 } 3129 va = (uint32_t)(v * sendLevel); 3130 } 3131 3132 // Delegate volume control to effect in track effect chain if needed 3133 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3134 // Do not ramp volume if volume is controlled by effect 3135 param = AudioMixer::VOLUME; 3136 track->mHasVolumeController = true; 3137 } else { 3138 // force no volume ramp when volume controller was just disabled or removed 3139 // from effect chain to avoid volume spike 3140 if (track->mHasVolumeController) { 3141 param = AudioMixer::VOLUME; 3142 } 3143 track->mHasVolumeController = false; 3144 } 3145 3146 // Convert volumes from 8.24 to 4.12 format 3147 // This additional clamping is needed in case chain->setVolume_l() overshot 3148 vl = (vl + (1 << 11)) >> 12; 3149 if (vl > MAX_GAIN_INT) { 3150 vl = MAX_GAIN_INT; 3151 } 3152 vr = (vr + (1 << 11)) >> 12; 3153 if (vr > MAX_GAIN_INT) { 3154 vr = MAX_GAIN_INT; 3155 } 3156 3157 if (va > MAX_GAIN_INT) { 3158 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3159 } 3160 3161 // XXX: these things DON'T need to be done each time 3162 mAudioMixer->setBufferProvider(name, track); 3163 mAudioMixer->enable(name); 3164 3165 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3166 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3167 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3168 mAudioMixer->setParameter( 3169 name, 3170 AudioMixer::TRACK, 3171 AudioMixer::FORMAT, (void *)track->format()); 3172 mAudioMixer->setParameter( 3173 name, 3174 AudioMixer::TRACK, 3175 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3176 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3177 uint32_t maxSampleRate = mSampleRate * 2; 3178 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3179 if (reqSampleRate == 0) { 3180 reqSampleRate = mSampleRate; 3181 } else if (reqSampleRate > maxSampleRate) { 3182 reqSampleRate = maxSampleRate; 3183 } 3184 mAudioMixer->setParameter( 3185 name, 3186 AudioMixer::RESAMPLE, 3187 AudioMixer::SAMPLE_RATE, 3188 (void *)reqSampleRate); 3189 mAudioMixer->setParameter( 3190 name, 3191 AudioMixer::TRACK, 3192 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3193 mAudioMixer->setParameter( 3194 name, 3195 AudioMixer::TRACK, 3196 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3197 3198 // reset retry count 3199 track->mRetryCount = kMaxTrackRetries; 3200 3201 // If one track is ready, set the mixer ready if: 3202 // - the mixer was not ready during previous round OR 3203 // - no other track is not ready 3204 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3205 mixerStatus != MIXER_TRACKS_ENABLED) { 3206 mixerStatus = MIXER_TRACKS_READY; 3207 } 3208 } else { 3209 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3210 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3211 } 3212 // clear effect chain input buffer if an active track underruns to avoid sending 3213 // previous audio buffer again to effects 3214 chain = getEffectChain_l(track->sessionId()); 3215 if (chain != 0) { 3216 chain->clearInputBuffer(); 3217 } 3218 3219 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3220 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3221 track->isStopped() || track->isPaused()) { 3222 // We have consumed all the buffers of this track. 3223 // Remove it from the list of active tracks. 3224 // TODO: use actual buffer filling status instead of latency when available from 3225 // audio HAL 3226 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3227 size_t framesWritten = mBytesWritten / mFrameSize; 3228 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3229 if (track->isStopped()) { 3230 track->reset(); 3231 } 3232 tracksToRemove->add(track); 3233 } 3234 } else { 3235 // No buffers for this track. Give it a few chances to 3236 // fill a buffer, then remove it from active list. 3237 if (--(track->mRetryCount) <= 0) { 3238 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3239 tracksToRemove->add(track); 3240 // indicate to client process that the track was disabled because of underrun; 3241 // it will then automatically call start() when data is available 3242 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3243 // If one track is not ready, mark the mixer also not ready if: 3244 // - the mixer was ready during previous round OR 3245 // - no other track is ready 3246 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3247 mixerStatus != MIXER_TRACKS_READY) { 3248 mixerStatus = MIXER_TRACKS_ENABLED; 3249 } 3250 } 3251 mAudioMixer->disable(name); 3252 } 3253 3254 } // local variable scope to avoid goto warning 3255track_is_ready: ; 3256 3257 } 3258 3259 // Push the new FastMixer state if necessary 3260 bool pauseAudioWatchdog = false; 3261 if (didModify) { 3262 state->mFastTracksGen++; 3263 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3264 if (kUseFastMixer == FastMixer_Dynamic && 3265 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3266 state->mCommand = FastMixerState::COLD_IDLE; 3267 state->mColdFutexAddr = &mFastMixerFutex; 3268 state->mColdGen++; 3269 mFastMixerFutex = 0; 3270 if (kUseFastMixer == FastMixer_Dynamic) { 3271 mNormalSink = mOutputSink; 3272 } 3273 // If we go into cold idle, need to wait for acknowledgement 3274 // so that fast mixer stops doing I/O. 3275 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3276 pauseAudioWatchdog = true; 3277 } 3278 } 3279 if (sq != NULL) { 3280 sq->end(didModify); 3281 sq->push(block); 3282 } 3283#ifdef AUDIO_WATCHDOG 3284 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3285 mAudioWatchdog->pause(); 3286 } 3287#endif 3288 3289 // Now perform the deferred reset on fast tracks that have stopped 3290 while (resetMask != 0) { 3291 size_t i = __builtin_ctz(resetMask); 3292 ALOG_ASSERT(i < count); 3293 resetMask &= ~(1 << i); 3294 sp<Track> t = mActiveTracks[i].promote(); 3295 if (t == 0) { 3296 continue; 3297 } 3298 Track* track = t.get(); 3299 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3300 track->reset(); 3301 } 3302 3303 // remove all the tracks that need to be... 3304 removeTracks_l(*tracksToRemove); 3305 3306 // mix buffer must be cleared if all tracks are connected to an 3307 // effect chain as in this case the mixer will not write to 3308 // mix buffer and track effects will accumulate into it 3309 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3310 (mixedTracks == 0 && fastTracks > 0))) { 3311 // FIXME as a performance optimization, should remember previous zero status 3312 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3313 } 3314 3315 // if any fast tracks, then status is ready 3316 mMixerStatusIgnoringFastTracks = mixerStatus; 3317 if (fastTracks > 0) { 3318 mixerStatus = MIXER_TRACKS_READY; 3319 } 3320 return mixerStatus; 3321} 3322 3323// getTrackName_l() must be called with ThreadBase::mLock held 3324int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3325{ 3326 return mAudioMixer->getTrackName(channelMask, sessionId); 3327} 3328 3329// deleteTrackName_l() must be called with ThreadBase::mLock held 3330void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3331{ 3332 ALOGV("remove track (%d) and delete from mixer", name); 3333 mAudioMixer->deleteTrackName(name); 3334} 3335 3336// checkForNewParameters_l() must be called with ThreadBase::mLock held 3337bool AudioFlinger::MixerThread::checkForNewParameters_l() 3338{ 3339 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3340 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3341 bool reconfig = false; 3342 3343 while (!mNewParameters.isEmpty()) { 3344 3345 if (mFastMixer != NULL) { 3346 FastMixerStateQueue *sq = mFastMixer->sq(); 3347 FastMixerState *state = sq->begin(); 3348 if (!(state->mCommand & FastMixerState::IDLE)) { 3349 previousCommand = state->mCommand; 3350 state->mCommand = FastMixerState::HOT_IDLE; 3351 sq->end(); 3352 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3353 } else { 3354 sq->end(false /*didModify*/); 3355 } 3356 } 3357 3358 status_t status = NO_ERROR; 3359 String8 keyValuePair = mNewParameters[0]; 3360 AudioParameter param = AudioParameter(keyValuePair); 3361 int value; 3362 3363 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3364 reconfig = true; 3365 } 3366 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3367 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3368 status = BAD_VALUE; 3369 } else { 3370 // no need to save value, since it's constant 3371 reconfig = true; 3372 } 3373 } 3374 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3375 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3376 status = BAD_VALUE; 3377 } else { 3378 // no need to save value, since it's constant 3379 reconfig = true; 3380 } 3381 } 3382 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3383 // do not accept frame count changes if tracks are open as the track buffer 3384 // size depends on frame count and correct behavior would not be guaranteed 3385 // if frame count is changed after track creation 3386 if (!mTracks.isEmpty()) { 3387 status = INVALID_OPERATION; 3388 } else { 3389 reconfig = true; 3390 } 3391 } 3392 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3393#ifdef ADD_BATTERY_DATA 3394 // when changing the audio output device, call addBatteryData to notify 3395 // the change 3396 if (mOutDevice != value) { 3397 uint32_t params = 0; 3398 // check whether speaker is on 3399 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3400 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3401 } 3402 3403 audio_devices_t deviceWithoutSpeaker 3404 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3405 // check if any other device (except speaker) is on 3406 if (value & deviceWithoutSpeaker ) { 3407 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3408 } 3409 3410 if (params != 0) { 3411 addBatteryData(params); 3412 } 3413 } 3414#endif 3415 3416 // forward device change to effects that have requested to be 3417 // aware of attached audio device. 3418 if (value != AUDIO_DEVICE_NONE) { 3419 mOutDevice = value; 3420 for (size_t i = 0; i < mEffectChains.size(); i++) { 3421 mEffectChains[i]->setDevice_l(mOutDevice); 3422 } 3423 } 3424 } 3425 3426 if (status == NO_ERROR) { 3427 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3428 keyValuePair.string()); 3429 if (!mStandby && status == INVALID_OPERATION) { 3430 mOutput->stream->common.standby(&mOutput->stream->common); 3431 mStandby = true; 3432 mBytesWritten = 0; 3433 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3434 keyValuePair.string()); 3435 } 3436 if (status == NO_ERROR && reconfig) { 3437 readOutputParameters(); 3438 delete mAudioMixer; 3439 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3440 for (size_t i = 0; i < mTracks.size() ; i++) { 3441 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3442 if (name < 0) { 3443 break; 3444 } 3445 mTracks[i]->mName = name; 3446 } 3447 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3448 } 3449 } 3450 3451 mNewParameters.removeAt(0); 3452 3453 mParamStatus = status; 3454 mParamCond.signal(); 3455 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3456 // already timed out waiting for the status and will never signal the condition. 3457 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3458 } 3459 3460 if (!(previousCommand & FastMixerState::IDLE)) { 3461 ALOG_ASSERT(mFastMixer != NULL); 3462 FastMixerStateQueue *sq = mFastMixer->sq(); 3463 FastMixerState *state = sq->begin(); 3464 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3465 state->mCommand = previousCommand; 3466 sq->end(); 3467 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3468 } 3469 3470 return reconfig; 3471} 3472 3473 3474void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3475{ 3476 const size_t SIZE = 256; 3477 char buffer[SIZE]; 3478 String8 result; 3479 3480 PlaybackThread::dumpInternals(fd, args); 3481 3482 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3483 result.append(buffer); 3484 write(fd, result.string(), result.size()); 3485 3486 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3487 const FastMixerDumpState copy(mFastMixerDumpState); 3488 copy.dump(fd); 3489 3490#ifdef STATE_QUEUE_DUMP 3491 // Similar for state queue 3492 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3493 observerCopy.dump(fd); 3494 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3495 mutatorCopy.dump(fd); 3496#endif 3497 3498#ifdef TEE_SINK 3499 // Write the tee output to a .wav file 3500 dumpTee(fd, mTeeSource, mId); 3501#endif 3502 3503#ifdef AUDIO_WATCHDOG 3504 if (mAudioWatchdog != 0) { 3505 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3506 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3507 wdCopy.dump(fd); 3508 } 3509#endif 3510} 3511 3512uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3513{ 3514 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3515} 3516 3517uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3518{ 3519 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3520} 3521 3522void AudioFlinger::MixerThread::cacheParameters_l() 3523{ 3524 PlaybackThread::cacheParameters_l(); 3525 3526 // FIXME: Relaxed timing because of a certain device that can't meet latency 3527 // Should be reduced to 2x after the vendor fixes the driver issue 3528 // increase threshold again due to low power audio mode. The way this warning 3529 // threshold is calculated and its usefulness should be reconsidered anyway. 3530 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3531} 3532 3533// ---------------------------------------------------------------------------- 3534 3535AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3536 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3537 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3538 // mLeftVolFloat, mRightVolFloat 3539{ 3540} 3541 3542AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3543 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3544 ThreadBase::type_t type) 3545 : PlaybackThread(audioFlinger, output, id, device, type) 3546 // mLeftVolFloat, mRightVolFloat 3547{ 3548} 3549 3550AudioFlinger::DirectOutputThread::~DirectOutputThread() 3551{ 3552} 3553 3554void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3555{ 3556 audio_track_cblk_t* cblk = track->cblk(); 3557 float left, right; 3558 3559 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3560 left = right = 0; 3561 } else { 3562 float typeVolume = mStreamTypes[track->streamType()].volume; 3563 float v = mMasterVolume * typeVolume; 3564 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3565 uint32_t vlr = proxy->getVolumeLR(); 3566 float v_clamped = v * (vlr & 0xFFFF); 3567 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3568 left = v_clamped/MAX_GAIN; 3569 v_clamped = v * (vlr >> 16); 3570 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3571 right = v_clamped/MAX_GAIN; 3572 } 3573 3574 if (lastTrack) { 3575 if (left != mLeftVolFloat || right != mRightVolFloat) { 3576 mLeftVolFloat = left; 3577 mRightVolFloat = right; 3578 3579 // Convert volumes from float to 8.24 3580 uint32_t vl = (uint32_t)(left * (1 << 24)); 3581 uint32_t vr = (uint32_t)(right * (1 << 24)); 3582 3583 // Delegate volume control to effect in track effect chain if needed 3584 // only one effect chain can be present on DirectOutputThread, so if 3585 // there is one, the track is connected to it 3586 if (!mEffectChains.isEmpty()) { 3587 mEffectChains[0]->setVolume_l(&vl, &vr); 3588 left = (float)vl / (1 << 24); 3589 right = (float)vr / (1 << 24); 3590 } 3591 if (mOutput->stream->set_volume) { 3592 mOutput->stream->set_volume(mOutput->stream, left, right); 3593 } 3594 } 3595 } 3596} 3597 3598 3599AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3600 Vector< sp<Track> > *tracksToRemove 3601) 3602{ 3603 size_t count = mActiveTracks.size(); 3604 mixer_state mixerStatus = MIXER_IDLE; 3605 3606 // find out which tracks need to be processed 3607 for (size_t i = 0; i < count; i++) { 3608 sp<Track> t = mActiveTracks[i].promote(); 3609 // The track died recently 3610 if (t == 0) { 3611 continue; 3612 } 3613 3614 Track* const track = t.get(); 3615 audio_track_cblk_t* cblk = track->cblk(); 3616 // Only consider last track started for volume and mixer state control. 3617 // In theory an older track could underrun and restart after the new one starts 3618 // but as we only care about the transition phase between two tracks on a 3619 // direct output, it is not a problem to ignore the underrun case. 3620 sp<Track> l = mLatestActiveTrack.promote(); 3621 bool last = l.get() == track; 3622 3623 // The first time a track is added we wait 3624 // for all its buffers to be filled before processing it 3625 uint32_t minFrames; 3626 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3627 minFrames = mNormalFrameCount; 3628 } else { 3629 minFrames = 1; 3630 } 3631 3632 if ((track->framesReady() >= minFrames) && track->isReady() && 3633 !track->isPaused() && !track->isTerminated()) 3634 { 3635 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3636 3637 if (track->mFillingUpStatus == Track::FS_FILLED) { 3638 track->mFillingUpStatus = Track::FS_ACTIVE; 3639 // make sure processVolume_l() will apply new volume even if 0 3640 mLeftVolFloat = mRightVolFloat = -1.0; 3641 if (track->mState == TrackBase::RESUMING) { 3642 track->mState = TrackBase::ACTIVE; 3643 } 3644 } 3645 3646 // compute volume for this track 3647 processVolume_l(track, last); 3648 if (last) { 3649 // reset retry count 3650 track->mRetryCount = kMaxTrackRetriesDirect; 3651 mActiveTrack = t; 3652 mixerStatus = MIXER_TRACKS_READY; 3653 } 3654 } else { 3655 // clear effect chain input buffer if the last active track started underruns 3656 // to avoid sending previous audio buffer again to effects 3657 if (!mEffectChains.isEmpty() && last) { 3658 mEffectChains[0]->clearInputBuffer(); 3659 } 3660 3661 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3662 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3663 track->isStopped() || track->isPaused()) { 3664 // We have consumed all the buffers of this track. 3665 // Remove it from the list of active tracks. 3666 // TODO: implement behavior for compressed audio 3667 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3668 size_t framesWritten = mBytesWritten / mFrameSize; 3669 if (mStandby || !last || 3670 track->presentationComplete(framesWritten, audioHALFrames)) { 3671 if (track->isStopped()) { 3672 track->reset(); 3673 } 3674 tracksToRemove->add(track); 3675 } 3676 } else { 3677 // No buffers for this track. Give it a few chances to 3678 // fill a buffer, then remove it from active list. 3679 // Only consider last track started for mixer state control 3680 if (--(track->mRetryCount) <= 0) { 3681 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3682 tracksToRemove->add(track); 3683 // indicate to client process that the track was disabled because of underrun; 3684 // it will then automatically call start() when data is available 3685 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3686 } else if (last) { 3687 mixerStatus = MIXER_TRACKS_ENABLED; 3688 } 3689 } 3690 } 3691 } 3692 3693 // remove all the tracks that need to be... 3694 removeTracks_l(*tracksToRemove); 3695 3696 return mixerStatus; 3697} 3698 3699void AudioFlinger::DirectOutputThread::threadLoop_mix() 3700{ 3701 size_t frameCount = mFrameCount; 3702 int8_t *curBuf = (int8_t *)mMixBuffer; 3703 // output audio to hardware 3704 while (frameCount) { 3705 AudioBufferProvider::Buffer buffer; 3706 buffer.frameCount = frameCount; 3707 mActiveTrack->getNextBuffer(&buffer); 3708 if (buffer.raw == NULL) { 3709 memset(curBuf, 0, frameCount * mFrameSize); 3710 break; 3711 } 3712 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3713 frameCount -= buffer.frameCount; 3714 curBuf += buffer.frameCount * mFrameSize; 3715 mActiveTrack->releaseBuffer(&buffer); 3716 } 3717 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3718 sleepTime = 0; 3719 standbyTime = systemTime() + standbyDelay; 3720 mActiveTrack.clear(); 3721} 3722 3723void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3724{ 3725 if (sleepTime == 0) { 3726 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3727 sleepTime = activeSleepTime; 3728 } else { 3729 sleepTime = idleSleepTime; 3730 } 3731 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3732 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3733 sleepTime = 0; 3734 } 3735} 3736 3737// getTrackName_l() must be called with ThreadBase::mLock held 3738int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3739 int sessionId) 3740{ 3741 return 0; 3742} 3743 3744// deleteTrackName_l() must be called with ThreadBase::mLock held 3745void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3746{ 3747} 3748 3749// checkForNewParameters_l() must be called with ThreadBase::mLock held 3750bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3751{ 3752 bool reconfig = false; 3753 3754 while (!mNewParameters.isEmpty()) { 3755 status_t status = NO_ERROR; 3756 String8 keyValuePair = mNewParameters[0]; 3757 AudioParameter param = AudioParameter(keyValuePair); 3758 int value; 3759 3760 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3761 // do not accept frame count changes if tracks are open as the track buffer 3762 // size depends on frame count and correct behavior would not be garantied 3763 // if frame count is changed after track creation 3764 if (!mTracks.isEmpty()) { 3765 status = INVALID_OPERATION; 3766 } else { 3767 reconfig = true; 3768 } 3769 } 3770 if (status == NO_ERROR) { 3771 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3772 keyValuePair.string()); 3773 if (!mStandby && status == INVALID_OPERATION) { 3774 mOutput->stream->common.standby(&mOutput->stream->common); 3775 mStandby = true; 3776 mBytesWritten = 0; 3777 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3778 keyValuePair.string()); 3779 } 3780 if (status == NO_ERROR && reconfig) { 3781 readOutputParameters(); 3782 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3783 } 3784 } 3785 3786 mNewParameters.removeAt(0); 3787 3788 mParamStatus = status; 3789 mParamCond.signal(); 3790 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3791 // already timed out waiting for the status and will never signal the condition. 3792 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3793 } 3794 return reconfig; 3795} 3796 3797uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3798{ 3799 uint32_t time; 3800 if (audio_is_linear_pcm(mFormat)) { 3801 time = PlaybackThread::activeSleepTimeUs(); 3802 } else { 3803 time = 10000; 3804 } 3805 return time; 3806} 3807 3808uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3809{ 3810 uint32_t time; 3811 if (audio_is_linear_pcm(mFormat)) { 3812 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3813 } else { 3814 time = 10000; 3815 } 3816 return time; 3817} 3818 3819uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3820{ 3821 uint32_t time; 3822 if (audio_is_linear_pcm(mFormat)) { 3823 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3824 } else { 3825 time = 10000; 3826 } 3827 return time; 3828} 3829 3830void AudioFlinger::DirectOutputThread::cacheParameters_l() 3831{ 3832 PlaybackThread::cacheParameters_l(); 3833 3834 // use shorter standby delay as on normal output to release 3835 // hardware resources as soon as possible 3836 if (audio_is_linear_pcm(mFormat)) { 3837 standbyDelay = microseconds(activeSleepTime*2); 3838 } else { 3839 standbyDelay = kOffloadStandbyDelayNs; 3840 } 3841} 3842 3843// ---------------------------------------------------------------------------- 3844 3845AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3846 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3847 : Thread(false /*canCallJava*/), 3848 mPlaybackThread(playbackThread), 3849 mWriteAckSequence(0), 3850 mDrainSequence(0) 3851{ 3852} 3853 3854AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3855{ 3856} 3857 3858void AudioFlinger::AsyncCallbackThread::onFirstRef() 3859{ 3860 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3861} 3862 3863bool AudioFlinger::AsyncCallbackThread::threadLoop() 3864{ 3865 while (!exitPending()) { 3866 uint32_t writeAckSequence; 3867 uint32_t drainSequence; 3868 3869 { 3870 Mutex::Autolock _l(mLock); 3871 while (!((mWriteAckSequence & 1) || 3872 (mDrainSequence & 1) || 3873 exitPending())) { 3874 mWaitWorkCV.wait(mLock); 3875 } 3876 3877 if (exitPending()) { 3878 break; 3879 } 3880 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3881 mWriteAckSequence, mDrainSequence); 3882 writeAckSequence = mWriteAckSequence; 3883 mWriteAckSequence &= ~1; 3884 drainSequence = mDrainSequence; 3885 mDrainSequence &= ~1; 3886 } 3887 { 3888 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3889 if (playbackThread != 0) { 3890 if (writeAckSequence & 1) { 3891 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3892 } 3893 if (drainSequence & 1) { 3894 playbackThread->resetDraining(drainSequence >> 1); 3895 } 3896 } 3897 } 3898 } 3899 return false; 3900} 3901 3902void AudioFlinger::AsyncCallbackThread::exit() 3903{ 3904 ALOGV("AsyncCallbackThread::exit"); 3905 Mutex::Autolock _l(mLock); 3906 requestExit(); 3907 mWaitWorkCV.broadcast(); 3908} 3909 3910void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3911{ 3912 Mutex::Autolock _l(mLock); 3913 // bit 0 is cleared 3914 mWriteAckSequence = sequence << 1; 3915} 3916 3917void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3918{ 3919 Mutex::Autolock _l(mLock); 3920 // ignore unexpected callbacks 3921 if (mWriteAckSequence & 2) { 3922 mWriteAckSequence |= 1; 3923 mWaitWorkCV.signal(); 3924 } 3925} 3926 3927void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3928{ 3929 Mutex::Autolock _l(mLock); 3930 // bit 0 is cleared 3931 mDrainSequence = sequence << 1; 3932} 3933 3934void AudioFlinger::AsyncCallbackThread::resetDraining() 3935{ 3936 Mutex::Autolock _l(mLock); 3937 // ignore unexpected callbacks 3938 if (mDrainSequence & 2) { 3939 mDrainSequence |= 1; 3940 mWaitWorkCV.signal(); 3941 } 3942} 3943 3944 3945// ---------------------------------------------------------------------------- 3946AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3947 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3948 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3949 mHwPaused(false), 3950 mFlushPending(false), 3951 mPausedBytesRemaining(0) 3952{ 3953 //FIXME: mStandby should be set to true by ThreadBase constructor 3954 mStandby = true; 3955} 3956 3957void AudioFlinger::OffloadThread::threadLoop_exit() 3958{ 3959 if (mFlushPending || mHwPaused) { 3960 // If a flush is pending or track was paused, just discard buffered data 3961 flushHw_l(); 3962 } else { 3963 mMixerStatus = MIXER_DRAIN_ALL; 3964 threadLoop_drain(); 3965 } 3966 mCallbackThread->exit(); 3967 PlaybackThread::threadLoop_exit(); 3968} 3969 3970AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3971 Vector< sp<Track> > *tracksToRemove 3972) 3973{ 3974 size_t count = mActiveTracks.size(); 3975 3976 mixer_state mixerStatus = MIXER_IDLE; 3977 bool doHwPause = false; 3978 bool doHwResume = false; 3979 3980 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3981 3982 // find out which tracks need to be processed 3983 for (size_t i = 0; i < count; i++) { 3984 sp<Track> t = mActiveTracks[i].promote(); 3985 // The track died recently 3986 if (t == 0) { 3987 continue; 3988 } 3989 Track* const track = t.get(); 3990 audio_track_cblk_t* cblk = track->cblk(); 3991 // Only consider last track started for volume and mixer state control. 3992 // In theory an older track could underrun and restart after the new one starts 3993 // but as we only care about the transition phase between two tracks on a 3994 // direct output, it is not a problem to ignore the underrun case. 3995 sp<Track> l = mLatestActiveTrack.promote(); 3996 bool last = l.get() == track; 3997 3998 if (track->isPausing()) { 3999 track->setPaused(); 4000 if (last) { 4001 if (!mHwPaused) { 4002 doHwPause = true; 4003 mHwPaused = true; 4004 } 4005 // If we were part way through writing the mixbuffer to 4006 // the HAL we must save this until we resume 4007 // BUG - this will be wrong if a different track is made active, 4008 // in that case we want to discard the pending data in the 4009 // mixbuffer and tell the client to present it again when the 4010 // track is resumed 4011 mPausedWriteLength = mCurrentWriteLength; 4012 mPausedBytesRemaining = mBytesRemaining; 4013 mBytesRemaining = 0; // stop writing 4014 } 4015 tracksToRemove->add(track); 4016 } else if (track->framesReady() && track->isReady() && 4017 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4018 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4019 if (track->mFillingUpStatus == Track::FS_FILLED) { 4020 track->mFillingUpStatus = Track::FS_ACTIVE; 4021 // make sure processVolume_l() will apply new volume even if 0 4022 mLeftVolFloat = mRightVolFloat = -1.0; 4023 if (track->mState == TrackBase::RESUMING) { 4024 track->mState = TrackBase::ACTIVE; 4025 if (last) { 4026 if (mPausedBytesRemaining) { 4027 // Need to continue write that was interrupted 4028 mCurrentWriteLength = mPausedWriteLength; 4029 mBytesRemaining = mPausedBytesRemaining; 4030 mPausedBytesRemaining = 0; 4031 } 4032 if (mHwPaused) { 4033 doHwResume = true; 4034 mHwPaused = false; 4035 // threadLoop_mix() will handle the case that we need to 4036 // resume an interrupted write 4037 } 4038 // enable write to audio HAL 4039 sleepTime = 0; 4040 } 4041 } 4042 } 4043 4044 if (last) { 4045 sp<Track> previousTrack = mPreviousTrack.promote(); 4046 if (previousTrack != 0) { 4047 if (track != previousTrack.get()) { 4048 // Flush any data still being written from last track 4049 mBytesRemaining = 0; 4050 if (mPausedBytesRemaining) { 4051 // Last track was paused so we also need to flush saved 4052 // mixbuffer state and invalidate track so that it will 4053 // re-submit that unwritten data when it is next resumed 4054 mPausedBytesRemaining = 0; 4055 // Invalidate is a bit drastic - would be more efficient 4056 // to have a flag to tell client that some of the 4057 // previously written data was lost 4058 previousTrack->invalidate(); 4059 } 4060 // flush data already sent to the DSP if changing audio session as audio 4061 // comes from a different source. Also invalidate previous track to force a 4062 // seek when resuming. 4063 if (previousTrack->sessionId() != track->sessionId()) { 4064 previousTrack->invalidate(); 4065 mFlushPending = true; 4066 } 4067 } 4068 } 4069 mPreviousTrack = track; 4070 // reset retry count 4071 track->mRetryCount = kMaxTrackRetriesOffload; 4072 mActiveTrack = t; 4073 mixerStatus = MIXER_TRACKS_READY; 4074 } 4075 } else { 4076 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4077 if (track->isStopping_1()) { 4078 // Hardware buffer can hold a large amount of audio so we must 4079 // wait for all current track's data to drain before we say 4080 // that the track is stopped. 4081 if (mBytesRemaining == 0) { 4082 // Only start draining when all data in mixbuffer 4083 // has been written 4084 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4085 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4086 // do not drain if no data was ever sent to HAL (mStandby == true) 4087 if (last && !mStandby) { 4088 // do not modify drain sequence if we are already draining. This happens 4089 // when resuming from pause after drain. 4090 if ((mDrainSequence & 1) == 0) { 4091 sleepTime = 0; 4092 standbyTime = systemTime() + standbyDelay; 4093 mixerStatus = MIXER_DRAIN_TRACK; 4094 mDrainSequence += 2; 4095 } 4096 if (mHwPaused) { 4097 // It is possible to move from PAUSED to STOPPING_1 without 4098 // a resume so we must ensure hardware is running 4099 doHwResume = true; 4100 mHwPaused = false; 4101 } 4102 } 4103 } 4104 } else if (track->isStopping_2()) { 4105 // Drain has completed or we are in standby, signal presentation complete 4106 if (!(mDrainSequence & 1) || !last || mStandby) { 4107 track->mState = TrackBase::STOPPED; 4108 size_t audioHALFrames = 4109 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4110 size_t framesWritten = 4111 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4112 track->presentationComplete(framesWritten, audioHALFrames); 4113 track->reset(); 4114 tracksToRemove->add(track); 4115 } 4116 } else { 4117 // No buffers for this track. Give it a few chances to 4118 // fill a buffer, then remove it from active list. 4119 if (--(track->mRetryCount) <= 0) { 4120 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4121 track->name()); 4122 tracksToRemove->add(track); 4123 // indicate to client process that the track was disabled because of underrun; 4124 // it will then automatically call start() when data is available 4125 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4126 } else if (last){ 4127 mixerStatus = MIXER_TRACKS_ENABLED; 4128 } 4129 } 4130 } 4131 // compute volume for this track 4132 processVolume_l(track, last); 4133 } 4134 4135 // make sure the pause/flush/resume sequence is executed in the right order. 4136 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4137 // before flush and then resume HW. This can happen in case of pause/flush/resume 4138 // if resume is received before pause is executed. 4139 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4140 mOutput->stream->pause(mOutput->stream); 4141 if (!doHwPause) { 4142 doHwResume = true; 4143 } 4144 } 4145 if (mFlushPending) { 4146 flushHw_l(); 4147 mFlushPending = false; 4148 } 4149 if (!mStandby && doHwResume) { 4150 mOutput->stream->resume(mOutput->stream); 4151 } 4152 4153 // remove all the tracks that need to be... 4154 removeTracks_l(*tracksToRemove); 4155 4156 return mixerStatus; 4157} 4158 4159void AudioFlinger::OffloadThread::flushOutput_l() 4160{ 4161 mFlushPending = true; 4162} 4163 4164// must be called with thread mutex locked 4165bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4166{ 4167 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4168 mWriteAckSequence, mDrainSequence); 4169 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4170 return true; 4171 } 4172 return false; 4173} 4174 4175// must be called with thread mutex locked 4176bool AudioFlinger::OffloadThread::shouldStandby_l() 4177{ 4178 bool trackPaused = false; 4179 4180 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4181 // after a timeout and we will enter standby then. 4182 if (mTracks.size() > 0) { 4183 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4184 } 4185 4186 return !mStandby && !trackPaused; 4187} 4188 4189 4190bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4191{ 4192 Mutex::Autolock _l(mLock); 4193 return waitingAsyncCallback_l(); 4194} 4195 4196void AudioFlinger::OffloadThread::flushHw_l() 4197{ 4198 mOutput->stream->flush(mOutput->stream); 4199 // Flush anything still waiting in the mixbuffer 4200 mCurrentWriteLength = 0; 4201 mBytesRemaining = 0; 4202 mPausedWriteLength = 0; 4203 mPausedBytesRemaining = 0; 4204 if (mUseAsyncWrite) { 4205 // discard any pending drain or write ack by incrementing sequence 4206 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4207 mDrainSequence = (mDrainSequence + 2) & ~1; 4208 ALOG_ASSERT(mCallbackThread != 0); 4209 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4210 mCallbackThread->setDraining(mDrainSequence); 4211 } 4212} 4213 4214// ---------------------------------------------------------------------------- 4215 4216AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4217 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4218 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4219 DUPLICATING), 4220 mWaitTimeMs(UINT_MAX) 4221{ 4222 addOutputTrack(mainThread); 4223} 4224 4225AudioFlinger::DuplicatingThread::~DuplicatingThread() 4226{ 4227 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4228 mOutputTracks[i]->destroy(); 4229 } 4230} 4231 4232void AudioFlinger::DuplicatingThread::threadLoop_mix() 4233{ 4234 // mix buffers... 4235 if (outputsReady(outputTracks)) { 4236 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4237 } else { 4238 memset(mMixBuffer, 0, mixBufferSize); 4239 } 4240 sleepTime = 0; 4241 writeFrames = mNormalFrameCount; 4242 mCurrentWriteLength = mixBufferSize; 4243 standbyTime = systemTime() + standbyDelay; 4244} 4245 4246void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4247{ 4248 if (sleepTime == 0) { 4249 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4250 sleepTime = activeSleepTime; 4251 } else { 4252 sleepTime = idleSleepTime; 4253 } 4254 } else if (mBytesWritten != 0) { 4255 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4256 writeFrames = mNormalFrameCount; 4257 memset(mMixBuffer, 0, mixBufferSize); 4258 } else { 4259 // flush remaining overflow buffers in output tracks 4260 writeFrames = 0; 4261 } 4262 sleepTime = 0; 4263 } 4264} 4265 4266ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4267{ 4268 for (size_t i = 0; i < outputTracks.size(); i++) { 4269 outputTracks[i]->write(mMixBuffer, writeFrames); 4270 } 4271 mStandby = false; 4272 return (ssize_t)mixBufferSize; 4273} 4274 4275void AudioFlinger::DuplicatingThread::threadLoop_standby() 4276{ 4277 // DuplicatingThread implements standby by stopping all tracks 4278 for (size_t i = 0; i < outputTracks.size(); i++) { 4279 outputTracks[i]->stop(); 4280 } 4281} 4282 4283void AudioFlinger::DuplicatingThread::saveOutputTracks() 4284{ 4285 outputTracks = mOutputTracks; 4286} 4287 4288void AudioFlinger::DuplicatingThread::clearOutputTracks() 4289{ 4290 outputTracks.clear(); 4291} 4292 4293void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4294{ 4295 Mutex::Autolock _l(mLock); 4296 // FIXME explain this formula 4297 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4298 OutputTrack *outputTrack = new OutputTrack(thread, 4299 this, 4300 mSampleRate, 4301 mFormat, 4302 mChannelMask, 4303 frameCount, 4304 IPCThreadState::self()->getCallingUid()); 4305 if (outputTrack->cblk() != NULL) { 4306 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4307 mOutputTracks.add(outputTrack); 4308 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4309 updateWaitTime_l(); 4310 } 4311} 4312 4313void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4314{ 4315 Mutex::Autolock _l(mLock); 4316 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4317 if (mOutputTracks[i]->thread() == thread) { 4318 mOutputTracks[i]->destroy(); 4319 mOutputTracks.removeAt(i); 4320 updateWaitTime_l(); 4321 return; 4322 } 4323 } 4324 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4325} 4326 4327// caller must hold mLock 4328void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4329{ 4330 mWaitTimeMs = UINT_MAX; 4331 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4332 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4333 if (strong != 0) { 4334 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4335 if (waitTimeMs < mWaitTimeMs) { 4336 mWaitTimeMs = waitTimeMs; 4337 } 4338 } 4339 } 4340} 4341 4342 4343bool AudioFlinger::DuplicatingThread::outputsReady( 4344 const SortedVector< sp<OutputTrack> > &outputTracks) 4345{ 4346 for (size_t i = 0; i < outputTracks.size(); i++) { 4347 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4348 if (thread == 0) { 4349 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4350 outputTracks[i].get()); 4351 return false; 4352 } 4353 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4354 // see note at standby() declaration 4355 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4356 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4357 thread.get()); 4358 return false; 4359 } 4360 } 4361 return true; 4362} 4363 4364uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4365{ 4366 return (mWaitTimeMs * 1000) / 2; 4367} 4368 4369void AudioFlinger::DuplicatingThread::cacheParameters_l() 4370{ 4371 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4372 updateWaitTime_l(); 4373 4374 MixerThread::cacheParameters_l(); 4375} 4376 4377// ---------------------------------------------------------------------------- 4378// Record 4379// ---------------------------------------------------------------------------- 4380 4381AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4382 AudioStreamIn *input, 4383 uint32_t sampleRate, 4384 audio_channel_mask_t channelMask, 4385 audio_io_handle_t id, 4386 audio_devices_t outDevice, 4387 audio_devices_t inDevice 4388#ifdef TEE_SINK 4389 , const sp<NBAIO_Sink>& teeSink 4390#endif 4391 ) : 4392 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4393 mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4394 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4395 // are set by readInputParameters() 4396 // mRsmpInIndex LEGACY 4397 mReqChannelCount(popcount(channelMask)), 4398 mReqSampleRate(sampleRate) 4399 // mBytesRead is only meaningful while active, and so is cleared in start() 4400 // (but might be better to also clear here for dump?) 4401#ifdef TEE_SINK 4402 , mTeeSink(teeSink) 4403#endif 4404{ 4405 snprintf(mName, kNameLength, "AudioIn_%X", id); 4406 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4407 4408 readInputParameters(); 4409} 4410 4411 4412AudioFlinger::RecordThread::~RecordThread() 4413{ 4414 mAudioFlinger->unregisterWriter(mNBLogWriter); 4415 delete[] mRsmpInBuffer; 4416 delete mResampler; 4417 delete[] mRsmpOutBuffer; 4418} 4419 4420void AudioFlinger::RecordThread::onFirstRef() 4421{ 4422 run(mName, PRIORITY_URGENT_AUDIO); 4423} 4424 4425bool AudioFlinger::RecordThread::threadLoop() 4426{ 4427 nsecs_t lastWarning = 0; 4428 4429 inputStandBy(); 4430 4431 // used to verify we've read at least once before evaluating how many bytes were read 4432 bool readOnce = false; 4433 4434 // used to request a deferred sleep, to be executed later while mutex is unlocked 4435 bool doSleep = false; 4436 4437reacquire_wakelock: 4438 sp<RecordTrack> activeTrack; 4439 int activeTracksGen; 4440 { 4441 Mutex::Autolock _l(mLock); 4442 size_t size = mActiveTracks.size(); 4443 activeTracksGen = mActiveTracksGen; 4444 if (size > 0) { 4445 // FIXME an arbitrary choice 4446 activeTrack = mActiveTracks[0]; 4447 acquireWakeLock_l(activeTrack->uid()); 4448 if (size > 1) { 4449 SortedVector<int> tmp; 4450 for (size_t i = 0; i < size; i++) { 4451 tmp.add(mActiveTracks[i]->uid()); 4452 } 4453 updateWakeLockUids_l(tmp); 4454 } 4455 } else { 4456 acquireWakeLock_l(-1); 4457 } 4458 } 4459 4460 // start recording 4461 for (;;) { 4462 TrackBase::track_state activeTrackState; 4463 Vector< sp<EffectChain> > effectChains; 4464 4465 // sleep with mutex unlocked 4466 if (doSleep) { 4467 doSleep = false; 4468 usleep(kRecordThreadSleepUs); 4469 } 4470 4471 { // scope for mLock 4472 Mutex::Autolock _l(mLock); 4473 if (exitPending()) { 4474 break; 4475 } 4476 processConfigEvents_l(); 4477 // return value 'reconfig' is currently unused 4478 bool reconfig = checkForNewParameters_l(); 4479 4480 // if no active track(s), then standby and release wakelock 4481 size_t size = mActiveTracks.size(); 4482 if (size == 0) { 4483 standbyIfNotAlreadyInStandby(); 4484 // exitPending() can't become true here 4485 releaseWakeLock_l(); 4486 ALOGV("RecordThread: loop stopping"); 4487 // go to sleep 4488 mWaitWorkCV.wait(mLock); 4489 ALOGV("RecordThread: loop starting"); 4490 goto reacquire_wakelock; 4491 } 4492 4493 if (mActiveTracksGen != activeTracksGen) { 4494 activeTracksGen = mActiveTracksGen; 4495 SortedVector<int> tmp; 4496 for (size_t i = 0; i < size; i++) { 4497 tmp.add(mActiveTracks[i]->uid()); 4498 } 4499 updateWakeLockUids_l(tmp); 4500 // FIXME an arbitrary choice 4501 activeTrack = mActiveTracks[0]; 4502 } 4503 4504 if (activeTrack->isTerminated()) { 4505 removeTrack_l(activeTrack); 4506 mActiveTracks.remove(activeTrack); 4507 mActiveTracksGen++; 4508 continue; 4509 } 4510 4511 activeTrackState = activeTrack->mState; 4512 switch (activeTrackState) { 4513 case TrackBase::PAUSING: 4514 standbyIfNotAlreadyInStandby(); 4515 mActiveTracks.remove(activeTrack); 4516 mActiveTracksGen++; 4517 mStartStopCond.broadcast(); 4518 doSleep = true; 4519 continue; 4520 4521 case TrackBase::RESUMING: 4522 mStandby = false; 4523 if (mReqChannelCount != activeTrack->channelCount()) { 4524 mActiveTracks.remove(activeTrack); 4525 mActiveTracksGen++; 4526 mStartStopCond.broadcast(); 4527 continue; 4528 } 4529 if (readOnce) { 4530 mStartStopCond.broadcast(); 4531 // record start succeeds only if first read from audio input succeeds 4532 if (mBytesRead < 0) { 4533 mActiveTracks.remove(activeTrack); 4534 mActiveTracksGen++; 4535 continue; 4536 } 4537 activeTrack->mState = TrackBase::ACTIVE; 4538 } 4539 break; 4540 4541 case TrackBase::ACTIVE: 4542 break; 4543 4544 case TrackBase::IDLE: 4545 doSleep = true; 4546 continue; 4547 4548 default: 4549 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4550 } 4551 4552 lockEffectChains_l(effectChains); 4553 } 4554 4555 // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable 4556 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4557 4558 for (size_t i = 0; i < effectChains.size(); i ++) { 4559 // thread mutex is not locked, but effect chain is locked 4560 effectChains[i]->process_l(); 4561 } 4562 4563 AudioBufferProvider::Buffer buffer; 4564 buffer.frameCount = mFrameCount; 4565 status_t status = activeTrack->getNextBuffer(&buffer); 4566 if (status == NO_ERROR) { 4567 readOnce = true; 4568 size_t framesOut = buffer.frameCount; 4569 if (mResampler == NULL) { 4570 // no resampling 4571 while (framesOut) { 4572 size_t framesIn = mFrameCount - mRsmpInIndex; 4573 if (framesIn > 0) { 4574 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4575 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4576 activeTrack->mFrameSize; 4577 if (framesIn > framesOut) { 4578 framesIn = framesOut; 4579 } 4580 mRsmpInIndex += framesIn; 4581 framesOut -= framesIn; 4582 if (mChannelCount == mReqChannelCount) { 4583 memcpy(dst, src, framesIn * mFrameSize); 4584 } else { 4585 if (mChannelCount == 1) { 4586 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4587 (int16_t *)src, framesIn); 4588 } else { 4589 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4590 (int16_t *)src, framesIn); 4591 } 4592 } 4593 } 4594 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4595 void *readInto; 4596 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4597 readInto = buffer.raw; 4598 framesOut = 0; 4599 } else { 4600 readInto = mRsmpInBuffer; 4601 mRsmpInIndex = 0; 4602 } 4603 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4604 mBufferSize); 4605 if (mBytesRead <= 0) { 4606 // TODO: verify that it's benign to use a stale track state 4607 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4608 { 4609 ALOGE("Error reading audio input"); 4610 // Force input into standby so that it tries to 4611 // recover at next read attempt 4612 inputStandBy(); 4613 doSleep = true; 4614 } 4615 mRsmpInIndex = mFrameCount; 4616 framesOut = 0; 4617 buffer.frameCount = 0; 4618 } 4619#ifdef TEE_SINK 4620 else if (mTeeSink != 0) { 4621 (void) mTeeSink->write(readInto, 4622 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4623 } 4624#endif 4625 } 4626 } 4627 } else { 4628 // resampling 4629 4630 // avoid busy-waiting if client doesn't keep up 4631 bool madeProgress = false; 4632 4633 // keep mRsmpInBuffer full so resampler always has sufficient input 4634 for (;;) { 4635 int32_t rear = mRsmpInRear; 4636 ssize_t filled = rear - mRsmpInFront; 4637 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4638 // exit once there is enough data in buffer for resampler 4639 if ((size_t) filled >= mRsmpInFrames) { 4640 break; 4641 } 4642 size_t avail = mRsmpInFramesP2 - filled; 4643 // Only try to read full HAL buffers. 4644 // But if the HAL read returns a partial buffer, use it. 4645 if (avail < mFrameCount) { 4646 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4647 avail, mFrameCount); 4648 break; 4649 } 4650 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4651 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4652 rear &= mRsmpInFramesP2 - 1; 4653 mBytesRead = mInput->stream->read(mInput->stream, 4654 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4655 if (mBytesRead <= 0) { 4656 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4657 break; 4658 } 4659 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4660 size_t framesRead = mBytesRead / mFrameSize; 4661 ALOG_ASSERT(framesRead > 0); 4662 madeProgress = true; 4663 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4664 size_t part1 = mRsmpInFramesP2 - rear; 4665 if (framesRead > part1) { 4666 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4667 (framesRead - part1) * mFrameSize); 4668 } 4669 mRsmpInRear += framesRead; 4670 } 4671 4672 if (!madeProgress) { 4673 ALOGV("Did not make progress"); 4674 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4675 } 4676 4677 // resampler accumulates, but we only have one source track 4678 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4679 mResampler->resample(mRsmpOutBuffer, framesOut, 4680 this /* AudioBufferProvider* */); 4681 // ditherAndClamp() works as long as all buffers returned by 4682 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4683 if (mReqChannelCount == 1) { 4684 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4685 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4686 // the resampler always outputs stereo samples: 4687 // do post stereo to mono conversion 4688 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4689 framesOut); 4690 } else { 4691 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4692 } 4693 // now done with mRsmpOutBuffer 4694 4695 } 4696 if (mFramestoDrop == 0) { 4697 activeTrack->releaseBuffer(&buffer); 4698 } else { 4699 if (mFramestoDrop > 0) { 4700 mFramestoDrop -= buffer.frameCount; 4701 if (mFramestoDrop <= 0) { 4702 clearSyncStartEvent(); 4703 } 4704 } else { 4705 mFramestoDrop += buffer.frameCount; 4706 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4707 mSyncStartEvent->isCancelled()) { 4708 ALOGW("Synced record %s, session %d, trigger session %d", 4709 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4710 activeTrack->sessionId(), 4711 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4712 clearSyncStartEvent(); 4713 } 4714 } 4715 } 4716 activeTrack->clearOverflow(); 4717 } 4718 // client isn't retrieving buffers fast enough 4719 else { 4720 if (!activeTrack->setOverflow()) { 4721 nsecs_t now = systemTime(); 4722 if ((now - lastWarning) > kWarningThrottleNs) { 4723 ALOGW("RecordThread: buffer overflow"); 4724 lastWarning = now; 4725 } 4726 } 4727 // Release the processor for a while before asking for a new buffer. 4728 // This will give the application more chance to read from the buffer and 4729 // clear the overflow. 4730 doSleep = true; 4731 } 4732 4733 // enable changes in effect chain 4734 unlockEffectChains(effectChains); 4735 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4736 } 4737 4738 standbyIfNotAlreadyInStandby(); 4739 4740 { 4741 Mutex::Autolock _l(mLock); 4742 for (size_t i = 0; i < mTracks.size(); i++) { 4743 sp<RecordTrack> track = mTracks[i]; 4744 track->invalidate(); 4745 } 4746 mActiveTracks.clear(); 4747 mActiveTracksGen++; 4748 mStartStopCond.broadcast(); 4749 } 4750 4751 releaseWakeLock(); 4752 4753 ALOGV("RecordThread %p exiting", this); 4754 return false; 4755} 4756 4757void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4758{ 4759 if (!mStandby) { 4760 inputStandBy(); 4761 mStandby = true; 4762 } 4763} 4764 4765void AudioFlinger::RecordThread::inputStandBy() 4766{ 4767 mInput->stream->common.standby(&mInput->stream->common); 4768} 4769 4770sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4771 const sp<AudioFlinger::Client>& client, 4772 uint32_t sampleRate, 4773 audio_format_t format, 4774 audio_channel_mask_t channelMask, 4775 size_t frameCount, 4776 int sessionId, 4777 int uid, 4778 IAudioFlinger::track_flags_t *flags, 4779 pid_t tid, 4780 status_t *status) 4781{ 4782 sp<RecordTrack> track; 4783 status_t lStatus; 4784 4785 lStatus = initCheck(); 4786 if (lStatus != NO_ERROR) { 4787 ALOGE("createRecordTrack_l() audio driver not initialized"); 4788 goto Exit; 4789 } 4790 // client expresses a preference for FAST, but we get the final say 4791 if (*flags & IAudioFlinger::TRACK_FAST) { 4792 if ( 4793 // use case: callback handler and frame count is default or at least as large as HAL 4794 ( 4795 (tid != -1) && 4796 ((frameCount == 0) || 4797 (frameCount >= mFrameCount)) 4798 ) && 4799 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4800 // mono or stereo 4801 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4802 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4803 // hardware sample rate 4804 (sampleRate == mSampleRate) && 4805 // record thread has an associated fast recorder 4806 hasFastRecorder() 4807 // FIXME test that RecordThread for this fast track has a capable output HAL 4808 // FIXME add a permission test also? 4809 ) { 4810 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4811 if (frameCount == 0) { 4812 frameCount = mFrameCount * kFastTrackMultiplier; 4813 } 4814 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4815 frameCount, mFrameCount); 4816 } else { 4817 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4818 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4819 "hasFastRecorder=%d tid=%d", 4820 frameCount, mFrameCount, format, 4821 audio_is_linear_pcm(format), 4822 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4823 *flags &= ~IAudioFlinger::TRACK_FAST; 4824 // For compatibility with AudioRecord calculation, buffer depth is forced 4825 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4826 // This is probably too conservative, but legacy application code may depend on it. 4827 // If you change this calculation, also review the start threshold which is related. 4828 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4829 size_t mNormalFrameCount = 2048; // FIXME 4830 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4831 if (minBufCount < 2) { 4832 minBufCount = 2; 4833 } 4834 size_t minFrameCount = mNormalFrameCount * minBufCount; 4835 if (frameCount < minFrameCount) { 4836 frameCount = minFrameCount; 4837 } 4838 } 4839 } 4840 4841 // FIXME use flags and tid similar to createTrack_l() 4842 4843 { // scope for mLock 4844 Mutex::Autolock _l(mLock); 4845 4846 track = new RecordTrack(this, client, sampleRate, 4847 format, channelMask, frameCount, sessionId, uid); 4848 4849 lStatus = track->initCheck(); 4850 if (lStatus != NO_ERROR) { 4851 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4852 track.clear(); 4853 goto Exit; 4854 } 4855 mTracks.add(track); 4856 4857 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4858 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4859 mAudioFlinger->btNrecIsOff(); 4860 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4861 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4862 4863 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4864 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4865 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4866 // so ask activity manager to do this on our behalf 4867 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4868 } 4869 } 4870 lStatus = NO_ERROR; 4871 4872Exit: 4873 *status = lStatus; 4874 return track; 4875} 4876 4877status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4878 AudioSystem::sync_event_t event, 4879 int triggerSession) 4880{ 4881 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4882 sp<ThreadBase> strongMe = this; 4883 status_t status = NO_ERROR; 4884 4885 if (event == AudioSystem::SYNC_EVENT_NONE) { 4886 clearSyncStartEvent(); 4887 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4888 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4889 triggerSession, 4890 recordTrack->sessionId(), 4891 syncStartEventCallback, 4892 this); 4893 // Sync event can be cancelled by the trigger session if the track is not in a 4894 // compatible state in which case we start record immediately 4895 if (mSyncStartEvent->isCancelled()) { 4896 clearSyncStartEvent(); 4897 } else { 4898 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4899 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4900 } 4901 } 4902 4903 { 4904 // This section is a rendezvous between binder thread executing start() and RecordThread 4905 AutoMutex lock(mLock); 4906 if (mActiveTracks.size() > 0) { 4907 // FIXME does not work for multiple active tracks 4908 if (mActiveTracks.indexOf(recordTrack) != 0) { 4909 status = -EBUSY; 4910 } else if (recordTrack->mState == TrackBase::PAUSING) { 4911 recordTrack->mState = TrackBase::ACTIVE; 4912 } 4913 return status; 4914 } 4915 4916 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4917 recordTrack->mState = TrackBase::IDLE; 4918 mActiveTracks.add(recordTrack); 4919 mActiveTracksGen++; 4920 mLock.unlock(); 4921 status_t status = AudioSystem::startInput(mId); 4922 mLock.lock(); 4923 // FIXME should verify that mActiveTrack is still == recordTrack 4924 if (status != NO_ERROR) { 4925 mActiveTracks.remove(recordTrack); 4926 mActiveTracksGen++; 4927 clearSyncStartEvent(); 4928 return status; 4929 } 4930 // FIXME LEGACY 4931 mRsmpInIndex = mFrameCount; 4932 mRsmpInFront = 0; 4933 mRsmpInRear = 0; 4934 mRsmpInUnrel = 0; 4935 mBytesRead = 0; 4936 if (mResampler != NULL) { 4937 mResampler->reset(); 4938 } 4939 // FIXME hijacking a playback track state name which was intended for start after pause; 4940 // here 'STARTING_2' would be more accurate 4941 recordTrack->mState = TrackBase::RESUMING; 4942 // signal thread to start 4943 ALOGV("Signal record thread"); 4944 mWaitWorkCV.broadcast(); 4945 // do not wait for mStartStopCond if exiting 4946 if (exitPending()) { 4947 mActiveTracks.remove(recordTrack); 4948 mActiveTracksGen++; 4949 status = INVALID_OPERATION; 4950 goto startError; 4951 } 4952 // FIXME incorrect usage of wait: no explicit predicate or loop 4953 mStartStopCond.wait(mLock); 4954 if (mActiveTracks.indexOf(recordTrack) < 0) { 4955 ALOGV("Record failed to start"); 4956 status = BAD_VALUE; 4957 goto startError; 4958 } 4959 ALOGV("Record started OK"); 4960 return status; 4961 } 4962 4963startError: 4964 AudioSystem::stopInput(mId); 4965 clearSyncStartEvent(); 4966 return status; 4967} 4968 4969void AudioFlinger::RecordThread::clearSyncStartEvent() 4970{ 4971 if (mSyncStartEvent != 0) { 4972 mSyncStartEvent->cancel(); 4973 } 4974 mSyncStartEvent.clear(); 4975 mFramestoDrop = 0; 4976} 4977 4978void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4979{ 4980 sp<SyncEvent> strongEvent = event.promote(); 4981 4982 if (strongEvent != 0) { 4983 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4984 me->handleSyncStartEvent(strongEvent); 4985 } 4986} 4987 4988void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4989{ 4990 if (event == mSyncStartEvent) { 4991 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4992 // from audio HAL 4993 mFramestoDrop = mFrameCount * 2; 4994 } 4995} 4996 4997bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4998 ALOGV("RecordThread::stop"); 4999 AutoMutex _l(mLock); 5000 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5001 return false; 5002 } 5003 // note that threadLoop may still be processing the track at this point [without lock] 5004 recordTrack->mState = TrackBase::PAUSING; 5005 // do not wait for mStartStopCond if exiting 5006 if (exitPending()) { 5007 return true; 5008 } 5009 // FIXME incorrect usage of wait: no explicit predicate or loop 5010 mStartStopCond.wait(mLock); 5011 // if we have been restarted, recordTrack is in mActiveTracks here 5012 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5013 ALOGV("Record stopped OK"); 5014 return true; 5015 } 5016 return false; 5017} 5018 5019bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 5020{ 5021 return false; 5022} 5023 5024status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5025{ 5026#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5027 if (!isValidSyncEvent(event)) { 5028 return BAD_VALUE; 5029 } 5030 5031 int eventSession = event->triggerSession(); 5032 status_t ret = NAME_NOT_FOUND; 5033 5034 Mutex::Autolock _l(mLock); 5035 5036 for (size_t i = 0; i < mTracks.size(); i++) { 5037 sp<RecordTrack> track = mTracks[i]; 5038 if (eventSession == track->sessionId()) { 5039 (void) track->setSyncEvent(event); 5040 ret = NO_ERROR; 5041 } 5042 } 5043 return ret; 5044#else 5045 return BAD_VALUE; 5046#endif 5047} 5048 5049// destroyTrack_l() must be called with ThreadBase::mLock held 5050void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5051{ 5052 track->terminate(); 5053 track->mState = TrackBase::STOPPED; 5054 // active tracks are removed by threadLoop() 5055 if (mActiveTracks.indexOf(track) < 0) { 5056 removeTrack_l(track); 5057 } 5058} 5059 5060void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5061{ 5062 mTracks.remove(track); 5063 // need anything related to effects here? 5064} 5065 5066void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5067{ 5068 dumpInternals(fd, args); 5069 dumpTracks(fd, args); 5070 dumpEffectChains(fd, args); 5071} 5072 5073void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5074{ 5075 const size_t SIZE = 256; 5076 char buffer[SIZE]; 5077 String8 result; 5078 5079 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5080 result.append(buffer); 5081 5082 if (mActiveTracks.size() > 0) { 5083 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5084 result.append(buffer); 5085 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5086 result.append(buffer); 5087 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5088 result.append(buffer); 5089 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5090 result.append(buffer); 5091 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5092 result.append(buffer); 5093 } else { 5094 result.append("No active record client\n"); 5095 } 5096 5097 write(fd, result.string(), result.size()); 5098 5099 dumpBase(fd, args); 5100} 5101 5102void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 5103{ 5104 const size_t SIZE = 256; 5105 char buffer[SIZE]; 5106 String8 result; 5107 5108 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5109 result.append(buffer); 5110 RecordTrack::appendDumpHeader(result); 5111 for (size_t i = 0; i < mTracks.size(); ++i) { 5112 sp<RecordTrack> track = mTracks[i]; 5113 if (track != 0) { 5114 track->dump(buffer, SIZE); 5115 result.append(buffer); 5116 } 5117 } 5118 5119 size_t size = mActiveTracks.size(); 5120 if (size > 0) { 5121 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5122 result.append(buffer); 5123 RecordTrack::appendDumpHeader(result); 5124 for (size_t i = 0; i < size; ++i) { 5125 sp<RecordTrack> track = mActiveTracks[i]; 5126 track->dump(buffer, SIZE); 5127 result.append(buffer); 5128 } 5129 5130 } 5131 write(fd, result.string(), result.size()); 5132} 5133 5134// AudioBufferProvider interface 5135status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5136{ 5137 int32_t rear = mRsmpInRear; 5138 int32_t front = mRsmpInFront; 5139 ssize_t filled = rear - front; 5140 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5141 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5142 front &= mRsmpInFramesP2 - 1; 5143 size_t part1 = mRsmpInFramesP2 - front; 5144 if (part1 > (size_t) filled) { 5145 part1 = filled; 5146 } 5147 size_t ask = buffer->frameCount; 5148 ALOG_ASSERT(ask > 0); 5149 if (part1 > ask) { 5150 part1 = ask; 5151 } 5152 if (part1 == 0) { 5153 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5154 ALOGE("RecordThread::getNextBuffer() starved"); 5155 buffer->raw = NULL; 5156 buffer->frameCount = 0; 5157 mRsmpInUnrel = 0; 5158 return NOT_ENOUGH_DATA; 5159 } 5160 5161 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5162 buffer->frameCount = part1; 5163 mRsmpInUnrel = part1; 5164 return NO_ERROR; 5165} 5166 5167// AudioBufferProvider interface 5168void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5169{ 5170 size_t stepCount = buffer->frameCount; 5171 if (stepCount == 0) { 5172 return; 5173 } 5174 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5175 mRsmpInUnrel -= stepCount; 5176 mRsmpInFront += stepCount; 5177 buffer->raw = NULL; 5178 buffer->frameCount = 0; 5179} 5180 5181bool AudioFlinger::RecordThread::checkForNewParameters_l() 5182{ 5183 bool reconfig = false; 5184 5185 while (!mNewParameters.isEmpty()) { 5186 status_t status = NO_ERROR; 5187 String8 keyValuePair = mNewParameters[0]; 5188 AudioParameter param = AudioParameter(keyValuePair); 5189 int value; 5190 audio_format_t reqFormat = mFormat; 5191 uint32_t reqSamplingRate = mReqSampleRate; 5192 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5193 5194 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5195 reqSamplingRate = value; 5196 reconfig = true; 5197 } 5198 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5199 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5200 status = BAD_VALUE; 5201 } else { 5202 reqFormat = (audio_format_t) value; 5203 reconfig = true; 5204 } 5205 } 5206 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5207 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5208 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5209 status = BAD_VALUE; 5210 } else { 5211 reqChannelMask = mask; 5212 reconfig = true; 5213 } 5214 } 5215 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5216 // do not accept frame count changes if tracks are open as the track buffer 5217 // size depends on frame count and correct behavior would not be guaranteed 5218 // if frame count is changed after track creation 5219 if (mActiveTracks.size() > 0) { 5220 status = INVALID_OPERATION; 5221 } else { 5222 reconfig = true; 5223 } 5224 } 5225 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5226 // forward device change to effects that have requested to be 5227 // aware of attached audio device. 5228 for (size_t i = 0; i < mEffectChains.size(); i++) { 5229 mEffectChains[i]->setDevice_l(value); 5230 } 5231 5232 // store input device and output device but do not forward output device to audio HAL. 5233 // Note that status is ignored by the caller for output device 5234 // (see AudioFlinger::setParameters() 5235 if (audio_is_output_devices(value)) { 5236 mOutDevice = value; 5237 status = BAD_VALUE; 5238 } else { 5239 mInDevice = value; 5240 // disable AEC and NS if the device is a BT SCO headset supporting those 5241 // pre processings 5242 if (mTracks.size() > 0) { 5243 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5244 mAudioFlinger->btNrecIsOff(); 5245 for (size_t i = 0; i < mTracks.size(); i++) { 5246 sp<RecordTrack> track = mTracks[i]; 5247 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5248 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5249 } 5250 } 5251 } 5252 } 5253 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5254 mAudioSource != (audio_source_t)value) { 5255 // forward device change to effects that have requested to be 5256 // aware of attached audio device. 5257 for (size_t i = 0; i < mEffectChains.size(); i++) { 5258 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5259 } 5260 mAudioSource = (audio_source_t)value; 5261 } 5262 5263 if (status == NO_ERROR) { 5264 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5265 keyValuePair.string()); 5266 if (status == INVALID_OPERATION) { 5267 inputStandBy(); 5268 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5269 keyValuePair.string()); 5270 } 5271 if (reconfig) { 5272 if (status == BAD_VALUE && 5273 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5274 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5275 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5276 <= (2 * reqSamplingRate)) && 5277 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5278 <= FCC_2 && 5279 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5280 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5281 status = NO_ERROR; 5282 } 5283 if (status == NO_ERROR) { 5284 readInputParameters(); 5285 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5286 } 5287 } 5288 } 5289 5290 mNewParameters.removeAt(0); 5291 5292 mParamStatus = status; 5293 mParamCond.signal(); 5294 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5295 // already timed out waiting for the status and will never signal the condition. 5296 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5297 } 5298 return reconfig; 5299} 5300 5301String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5302{ 5303 Mutex::Autolock _l(mLock); 5304 if (initCheck() != NO_ERROR) { 5305 return String8(); 5306 } 5307 5308 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5309 const String8 out_s8(s); 5310 free(s); 5311 return out_s8; 5312} 5313 5314void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5315 AudioSystem::OutputDescriptor desc; 5316 const void *param2 = NULL; 5317 5318 switch (event) { 5319 case AudioSystem::INPUT_OPENED: 5320 case AudioSystem::INPUT_CONFIG_CHANGED: 5321 desc.channelMask = mChannelMask; 5322 desc.samplingRate = mSampleRate; 5323 desc.format = mFormat; 5324 desc.frameCount = mFrameCount; 5325 desc.latency = 0; 5326 param2 = &desc; 5327 break; 5328 5329 case AudioSystem::INPUT_CLOSED: 5330 default: 5331 break; 5332 } 5333 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5334} 5335 5336void AudioFlinger::RecordThread::readInputParameters() 5337{ 5338 delete[] mRsmpInBuffer; 5339 // mRsmpInBuffer is always assigned a new[] below 5340 delete[] mRsmpOutBuffer; 5341 mRsmpOutBuffer = NULL; 5342 delete mResampler; 5343 mResampler = NULL; 5344 5345 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5346 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5347 mChannelCount = popcount(mChannelMask); 5348 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5349 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5350 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5351 } 5352 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5353 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5354 mFrameCount = mBufferSize / mFrameSize; 5355 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5356 // 1 full output buffer, regardless of the alignment of the available input. 5357 mRsmpInFrames = mFrameCount * 3; 5358 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5359 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5360 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5361 mRsmpInFront = 0; 5362 mRsmpInRear = 0; 5363 mRsmpInUnrel = 0; 5364 5365 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5366 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5367 mResampler->setSampleRate(mSampleRate); 5368 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5369 // resampler always outputs stereo 5370 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5371 } 5372 mRsmpInIndex = mFrameCount; 5373} 5374 5375unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5376{ 5377 Mutex::Autolock _l(mLock); 5378 if (initCheck() != NO_ERROR) { 5379 return 0; 5380 } 5381 5382 return mInput->stream->get_input_frames_lost(mInput->stream); 5383} 5384 5385uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5386{ 5387 Mutex::Autolock _l(mLock); 5388 uint32_t result = 0; 5389 if (getEffectChain_l(sessionId) != 0) { 5390 result = EFFECT_SESSION; 5391 } 5392 5393 for (size_t i = 0; i < mTracks.size(); ++i) { 5394 if (sessionId == mTracks[i]->sessionId()) { 5395 result |= TRACK_SESSION; 5396 break; 5397 } 5398 } 5399 5400 return result; 5401} 5402 5403KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5404{ 5405 KeyedVector<int, bool> ids; 5406 Mutex::Autolock _l(mLock); 5407 for (size_t j = 0; j < mTracks.size(); ++j) { 5408 sp<RecordThread::RecordTrack> track = mTracks[j]; 5409 int sessionId = track->sessionId(); 5410 if (ids.indexOfKey(sessionId) < 0) { 5411 ids.add(sessionId, true); 5412 } 5413 } 5414 return ids; 5415} 5416 5417AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5418{ 5419 Mutex::Autolock _l(mLock); 5420 AudioStreamIn *input = mInput; 5421 mInput = NULL; 5422 return input; 5423} 5424 5425// this method must always be called either with ThreadBase mLock held or inside the thread loop 5426audio_stream_t* AudioFlinger::RecordThread::stream() const 5427{ 5428 if (mInput == NULL) { 5429 return NULL; 5430 } 5431 return &mInput->stream->common; 5432} 5433 5434status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5435{ 5436 // only one chain per input thread 5437 if (mEffectChains.size() != 0) { 5438 return INVALID_OPERATION; 5439 } 5440 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5441 5442 chain->setInBuffer(NULL); 5443 chain->setOutBuffer(NULL); 5444 5445 checkSuspendOnAddEffectChain_l(chain); 5446 5447 mEffectChains.add(chain); 5448 5449 return NO_ERROR; 5450} 5451 5452size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5453{ 5454 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5455 ALOGW_IF(mEffectChains.size() != 1, 5456 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5457 chain.get(), mEffectChains.size(), this); 5458 if (mEffectChains.size() == 1) { 5459 mEffectChains.removeAt(0); 5460 } 5461 return 0; 5462} 5463 5464}; // namespace android 5465