Threads.cpp revision 000a4193dd82549192277fd4b9bb571d8a4c262f
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 *pFrameCount, 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 size_t frameCount = *pFrameCount; 1202 sp<Track> track; 1203 status_t lStatus; 1204 1205 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1206 1207 // client expresses a preference for FAST, but we get the final say 1208 if (*flags & IAudioFlinger::TRACK_FAST) { 1209 if ( 1210 // not timed 1211 (!isTimed) && 1212 // either of these use cases: 1213 ( 1214 // use case 1: shared buffer with any frame count 1215 ( 1216 (sharedBuffer != 0) 1217 ) || 1218 // use case 2: callback handler and frame count is default or at least as large as HAL 1219 ( 1220 (tid != -1) && 1221 ((frameCount == 0) || 1222 (frameCount >= mFrameCount)) 1223 ) 1224 ) && 1225 // PCM data 1226 audio_is_linear_pcm(format) && 1227 // mono or stereo 1228 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1229 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1230#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1231 // hardware sample rate 1232 (sampleRate == mSampleRate) && 1233#endif 1234 // normal mixer has an associated fast mixer 1235 hasFastMixer() && 1236 // there are sufficient fast track slots available 1237 (mFastTrackAvailMask != 0) 1238 // FIXME test that MixerThread for this fast track has a capable output HAL 1239 // FIXME add a permission test also? 1240 ) { 1241 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1242 if (frameCount == 0) { 1243 frameCount = mFrameCount * kFastTrackMultiplier; 1244 } 1245 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1246 frameCount, mFrameCount); 1247 } else { 1248 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1249 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1250 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1251 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1252 audio_is_linear_pcm(format), 1253 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1254 *flags &= ~IAudioFlinger::TRACK_FAST; 1255 // For compatibility with AudioTrack calculation, buffer depth is forced 1256 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1257 // This is probably too conservative, but legacy application code may depend on it. 1258 // If you change this calculation, also review the start threshold which is related. 1259 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1260 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1261 if (minBufCount < 2) { 1262 minBufCount = 2; 1263 } 1264 size_t minFrameCount = mNormalFrameCount * minBufCount; 1265 if (frameCount < minFrameCount) { 1266 frameCount = minFrameCount; 1267 } 1268 } 1269 } 1270 *pFrameCount = frameCount; 1271 1272 if (mType == DIRECT) { 1273 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1274 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1275 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1276 "for output %p with format %d", 1277 sampleRate, format, channelMask, mOutput, mFormat); 1278 lStatus = BAD_VALUE; 1279 goto Exit; 1280 } 1281 } 1282 } else if (mType == OFFLOAD) { 1283 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1284 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1285 "for output %p with format %d", 1286 sampleRate, format, channelMask, mOutput, mFormat); 1287 lStatus = BAD_VALUE; 1288 goto Exit; 1289 } 1290 } else { 1291 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1292 ALOGE("createTrack_l() Bad parameter: format %d \"" 1293 "for output %p with format %d", 1294 format, mOutput, mFormat); 1295 lStatus = BAD_VALUE; 1296 goto Exit; 1297 } 1298 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1299 if (sampleRate > mSampleRate*2) { 1300 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1301 lStatus = BAD_VALUE; 1302 goto Exit; 1303 } 1304 } 1305 1306 lStatus = initCheck(); 1307 if (lStatus != NO_ERROR) { 1308 ALOGE("Audio driver not initialized."); 1309 goto Exit; 1310 } 1311 1312 { // scope for mLock 1313 Mutex::Autolock _l(mLock); 1314 1315 // all tracks in same audio session must share the same routing strategy otherwise 1316 // conflicts will happen when tracks are moved from one output to another by audio policy 1317 // manager 1318 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1319 for (size_t i = 0; i < mTracks.size(); ++i) { 1320 sp<Track> t = mTracks[i]; 1321 if (t != 0 && !t->isOutputTrack()) { 1322 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1323 if (sessionId == t->sessionId() && strategy != actual) { 1324 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1325 strategy, actual); 1326 lStatus = BAD_VALUE; 1327 goto Exit; 1328 } 1329 } 1330 } 1331 1332 if (!isTimed) { 1333 track = new Track(this, client, streamType, sampleRate, format, 1334 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1335 } else { 1336 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1337 channelMask, frameCount, sharedBuffer, sessionId, uid); 1338 } 1339 1340 // new Track always returns non-NULL, 1341 // but TimedTrack::create() is a factory that could fail by returning NULL 1342 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1343 if (lStatus != NO_ERROR) { 1344 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1345 track.clear(); 1346 goto Exit; 1347 } 1348 1349 mTracks.add(track); 1350 1351 sp<EffectChain> chain = getEffectChain_l(sessionId); 1352 if (chain != 0) { 1353 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1354 track->setMainBuffer(chain->inBuffer()); 1355 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1356 chain->incTrackCnt(); 1357 } 1358 1359 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1360 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1361 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1362 // so ask activity manager to do this on our behalf 1363 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1364 } 1365 } 1366 1367 lStatus = NO_ERROR; 1368 1369Exit: 1370 *status = lStatus; 1371 return track; 1372} 1373 1374uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1375{ 1376 return latency; 1377} 1378 1379uint32_t AudioFlinger::PlaybackThread::latency() const 1380{ 1381 Mutex::Autolock _l(mLock); 1382 return latency_l(); 1383} 1384uint32_t AudioFlinger::PlaybackThread::latency_l() const 1385{ 1386 if (initCheck() == NO_ERROR) { 1387 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1388 } else { 1389 return 0; 1390 } 1391} 1392 1393void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1394{ 1395 Mutex::Autolock _l(mLock); 1396 // Don't apply master volume in SW if our HAL can do it for us. 1397 if (mOutput && mOutput->audioHwDev && 1398 mOutput->audioHwDev->canSetMasterVolume()) { 1399 mMasterVolume = 1.0; 1400 } else { 1401 mMasterVolume = value; 1402 } 1403} 1404 1405void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1406{ 1407 Mutex::Autolock _l(mLock); 1408 // Don't apply master mute in SW if our HAL can do it for us. 1409 if (mOutput && mOutput->audioHwDev && 1410 mOutput->audioHwDev->canSetMasterMute()) { 1411 mMasterMute = false; 1412 } else { 1413 mMasterMute = muted; 1414 } 1415} 1416 1417void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1418{ 1419 Mutex::Autolock _l(mLock); 1420 mStreamTypes[stream].volume = value; 1421 broadcast_l(); 1422} 1423 1424void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1425{ 1426 Mutex::Autolock _l(mLock); 1427 mStreamTypes[stream].mute = muted; 1428 broadcast_l(); 1429} 1430 1431float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1432{ 1433 Mutex::Autolock _l(mLock); 1434 return mStreamTypes[stream].volume; 1435} 1436 1437// addTrack_l() must be called with ThreadBase::mLock held 1438status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1439{ 1440 status_t status = ALREADY_EXISTS; 1441 1442 // set retry count for buffer fill 1443 track->mRetryCount = kMaxTrackStartupRetries; 1444 if (mActiveTracks.indexOf(track) < 0) { 1445 // the track is newly added, make sure it fills up all its 1446 // buffers before playing. This is to ensure the client will 1447 // effectively get the latency it requested. 1448 if (!track->isOutputTrack()) { 1449 TrackBase::track_state state = track->mState; 1450 mLock.unlock(); 1451 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1452 mLock.lock(); 1453 // abort track was stopped/paused while we released the lock 1454 if (state != track->mState) { 1455 if (status == NO_ERROR) { 1456 mLock.unlock(); 1457 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1458 mLock.lock(); 1459 } 1460 return INVALID_OPERATION; 1461 } 1462 // abort if start is rejected by audio policy manager 1463 if (status != NO_ERROR) { 1464 return PERMISSION_DENIED; 1465 } 1466#ifdef ADD_BATTERY_DATA 1467 // to track the speaker usage 1468 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1469#endif 1470 } 1471 1472 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1473 track->mResetDone = false; 1474 track->mPresentationCompleteFrames = 0; 1475 mActiveTracks.add(track); 1476 mWakeLockUids.add(track->uid()); 1477 mActiveTracksGeneration++; 1478 mLatestActiveTrack = track; 1479 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1480 if (chain != 0) { 1481 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1482 track->sessionId()); 1483 chain->incActiveTrackCnt(); 1484 } 1485 1486 status = NO_ERROR; 1487 } 1488 1489 ALOGV("signal playback thread"); 1490 broadcast_l(); 1491 1492 return status; 1493} 1494 1495bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1496{ 1497 track->terminate(); 1498 // active tracks are removed by threadLoop() 1499 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1500 track->mState = TrackBase::STOPPED; 1501 if (!trackActive) { 1502 removeTrack_l(track); 1503 } else if (track->isFastTrack() || track->isOffloaded()) { 1504 track->mState = TrackBase::STOPPING_1; 1505 } 1506 1507 return trackActive; 1508} 1509 1510void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1511{ 1512 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1513 mTracks.remove(track); 1514 deleteTrackName_l(track->name()); 1515 // redundant as track is about to be destroyed, for dumpsys only 1516 track->mName = -1; 1517 if (track->isFastTrack()) { 1518 int index = track->mFastIndex; 1519 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1520 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1521 mFastTrackAvailMask |= 1 << index; 1522 // redundant as track is about to be destroyed, for dumpsys only 1523 track->mFastIndex = -1; 1524 } 1525 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1526 if (chain != 0) { 1527 chain->decTrackCnt(); 1528 } 1529} 1530 1531void AudioFlinger::PlaybackThread::broadcast_l() 1532{ 1533 // Thread could be blocked waiting for async 1534 // so signal it to handle state changes immediately 1535 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1536 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1537 mSignalPending = true; 1538 mWaitWorkCV.broadcast(); 1539} 1540 1541String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1542{ 1543 Mutex::Autolock _l(mLock); 1544 if (initCheck() != NO_ERROR) { 1545 return String8(); 1546 } 1547 1548 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1549 const String8 out_s8(s); 1550 free(s); 1551 return out_s8; 1552} 1553 1554// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1555void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1556 AudioSystem::OutputDescriptor desc; 1557 void *param2 = NULL; 1558 1559 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1560 param); 1561 1562 switch (event) { 1563 case AudioSystem::OUTPUT_OPENED: 1564 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1565 desc.channelMask = mChannelMask; 1566 desc.samplingRate = mSampleRate; 1567 desc.format = mFormat; 1568 desc.frameCount = mNormalFrameCount; // FIXME see 1569 // AudioFlinger::frameCount(audio_io_handle_t) 1570 desc.latency = latency(); 1571 param2 = &desc; 1572 break; 1573 1574 case AudioSystem::STREAM_CONFIG_CHANGED: 1575 param2 = ¶m; 1576 case AudioSystem::OUTPUT_CLOSED: 1577 default: 1578 break; 1579 } 1580 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1581} 1582 1583void AudioFlinger::PlaybackThread::writeCallback() 1584{ 1585 ALOG_ASSERT(mCallbackThread != 0); 1586 mCallbackThread->resetWriteBlocked(); 1587} 1588 1589void AudioFlinger::PlaybackThread::drainCallback() 1590{ 1591 ALOG_ASSERT(mCallbackThread != 0); 1592 mCallbackThread->resetDraining(); 1593} 1594 1595void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1596{ 1597 Mutex::Autolock _l(mLock); 1598 // reject out of sequence requests 1599 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1600 mWriteAckSequence &= ~1; 1601 mWaitWorkCV.signal(); 1602 } 1603} 1604 1605void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1606{ 1607 Mutex::Autolock _l(mLock); 1608 // reject out of sequence requests 1609 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1610 mDrainSequence &= ~1; 1611 mWaitWorkCV.signal(); 1612 } 1613} 1614 1615// static 1616int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1617 void *param, 1618 void *cookie) 1619{ 1620 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1621 ALOGV("asyncCallback() event %d", event); 1622 switch (event) { 1623 case STREAM_CBK_EVENT_WRITE_READY: 1624 me->writeCallback(); 1625 break; 1626 case STREAM_CBK_EVENT_DRAIN_READY: 1627 me->drainCallback(); 1628 break; 1629 default: 1630 ALOGW("asyncCallback() unknown event %d", event); 1631 break; 1632 } 1633 return 0; 1634} 1635 1636void AudioFlinger::PlaybackThread::readOutputParameters() 1637{ 1638 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1639 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1640 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1641 if (!audio_is_output_channel(mChannelMask)) { 1642 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1643 } 1644 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1645 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1646 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1647 } 1648 mChannelCount = popcount(mChannelMask); 1649 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1650 if (!audio_is_valid_format(mFormat)) { 1651 LOG_FATAL("HAL format %d not valid for output", mFormat); 1652 } 1653 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1654 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1655 mFormat); 1656 } 1657 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1658 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1659 mFrameCount = mBufferSize / mFrameSize; 1660 if (mFrameCount & 15) { 1661 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1662 mFrameCount); 1663 } 1664 1665 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1666 (mOutput->stream->set_callback != NULL)) { 1667 if (mOutput->stream->set_callback(mOutput->stream, 1668 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1669 mUseAsyncWrite = true; 1670 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1671 } 1672 } 1673 1674 // Calculate size of normal mix buffer relative to the HAL output buffer size 1675 double multiplier = 1.0; 1676 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1677 kUseFastMixer == FastMixer_Dynamic)) { 1678 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1679 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1680 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1681 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1682 maxNormalFrameCount = maxNormalFrameCount & ~15; 1683 if (maxNormalFrameCount < minNormalFrameCount) { 1684 maxNormalFrameCount = minNormalFrameCount; 1685 } 1686 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1687 if (multiplier <= 1.0) { 1688 multiplier = 1.0; 1689 } else if (multiplier <= 2.0) { 1690 if (2 * mFrameCount <= maxNormalFrameCount) { 1691 multiplier = 2.0; 1692 } else { 1693 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1694 } 1695 } else { 1696 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1697 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1698 // track, but we sometimes have to do this to satisfy the maximum frame count 1699 // constraint) 1700 // FIXME this rounding up should not be done if no HAL SRC 1701 uint32_t truncMult = (uint32_t) multiplier; 1702 if ((truncMult & 1)) { 1703 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1704 ++truncMult; 1705 } 1706 } 1707 multiplier = (double) truncMult; 1708 } 1709 } 1710 mNormalFrameCount = multiplier * mFrameCount; 1711 // round up to nearest 16 frames to satisfy AudioMixer 1712 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1713 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1714 mNormalFrameCount); 1715 1716 delete[] mMixBuffer; 1717 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1718 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1719 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1720 memset(mMixBuffer, 0, normalBufferSize); 1721 1722 // force reconfiguration of effect chains and engines to take new buffer size and audio 1723 // parameters into account 1724 // Note that mLock is not held when readOutputParameters() is called from the constructor 1725 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1726 // matter. 1727 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1728 Vector< sp<EffectChain> > effectChains = mEffectChains; 1729 for (size_t i = 0; i < effectChains.size(); i ++) { 1730 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1731 } 1732} 1733 1734 1735status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1736{ 1737 if (halFrames == NULL || dspFrames == NULL) { 1738 return BAD_VALUE; 1739 } 1740 Mutex::Autolock _l(mLock); 1741 if (initCheck() != NO_ERROR) { 1742 return INVALID_OPERATION; 1743 } 1744 size_t framesWritten = mBytesWritten / mFrameSize; 1745 *halFrames = framesWritten; 1746 1747 if (isSuspended()) { 1748 // return an estimation of rendered frames when the output is suspended 1749 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1750 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1751 return NO_ERROR; 1752 } else { 1753 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1754 } 1755} 1756 1757uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1758{ 1759 Mutex::Autolock _l(mLock); 1760 uint32_t result = 0; 1761 if (getEffectChain_l(sessionId) != 0) { 1762 result = EFFECT_SESSION; 1763 } 1764 1765 for (size_t i = 0; i < mTracks.size(); ++i) { 1766 sp<Track> track = mTracks[i]; 1767 if (sessionId == track->sessionId() && !track->isInvalid()) { 1768 result |= TRACK_SESSION; 1769 break; 1770 } 1771 } 1772 1773 return result; 1774} 1775 1776uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1777{ 1778 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1779 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1780 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1781 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1782 } 1783 for (size_t i = 0; i < mTracks.size(); i++) { 1784 sp<Track> track = mTracks[i]; 1785 if (sessionId == track->sessionId() && !track->isInvalid()) { 1786 return AudioSystem::getStrategyForStream(track->streamType()); 1787 } 1788 } 1789 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1790} 1791 1792 1793AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1794{ 1795 Mutex::Autolock _l(mLock); 1796 return mOutput; 1797} 1798 1799AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1800{ 1801 Mutex::Autolock _l(mLock); 1802 AudioStreamOut *output = mOutput; 1803 mOutput = NULL; 1804 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1805 // must push a NULL and wait for ack 1806 mOutputSink.clear(); 1807 mPipeSink.clear(); 1808 mNormalSink.clear(); 1809 return output; 1810} 1811 1812// this method must always be called either with ThreadBase mLock held or inside the thread loop 1813audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1814{ 1815 if (mOutput == NULL) { 1816 return NULL; 1817 } 1818 return &mOutput->stream->common; 1819} 1820 1821uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1822{ 1823 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1824} 1825 1826status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1827{ 1828 if (!isValidSyncEvent(event)) { 1829 return BAD_VALUE; 1830 } 1831 1832 Mutex::Autolock _l(mLock); 1833 1834 for (size_t i = 0; i < mTracks.size(); ++i) { 1835 sp<Track> track = mTracks[i]; 1836 if (event->triggerSession() == track->sessionId()) { 1837 (void) track->setSyncEvent(event); 1838 return NO_ERROR; 1839 } 1840 } 1841 1842 return NAME_NOT_FOUND; 1843} 1844 1845bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1846{ 1847 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1848} 1849 1850void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1851 const Vector< sp<Track> >& tracksToRemove) 1852{ 1853 size_t count = tracksToRemove.size(); 1854 if (count > 0) { 1855 for (size_t i = 0 ; i < count ; i++) { 1856 const sp<Track>& track = tracksToRemove.itemAt(i); 1857 if (!track->isOutputTrack()) { 1858 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1859#ifdef ADD_BATTERY_DATA 1860 // to track the speaker usage 1861 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1862#endif 1863 if (track->isTerminated()) { 1864 AudioSystem::releaseOutput(mId); 1865 } 1866 } 1867 } 1868 } 1869} 1870 1871void AudioFlinger::PlaybackThread::checkSilentMode_l() 1872{ 1873 if (!mMasterMute) { 1874 char value[PROPERTY_VALUE_MAX]; 1875 if (property_get("ro.audio.silent", value, "0") > 0) { 1876 char *endptr; 1877 unsigned long ul = strtoul(value, &endptr, 0); 1878 if (*endptr == '\0' && ul != 0) { 1879 ALOGD("Silence is golden"); 1880 // The setprop command will not allow a property to be changed after 1881 // the first time it is set, so we don't have to worry about un-muting. 1882 setMasterMute_l(true); 1883 } 1884 } 1885 } 1886} 1887 1888// shared by MIXER and DIRECT, overridden by DUPLICATING 1889ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1890{ 1891 // FIXME rewrite to reduce number of system calls 1892 mLastWriteTime = systemTime(); 1893 mInWrite = true; 1894 ssize_t bytesWritten; 1895 1896 // If an NBAIO sink is present, use it to write the normal mixer's submix 1897 if (mNormalSink != 0) { 1898#define mBitShift 2 // FIXME 1899 size_t count = mBytesRemaining >> mBitShift; 1900 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1901 ATRACE_BEGIN("write"); 1902 // update the setpoint when AudioFlinger::mScreenState changes 1903 uint32_t screenState = AudioFlinger::mScreenState; 1904 if (screenState != mScreenState) { 1905 mScreenState = screenState; 1906 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1907 if (pipe != NULL) { 1908 pipe->setAvgFrames((mScreenState & 1) ? 1909 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1910 } 1911 } 1912 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1913 ATRACE_END(); 1914 if (framesWritten > 0) { 1915 bytesWritten = framesWritten << mBitShift; 1916 } else { 1917 bytesWritten = framesWritten; 1918 } 1919 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1920 if (status == NO_ERROR) { 1921 size_t totalFramesWritten = mNormalSink->framesWritten(); 1922 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1923 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1924 mLatchDValid = true; 1925 } 1926 } 1927 // otherwise use the HAL / AudioStreamOut directly 1928 } else { 1929 // Direct output and offload threads 1930 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1931 if (mUseAsyncWrite) { 1932 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1933 mWriteAckSequence += 2; 1934 mWriteAckSequence |= 1; 1935 ALOG_ASSERT(mCallbackThread != 0); 1936 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1937 } 1938 // FIXME We should have an implementation of timestamps for direct output threads. 1939 // They are used e.g for multichannel PCM playback over HDMI. 1940 bytesWritten = mOutput->stream->write(mOutput->stream, 1941 (char *)mMixBuffer + offset, mBytesRemaining); 1942 if (mUseAsyncWrite && 1943 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1944 // do not wait for async callback in case of error of full write 1945 mWriteAckSequence &= ~1; 1946 ALOG_ASSERT(mCallbackThread != 0); 1947 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1948 } 1949 } 1950 1951 mNumWrites++; 1952 mInWrite = false; 1953 mStandby = false; 1954 return bytesWritten; 1955} 1956 1957void AudioFlinger::PlaybackThread::threadLoop_drain() 1958{ 1959 if (mOutput->stream->drain) { 1960 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1961 if (mUseAsyncWrite) { 1962 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1963 mDrainSequence |= 1; 1964 ALOG_ASSERT(mCallbackThread != 0); 1965 mCallbackThread->setDraining(mDrainSequence); 1966 } 1967 mOutput->stream->drain(mOutput->stream, 1968 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1969 : AUDIO_DRAIN_ALL); 1970 } 1971} 1972 1973void AudioFlinger::PlaybackThread::threadLoop_exit() 1974{ 1975 // Default implementation has nothing to do 1976} 1977 1978/* 1979The derived values that are cached: 1980 - mixBufferSize from frame count * frame size 1981 - activeSleepTime from activeSleepTimeUs() 1982 - idleSleepTime from idleSleepTimeUs() 1983 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1984 - maxPeriod from frame count and sample rate (MIXER only) 1985 1986The parameters that affect these derived values are: 1987 - frame count 1988 - frame size 1989 - sample rate 1990 - device type: A2DP or not 1991 - device latency 1992 - format: PCM or not 1993 - active sleep time 1994 - idle sleep time 1995*/ 1996 1997void AudioFlinger::PlaybackThread::cacheParameters_l() 1998{ 1999 mixBufferSize = mNormalFrameCount * mFrameSize; 2000 activeSleepTime = activeSleepTimeUs(); 2001 idleSleepTime = idleSleepTimeUs(); 2002} 2003 2004void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2005{ 2006 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2007 this, streamType, mTracks.size()); 2008 Mutex::Autolock _l(mLock); 2009 2010 size_t size = mTracks.size(); 2011 for (size_t i = 0; i < size; i++) { 2012 sp<Track> t = mTracks[i]; 2013 if (t->streamType() == streamType) { 2014 t->invalidate(); 2015 } 2016 } 2017} 2018 2019status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2020{ 2021 int session = chain->sessionId(); 2022 int16_t *buffer = mMixBuffer; 2023 bool ownsBuffer = false; 2024 2025 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2026 if (session > 0) { 2027 // Only one effect chain can be present in direct output thread and it uses 2028 // the mix buffer as input 2029 if (mType != DIRECT) { 2030 size_t numSamples = mNormalFrameCount * mChannelCount; 2031 buffer = new int16_t[numSamples]; 2032 memset(buffer, 0, numSamples * sizeof(int16_t)); 2033 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2034 ownsBuffer = true; 2035 } 2036 2037 // Attach all tracks with same session ID to this chain. 2038 for (size_t i = 0; i < mTracks.size(); ++i) { 2039 sp<Track> track = mTracks[i]; 2040 if (session == track->sessionId()) { 2041 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2042 buffer); 2043 track->setMainBuffer(buffer); 2044 chain->incTrackCnt(); 2045 } 2046 } 2047 2048 // indicate all active tracks in the chain 2049 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2050 sp<Track> track = mActiveTracks[i].promote(); 2051 if (track == 0) { 2052 continue; 2053 } 2054 if (session == track->sessionId()) { 2055 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2056 chain->incActiveTrackCnt(); 2057 } 2058 } 2059 } 2060 2061 chain->setInBuffer(buffer, ownsBuffer); 2062 chain->setOutBuffer(mMixBuffer); 2063 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2064 // chains list in order to be processed last as it contains output stage effects 2065 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2066 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2067 // after track specific effects and before output stage 2068 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2069 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2070 // Effect chain for other sessions are inserted at beginning of effect 2071 // chains list to be processed before output mix effects. Relative order between other 2072 // sessions is not important 2073 size_t size = mEffectChains.size(); 2074 size_t i = 0; 2075 for (i = 0; i < size; i++) { 2076 if (mEffectChains[i]->sessionId() < session) { 2077 break; 2078 } 2079 } 2080 mEffectChains.insertAt(chain, i); 2081 checkSuspendOnAddEffectChain_l(chain); 2082 2083 return NO_ERROR; 2084} 2085 2086size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2087{ 2088 int session = chain->sessionId(); 2089 2090 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2091 2092 for (size_t i = 0; i < mEffectChains.size(); i++) { 2093 if (chain == mEffectChains[i]) { 2094 mEffectChains.removeAt(i); 2095 // detach all active tracks from the chain 2096 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2097 sp<Track> track = mActiveTracks[i].promote(); 2098 if (track == 0) { 2099 continue; 2100 } 2101 if (session == track->sessionId()) { 2102 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2103 chain.get(), session); 2104 chain->decActiveTrackCnt(); 2105 } 2106 } 2107 2108 // detach all tracks with same session ID from this chain 2109 for (size_t i = 0; i < mTracks.size(); ++i) { 2110 sp<Track> track = mTracks[i]; 2111 if (session == track->sessionId()) { 2112 track->setMainBuffer(mMixBuffer); 2113 chain->decTrackCnt(); 2114 } 2115 } 2116 break; 2117 } 2118 } 2119 return mEffectChains.size(); 2120} 2121 2122status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2123 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2124{ 2125 Mutex::Autolock _l(mLock); 2126 return attachAuxEffect_l(track, EffectId); 2127} 2128 2129status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2130 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2131{ 2132 status_t status = NO_ERROR; 2133 2134 if (EffectId == 0) { 2135 track->setAuxBuffer(0, NULL); 2136 } else { 2137 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2138 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2139 if (effect != 0) { 2140 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2141 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2142 } else { 2143 status = INVALID_OPERATION; 2144 } 2145 } else { 2146 status = BAD_VALUE; 2147 } 2148 } 2149 return status; 2150} 2151 2152void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2153{ 2154 for (size_t i = 0; i < mTracks.size(); ++i) { 2155 sp<Track> track = mTracks[i]; 2156 if (track->auxEffectId() == effectId) { 2157 attachAuxEffect_l(track, 0); 2158 } 2159 } 2160} 2161 2162bool AudioFlinger::PlaybackThread::threadLoop() 2163{ 2164 Vector< sp<Track> > tracksToRemove; 2165 2166 standbyTime = systemTime(); 2167 2168 // MIXER 2169 nsecs_t lastWarning = 0; 2170 2171 // DUPLICATING 2172 // FIXME could this be made local to while loop? 2173 writeFrames = 0; 2174 2175 int lastGeneration = 0; 2176 2177 cacheParameters_l(); 2178 sleepTime = idleSleepTime; 2179 2180 if (mType == MIXER) { 2181 sleepTimeShift = 0; 2182 } 2183 2184 CpuStats cpuStats; 2185 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2186 2187 acquireWakeLock(); 2188 2189 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2190 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2191 // and then that string will be logged at the next convenient opportunity. 2192 const char *logString = NULL; 2193 2194 checkSilentMode_l(); 2195 2196 while (!exitPending()) 2197 { 2198 cpuStats.sample(myName); 2199 2200 Vector< sp<EffectChain> > effectChains; 2201 2202 processConfigEvents(); 2203 2204 { // scope for mLock 2205 2206 Mutex::Autolock _l(mLock); 2207 2208 if (logString != NULL) { 2209 mNBLogWriter->logTimestamp(); 2210 mNBLogWriter->log(logString); 2211 logString = NULL; 2212 } 2213 2214 if (mLatchDValid) { 2215 mLatchQ = mLatchD; 2216 mLatchDValid = false; 2217 mLatchQValid = true; 2218 } 2219 2220 if (checkForNewParameters_l()) { 2221 cacheParameters_l(); 2222 } 2223 2224 saveOutputTracks(); 2225 if (mSignalPending) { 2226 // A signal was raised while we were unlocked 2227 mSignalPending = false; 2228 } else if (waitingAsyncCallback_l()) { 2229 if (exitPending()) { 2230 break; 2231 } 2232 releaseWakeLock_l(); 2233 mWakeLockUids.clear(); 2234 mActiveTracksGeneration++; 2235 ALOGV("wait async completion"); 2236 mWaitWorkCV.wait(mLock); 2237 ALOGV("async completion/wake"); 2238 acquireWakeLock_l(); 2239 standbyTime = systemTime() + standbyDelay; 2240 sleepTime = 0; 2241 2242 continue; 2243 } 2244 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2245 isSuspended()) { 2246 // put audio hardware into standby after short delay 2247 if (shouldStandby_l()) { 2248 2249 threadLoop_standby(); 2250 2251 mStandby = true; 2252 } 2253 2254 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2255 // we're about to wait, flush the binder command buffer 2256 IPCThreadState::self()->flushCommands(); 2257 2258 clearOutputTracks(); 2259 2260 if (exitPending()) { 2261 break; 2262 } 2263 2264 releaseWakeLock_l(); 2265 mWakeLockUids.clear(); 2266 mActiveTracksGeneration++; 2267 // wait until we have something to do... 2268 ALOGV("%s going to sleep", myName.string()); 2269 mWaitWorkCV.wait(mLock); 2270 ALOGV("%s waking up", myName.string()); 2271 acquireWakeLock_l(); 2272 2273 mMixerStatus = MIXER_IDLE; 2274 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2275 mBytesWritten = 0; 2276 mBytesRemaining = 0; 2277 checkSilentMode_l(); 2278 2279 standbyTime = systemTime() + standbyDelay; 2280 sleepTime = idleSleepTime; 2281 if (mType == MIXER) { 2282 sleepTimeShift = 0; 2283 } 2284 2285 continue; 2286 } 2287 } 2288 // mMixerStatusIgnoringFastTracks is also updated internally 2289 mMixerStatus = prepareTracks_l(&tracksToRemove); 2290 2291 // compare with previously applied list 2292 if (lastGeneration != mActiveTracksGeneration) { 2293 // update wakelock 2294 updateWakeLockUids_l(mWakeLockUids); 2295 lastGeneration = mActiveTracksGeneration; 2296 } 2297 2298 // prevent any changes in effect chain list and in each effect chain 2299 // during mixing and effect process as the audio buffers could be deleted 2300 // or modified if an effect is created or deleted 2301 lockEffectChains_l(effectChains); 2302 } // mLock scope ends 2303 2304 if (mBytesRemaining == 0) { 2305 mCurrentWriteLength = 0; 2306 if (mMixerStatus == MIXER_TRACKS_READY) { 2307 // threadLoop_mix() sets mCurrentWriteLength 2308 threadLoop_mix(); 2309 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2310 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2311 // threadLoop_sleepTime sets sleepTime to 0 if data 2312 // must be written to HAL 2313 threadLoop_sleepTime(); 2314 if (sleepTime == 0) { 2315 mCurrentWriteLength = mixBufferSize; 2316 } 2317 } 2318 mBytesRemaining = mCurrentWriteLength; 2319 if (isSuspended()) { 2320 sleepTime = suspendSleepTimeUs(); 2321 // simulate write to HAL when suspended 2322 mBytesWritten += mixBufferSize; 2323 mBytesRemaining = 0; 2324 } 2325 2326 // only process effects if we're going to write 2327 if (sleepTime == 0 && mType != OFFLOAD) { 2328 for (size_t i = 0; i < effectChains.size(); i ++) { 2329 effectChains[i]->process_l(); 2330 } 2331 } 2332 } 2333 // Process effect chains for offloaded thread even if no audio 2334 // was read from audio track: process only updates effect state 2335 // and thus does have to be synchronized with audio writes but may have 2336 // to be called while waiting for async write callback 2337 if (mType == OFFLOAD) { 2338 for (size_t i = 0; i < effectChains.size(); i ++) { 2339 effectChains[i]->process_l(); 2340 } 2341 } 2342 2343 // enable changes in effect chain 2344 unlockEffectChains(effectChains); 2345 2346 if (!waitingAsyncCallback()) { 2347 // sleepTime == 0 means we must write to audio hardware 2348 if (sleepTime == 0) { 2349 if (mBytesRemaining) { 2350 ssize_t ret = threadLoop_write(); 2351 if (ret < 0) { 2352 mBytesRemaining = 0; 2353 } else { 2354 mBytesWritten += ret; 2355 mBytesRemaining -= ret; 2356 } 2357 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2358 (mMixerStatus == MIXER_DRAIN_ALL)) { 2359 threadLoop_drain(); 2360 } 2361if (mType == MIXER) { 2362 // write blocked detection 2363 nsecs_t now = systemTime(); 2364 nsecs_t delta = now - mLastWriteTime; 2365 if (!mStandby && delta > maxPeriod) { 2366 mNumDelayedWrites++; 2367 if ((now - lastWarning) > kWarningThrottleNs) { 2368 ATRACE_NAME("underrun"); 2369 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2370 ns2ms(delta), mNumDelayedWrites, this); 2371 lastWarning = now; 2372 } 2373 } 2374} 2375 2376 } else { 2377 usleep(sleepTime); 2378 } 2379 } 2380 2381 // Finally let go of removed track(s), without the lock held 2382 // since we can't guarantee the destructors won't acquire that 2383 // same lock. This will also mutate and push a new fast mixer state. 2384 threadLoop_removeTracks(tracksToRemove); 2385 tracksToRemove.clear(); 2386 2387 // FIXME I don't understand the need for this here; 2388 // it was in the original code but maybe the 2389 // assignment in saveOutputTracks() makes this unnecessary? 2390 clearOutputTracks(); 2391 2392 // Effect chains will be actually deleted here if they were removed from 2393 // mEffectChains list during mixing or effects processing 2394 effectChains.clear(); 2395 2396 // FIXME Note that the above .clear() is no longer necessary since effectChains 2397 // is now local to this block, but will keep it for now (at least until merge done). 2398 } 2399 2400 threadLoop_exit(); 2401 2402 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2403 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2404 // put output stream into standby mode 2405 if (!mStandby) { 2406 mOutput->stream->common.standby(&mOutput->stream->common); 2407 } 2408 } 2409 2410 releaseWakeLock(); 2411 mWakeLockUids.clear(); 2412 mActiveTracksGeneration++; 2413 2414 ALOGV("Thread %p type %d exiting", this, mType); 2415 return false; 2416} 2417 2418// removeTracks_l() must be called with ThreadBase::mLock held 2419void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2420{ 2421 size_t count = tracksToRemove.size(); 2422 if (count > 0) { 2423 for (size_t i=0 ; i<count ; i++) { 2424 const sp<Track>& track = tracksToRemove.itemAt(i); 2425 mActiveTracks.remove(track); 2426 mWakeLockUids.remove(track->uid()); 2427 mActiveTracksGeneration++; 2428 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2429 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2430 if (chain != 0) { 2431 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2432 track->sessionId()); 2433 chain->decActiveTrackCnt(); 2434 } 2435 if (track->isTerminated()) { 2436 removeTrack_l(track); 2437 } 2438 } 2439 } 2440 2441} 2442 2443status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2444{ 2445 if (mNormalSink != 0) { 2446 return mNormalSink->getTimestamp(timestamp); 2447 } 2448 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2449 uint64_t position64; 2450 int ret = mOutput->stream->get_presentation_position( 2451 mOutput->stream, &position64, ×tamp.mTime); 2452 if (ret == 0) { 2453 timestamp.mPosition = (uint32_t)position64; 2454 return NO_ERROR; 2455 } 2456 } 2457 return INVALID_OPERATION; 2458} 2459// ---------------------------------------------------------------------------- 2460 2461AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2462 audio_io_handle_t id, audio_devices_t device, type_t type) 2463 : PlaybackThread(audioFlinger, output, id, device, type), 2464 // mAudioMixer below 2465 // mFastMixer below 2466 mFastMixerFutex(0) 2467 // mOutputSink below 2468 // mPipeSink below 2469 // mNormalSink below 2470{ 2471 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2472 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2473 "mFrameCount=%d, mNormalFrameCount=%d", 2474 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2475 mNormalFrameCount); 2476 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2477 2478 // FIXME - Current mixer implementation only supports stereo output 2479 if (mChannelCount != FCC_2) { 2480 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2481 } 2482 2483 // create an NBAIO sink for the HAL output stream, and negotiate 2484 mOutputSink = new AudioStreamOutSink(output->stream); 2485 size_t numCounterOffers = 0; 2486 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2487 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2488 ALOG_ASSERT(index == 0); 2489 2490 // initialize fast mixer depending on configuration 2491 bool initFastMixer; 2492 switch (kUseFastMixer) { 2493 case FastMixer_Never: 2494 initFastMixer = false; 2495 break; 2496 case FastMixer_Always: 2497 initFastMixer = true; 2498 break; 2499 case FastMixer_Static: 2500 case FastMixer_Dynamic: 2501 initFastMixer = mFrameCount < mNormalFrameCount; 2502 break; 2503 } 2504 if (initFastMixer) { 2505 2506 // create a MonoPipe to connect our submix to FastMixer 2507 NBAIO_Format format = mOutputSink->format(); 2508 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2509 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2510 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2511 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2512 const NBAIO_Format offers[1] = {format}; 2513 size_t numCounterOffers = 0; 2514 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2515 ALOG_ASSERT(index == 0); 2516 monoPipe->setAvgFrames((mScreenState & 1) ? 2517 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2518 mPipeSink = monoPipe; 2519 2520#ifdef TEE_SINK 2521 if (mTeeSinkOutputEnabled) { 2522 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2523 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2524 numCounterOffers = 0; 2525 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2526 ALOG_ASSERT(index == 0); 2527 mTeeSink = teeSink; 2528 PipeReader *teeSource = new PipeReader(*teeSink); 2529 numCounterOffers = 0; 2530 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2531 ALOG_ASSERT(index == 0); 2532 mTeeSource = teeSource; 2533 } 2534#endif 2535 2536 // create fast mixer and configure it initially with just one fast track for our submix 2537 mFastMixer = new FastMixer(); 2538 FastMixerStateQueue *sq = mFastMixer->sq(); 2539#ifdef STATE_QUEUE_DUMP 2540 sq->setObserverDump(&mStateQueueObserverDump); 2541 sq->setMutatorDump(&mStateQueueMutatorDump); 2542#endif 2543 FastMixerState *state = sq->begin(); 2544 FastTrack *fastTrack = &state->mFastTracks[0]; 2545 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2546 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2547 fastTrack->mVolumeProvider = NULL; 2548 fastTrack->mGeneration++; 2549 state->mFastTracksGen++; 2550 state->mTrackMask = 1; 2551 // fast mixer will use the HAL output sink 2552 state->mOutputSink = mOutputSink.get(); 2553 state->mOutputSinkGen++; 2554 state->mFrameCount = mFrameCount; 2555 state->mCommand = FastMixerState::COLD_IDLE; 2556 // already done in constructor initialization list 2557 //mFastMixerFutex = 0; 2558 state->mColdFutexAddr = &mFastMixerFutex; 2559 state->mColdGen++; 2560 state->mDumpState = &mFastMixerDumpState; 2561#ifdef TEE_SINK 2562 state->mTeeSink = mTeeSink.get(); 2563#endif 2564 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2565 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2566 sq->end(); 2567 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2568 2569 // start the fast mixer 2570 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2571 pid_t tid = mFastMixer->getTid(); 2572 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2573 if (err != 0) { 2574 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2575 kPriorityFastMixer, getpid_cached, tid, err); 2576 } 2577 2578#ifdef AUDIO_WATCHDOG 2579 // create and start the watchdog 2580 mAudioWatchdog = new AudioWatchdog(); 2581 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2582 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2583 tid = mAudioWatchdog->getTid(); 2584 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2585 if (err != 0) { 2586 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2587 kPriorityFastMixer, getpid_cached, tid, err); 2588 } 2589#endif 2590 2591 } else { 2592 mFastMixer = NULL; 2593 } 2594 2595 switch (kUseFastMixer) { 2596 case FastMixer_Never: 2597 case FastMixer_Dynamic: 2598 mNormalSink = mOutputSink; 2599 break; 2600 case FastMixer_Always: 2601 mNormalSink = mPipeSink; 2602 break; 2603 case FastMixer_Static: 2604 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2605 break; 2606 } 2607} 2608 2609AudioFlinger::MixerThread::~MixerThread() 2610{ 2611 if (mFastMixer != NULL) { 2612 FastMixerStateQueue *sq = mFastMixer->sq(); 2613 FastMixerState *state = sq->begin(); 2614 if (state->mCommand == FastMixerState::COLD_IDLE) { 2615 int32_t old = android_atomic_inc(&mFastMixerFutex); 2616 if (old == -1) { 2617 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2618 } 2619 } 2620 state->mCommand = FastMixerState::EXIT; 2621 sq->end(); 2622 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2623 mFastMixer->join(); 2624 // Though the fast mixer thread has exited, it's state queue is still valid. 2625 // We'll use that extract the final state which contains one remaining fast track 2626 // corresponding to our sub-mix. 2627 state = sq->begin(); 2628 ALOG_ASSERT(state->mTrackMask == 1); 2629 FastTrack *fastTrack = &state->mFastTracks[0]; 2630 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2631 delete fastTrack->mBufferProvider; 2632 sq->end(false /*didModify*/); 2633 delete mFastMixer; 2634#ifdef AUDIO_WATCHDOG 2635 if (mAudioWatchdog != 0) { 2636 mAudioWatchdog->requestExit(); 2637 mAudioWatchdog->requestExitAndWait(); 2638 mAudioWatchdog.clear(); 2639 } 2640#endif 2641 } 2642 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2643 delete mAudioMixer; 2644} 2645 2646 2647uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2648{ 2649 if (mFastMixer != NULL) { 2650 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2651 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2652 } 2653 return latency; 2654} 2655 2656 2657void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2658{ 2659 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2660} 2661 2662ssize_t AudioFlinger::MixerThread::threadLoop_write() 2663{ 2664 // FIXME we should only do one push per cycle; confirm this is true 2665 // Start the fast mixer if it's not already running 2666 if (mFastMixer != NULL) { 2667 FastMixerStateQueue *sq = mFastMixer->sq(); 2668 FastMixerState *state = sq->begin(); 2669 if (state->mCommand != FastMixerState::MIX_WRITE && 2670 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2671 if (state->mCommand == FastMixerState::COLD_IDLE) { 2672 int32_t old = android_atomic_inc(&mFastMixerFutex); 2673 if (old == -1) { 2674 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2675 } 2676#ifdef AUDIO_WATCHDOG 2677 if (mAudioWatchdog != 0) { 2678 mAudioWatchdog->resume(); 2679 } 2680#endif 2681 } 2682 state->mCommand = FastMixerState::MIX_WRITE; 2683 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2684 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2685 sq->end(); 2686 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2687 if (kUseFastMixer == FastMixer_Dynamic) { 2688 mNormalSink = mPipeSink; 2689 } 2690 } else { 2691 sq->end(false /*didModify*/); 2692 } 2693 } 2694 return PlaybackThread::threadLoop_write(); 2695} 2696 2697void AudioFlinger::MixerThread::threadLoop_standby() 2698{ 2699 // Idle the fast mixer if it's currently running 2700 if (mFastMixer != NULL) { 2701 FastMixerStateQueue *sq = mFastMixer->sq(); 2702 FastMixerState *state = sq->begin(); 2703 if (!(state->mCommand & FastMixerState::IDLE)) { 2704 state->mCommand = FastMixerState::COLD_IDLE; 2705 state->mColdFutexAddr = &mFastMixerFutex; 2706 state->mColdGen++; 2707 mFastMixerFutex = 0; 2708 sq->end(); 2709 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2710 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2711 if (kUseFastMixer == FastMixer_Dynamic) { 2712 mNormalSink = mOutputSink; 2713 } 2714#ifdef AUDIO_WATCHDOG 2715 if (mAudioWatchdog != 0) { 2716 mAudioWatchdog->pause(); 2717 } 2718#endif 2719 } else { 2720 sq->end(false /*didModify*/); 2721 } 2722 } 2723 PlaybackThread::threadLoop_standby(); 2724} 2725 2726// Empty implementation for standard mixer 2727// Overridden for offloaded playback 2728void AudioFlinger::PlaybackThread::flushOutput_l() 2729{ 2730} 2731 2732bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2733{ 2734 return false; 2735} 2736 2737bool AudioFlinger::PlaybackThread::shouldStandby_l() 2738{ 2739 return !mStandby; 2740} 2741 2742bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2743{ 2744 Mutex::Autolock _l(mLock); 2745 return waitingAsyncCallback_l(); 2746} 2747 2748// shared by MIXER and DIRECT, overridden by DUPLICATING 2749void AudioFlinger::PlaybackThread::threadLoop_standby() 2750{ 2751 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2752 mOutput->stream->common.standby(&mOutput->stream->common); 2753 if (mUseAsyncWrite != 0) { 2754 // discard any pending drain or write ack by incrementing sequence 2755 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2756 mDrainSequence = (mDrainSequence + 2) & ~1; 2757 ALOG_ASSERT(mCallbackThread != 0); 2758 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2759 mCallbackThread->setDraining(mDrainSequence); 2760 } 2761} 2762 2763void AudioFlinger::MixerThread::threadLoop_mix() 2764{ 2765 // obtain the presentation timestamp of the next output buffer 2766 int64_t pts; 2767 status_t status = INVALID_OPERATION; 2768 2769 if (mNormalSink != 0) { 2770 status = mNormalSink->getNextWriteTimestamp(&pts); 2771 } else { 2772 status = mOutputSink->getNextWriteTimestamp(&pts); 2773 } 2774 2775 if (status != NO_ERROR) { 2776 pts = AudioBufferProvider::kInvalidPTS; 2777 } 2778 2779 // mix buffers... 2780 mAudioMixer->process(pts); 2781 mCurrentWriteLength = mixBufferSize; 2782 // increase sleep time progressively when application underrun condition clears. 2783 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2784 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2785 // such that we would underrun the audio HAL. 2786 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2787 sleepTimeShift--; 2788 } 2789 sleepTime = 0; 2790 standbyTime = systemTime() + standbyDelay; 2791 //TODO: delay standby when effects have a tail 2792} 2793 2794void AudioFlinger::MixerThread::threadLoop_sleepTime() 2795{ 2796 // If no tracks are ready, sleep once for the duration of an output 2797 // buffer size, then write 0s to the output 2798 if (sleepTime == 0) { 2799 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2800 sleepTime = activeSleepTime >> sleepTimeShift; 2801 if (sleepTime < kMinThreadSleepTimeUs) { 2802 sleepTime = kMinThreadSleepTimeUs; 2803 } 2804 // reduce sleep time in case of consecutive application underruns to avoid 2805 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2806 // duration we would end up writing less data than needed by the audio HAL if 2807 // the condition persists. 2808 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2809 sleepTimeShift++; 2810 } 2811 } else { 2812 sleepTime = idleSleepTime; 2813 } 2814 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2815 memset(mMixBuffer, 0, mixBufferSize); 2816 sleepTime = 0; 2817 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2818 "anticipated start"); 2819 } 2820 // TODO add standby time extension fct of effect tail 2821} 2822 2823// prepareTracks_l() must be called with ThreadBase::mLock held 2824AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2825 Vector< sp<Track> > *tracksToRemove) 2826{ 2827 2828 mixer_state mixerStatus = MIXER_IDLE; 2829 // find out which tracks need to be processed 2830 size_t count = mActiveTracks.size(); 2831 size_t mixedTracks = 0; 2832 size_t tracksWithEffect = 0; 2833 // counts only _active_ fast tracks 2834 size_t fastTracks = 0; 2835 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2836 2837 float masterVolume = mMasterVolume; 2838 bool masterMute = mMasterMute; 2839 2840 if (masterMute) { 2841 masterVolume = 0; 2842 } 2843 // Delegate master volume control to effect in output mix effect chain if needed 2844 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2845 if (chain != 0) { 2846 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2847 chain->setVolume_l(&v, &v); 2848 masterVolume = (float)((v + (1 << 23)) >> 24); 2849 chain.clear(); 2850 } 2851 2852 // prepare a new state to push 2853 FastMixerStateQueue *sq = NULL; 2854 FastMixerState *state = NULL; 2855 bool didModify = false; 2856 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2857 if (mFastMixer != NULL) { 2858 sq = mFastMixer->sq(); 2859 state = sq->begin(); 2860 } 2861 2862 for (size_t i=0 ; i<count ; i++) { 2863 const sp<Track> t = mActiveTracks[i].promote(); 2864 if (t == 0) { 2865 continue; 2866 } 2867 2868 // this const just means the local variable doesn't change 2869 Track* const track = t.get(); 2870 2871 // process fast tracks 2872 if (track->isFastTrack()) { 2873 2874 // It's theoretically possible (though unlikely) for a fast track to be created 2875 // and then removed within the same normal mix cycle. This is not a problem, as 2876 // the track never becomes active so it's fast mixer slot is never touched. 2877 // The converse, of removing an (active) track and then creating a new track 2878 // at the identical fast mixer slot within the same normal mix cycle, 2879 // is impossible because the slot isn't marked available until the end of each cycle. 2880 int j = track->mFastIndex; 2881 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2882 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2883 FastTrack *fastTrack = &state->mFastTracks[j]; 2884 2885 // Determine whether the track is currently in underrun condition, 2886 // and whether it had a recent underrun. 2887 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2888 FastTrackUnderruns underruns = ftDump->mUnderruns; 2889 uint32_t recentFull = (underruns.mBitFields.mFull - 2890 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2891 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2892 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2893 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2894 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2895 uint32_t recentUnderruns = recentPartial + recentEmpty; 2896 track->mObservedUnderruns = underruns; 2897 // don't count underruns that occur while stopping or pausing 2898 // or stopped which can occur when flush() is called while active 2899 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2900 recentUnderruns > 0) { 2901 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2902 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2903 } 2904 2905 // This is similar to the state machine for normal tracks, 2906 // with a few modifications for fast tracks. 2907 bool isActive = true; 2908 switch (track->mState) { 2909 case TrackBase::STOPPING_1: 2910 // track stays active in STOPPING_1 state until first underrun 2911 if (recentUnderruns > 0 || track->isTerminated()) { 2912 track->mState = TrackBase::STOPPING_2; 2913 } 2914 break; 2915 case TrackBase::PAUSING: 2916 // ramp down is not yet implemented 2917 track->setPaused(); 2918 break; 2919 case TrackBase::RESUMING: 2920 // ramp up is not yet implemented 2921 track->mState = TrackBase::ACTIVE; 2922 break; 2923 case TrackBase::ACTIVE: 2924 if (recentFull > 0 || recentPartial > 0) { 2925 // track has provided at least some frames recently: reset retry count 2926 track->mRetryCount = kMaxTrackRetries; 2927 } 2928 if (recentUnderruns == 0) { 2929 // no recent underruns: stay active 2930 break; 2931 } 2932 // there has recently been an underrun of some kind 2933 if (track->sharedBuffer() == 0) { 2934 // were any of the recent underruns "empty" (no frames available)? 2935 if (recentEmpty == 0) { 2936 // no, then ignore the partial underruns as they are allowed indefinitely 2937 break; 2938 } 2939 // there has recently been an "empty" underrun: decrement the retry counter 2940 if (--(track->mRetryCount) > 0) { 2941 break; 2942 } 2943 // indicate to client process that the track was disabled because of underrun; 2944 // it will then automatically call start() when data is available 2945 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2946 // remove from active list, but state remains ACTIVE [confusing but true] 2947 isActive = false; 2948 break; 2949 } 2950 // fall through 2951 case TrackBase::STOPPING_2: 2952 case TrackBase::PAUSED: 2953 case TrackBase::STOPPED: 2954 case TrackBase::FLUSHED: // flush() while active 2955 // Check for presentation complete if track is inactive 2956 // We have consumed all the buffers of this track. 2957 // This would be incomplete if we auto-paused on underrun 2958 { 2959 size_t audioHALFrames = 2960 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2961 size_t framesWritten = mBytesWritten / mFrameSize; 2962 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2963 // track stays in active list until presentation is complete 2964 break; 2965 } 2966 } 2967 if (track->isStopping_2()) { 2968 track->mState = TrackBase::STOPPED; 2969 } 2970 if (track->isStopped()) { 2971 // Can't reset directly, as fast mixer is still polling this track 2972 // track->reset(); 2973 // So instead mark this track as needing to be reset after push with ack 2974 resetMask |= 1 << i; 2975 } 2976 isActive = false; 2977 break; 2978 case TrackBase::IDLE: 2979 default: 2980 LOG_FATAL("unexpected track state %d", track->mState); 2981 } 2982 2983 if (isActive) { 2984 // was it previously inactive? 2985 if (!(state->mTrackMask & (1 << j))) { 2986 ExtendedAudioBufferProvider *eabp = track; 2987 VolumeProvider *vp = track; 2988 fastTrack->mBufferProvider = eabp; 2989 fastTrack->mVolumeProvider = vp; 2990 fastTrack->mSampleRate = track->mSampleRate; 2991 fastTrack->mChannelMask = track->mChannelMask; 2992 fastTrack->mGeneration++; 2993 state->mTrackMask |= 1 << j; 2994 didModify = true; 2995 // no acknowledgement required for newly active tracks 2996 } 2997 // cache the combined master volume and stream type volume for fast mixer; this 2998 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2999 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3000 ++fastTracks; 3001 } else { 3002 // was it previously active? 3003 if (state->mTrackMask & (1 << j)) { 3004 fastTrack->mBufferProvider = NULL; 3005 fastTrack->mGeneration++; 3006 state->mTrackMask &= ~(1 << j); 3007 didModify = true; 3008 // If any fast tracks were removed, we must wait for acknowledgement 3009 // because we're about to decrement the last sp<> on those tracks. 3010 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3011 } else { 3012 LOG_FATAL("fast track %d should have been active", j); 3013 } 3014 tracksToRemove->add(track); 3015 // Avoids a misleading display in dumpsys 3016 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3017 } 3018 continue; 3019 } 3020 3021 { // local variable scope to avoid goto warning 3022 3023 audio_track_cblk_t* cblk = track->cblk(); 3024 3025 // The first time a track is added we wait 3026 // for all its buffers to be filled before processing it 3027 int name = track->name(); 3028 // make sure that we have enough frames to mix one full buffer. 3029 // enforce this condition only once to enable draining the buffer in case the client 3030 // app does not call stop() and relies on underrun to stop: 3031 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3032 // during last round 3033 size_t desiredFrames; 3034 uint32_t sr = track->sampleRate(); 3035 if (sr == mSampleRate) { 3036 desiredFrames = mNormalFrameCount; 3037 } else { 3038 // +1 for rounding and +1 for additional sample needed for interpolation 3039 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3040 // add frames already consumed but not yet released by the resampler 3041 // because mAudioTrackServerProxy->framesReady() will include these frames 3042 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3043#if 0 3044 // the minimum track buffer size is normally twice the number of frames necessary 3045 // to fill one buffer and the resampler should not leave more than one buffer worth 3046 // of unreleased frames after each pass, but just in case... 3047 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3048#endif 3049 } 3050 uint32_t minFrames = 1; 3051 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3052 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3053 minFrames = desiredFrames; 3054 } 3055 3056 size_t framesReady = track->framesReady(); 3057 if ((framesReady >= minFrames) && track->isReady() && 3058 !track->isPaused() && !track->isTerminated()) 3059 { 3060 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3061 3062 mixedTracks++; 3063 3064 // track->mainBuffer() != mMixBuffer means there is an effect chain 3065 // connected to the track 3066 chain.clear(); 3067 if (track->mainBuffer() != mMixBuffer) { 3068 chain = getEffectChain_l(track->sessionId()); 3069 // Delegate volume control to effect in track effect chain if needed 3070 if (chain != 0) { 3071 tracksWithEffect++; 3072 } else { 3073 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3074 "session %d", 3075 name, track->sessionId()); 3076 } 3077 } 3078 3079 3080 int param = AudioMixer::VOLUME; 3081 if (track->mFillingUpStatus == Track::FS_FILLED) { 3082 // no ramp for the first volume setting 3083 track->mFillingUpStatus = Track::FS_ACTIVE; 3084 if (track->mState == TrackBase::RESUMING) { 3085 track->mState = TrackBase::ACTIVE; 3086 param = AudioMixer::RAMP_VOLUME; 3087 } 3088 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3089 // FIXME should not make a decision based on mServer 3090 } else if (cblk->mServer != 0) { 3091 // If the track is stopped before the first frame was mixed, 3092 // do not apply ramp 3093 param = AudioMixer::RAMP_VOLUME; 3094 } 3095 3096 // compute volume for this track 3097 uint32_t vl, vr, va; 3098 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3099 vl = vr = va = 0; 3100 if (track->isPausing()) { 3101 track->setPaused(); 3102 } 3103 } else { 3104 3105 // read original volumes with volume control 3106 float typeVolume = mStreamTypes[track->streamType()].volume; 3107 float v = masterVolume * typeVolume; 3108 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3109 uint32_t vlr = proxy->getVolumeLR(); 3110 vl = vlr & 0xFFFF; 3111 vr = vlr >> 16; 3112 // track volumes come from shared memory, so can't be trusted and must be clamped 3113 if (vl > MAX_GAIN_INT) { 3114 ALOGV("Track left volume out of range: %04X", vl); 3115 vl = MAX_GAIN_INT; 3116 } 3117 if (vr > MAX_GAIN_INT) { 3118 ALOGV("Track right volume out of range: %04X", vr); 3119 vr = MAX_GAIN_INT; 3120 } 3121 // now apply the master volume and stream type volume 3122 vl = (uint32_t)(v * vl) << 12; 3123 vr = (uint32_t)(v * vr) << 12; 3124 // assuming master volume and stream type volume each go up to 1.0, 3125 // vl and vr are now in 8.24 format 3126 3127 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3128 // send level comes from shared memory and so may be corrupt 3129 if (sendLevel > MAX_GAIN_INT) { 3130 ALOGV("Track send level out of range: %04X", sendLevel); 3131 sendLevel = MAX_GAIN_INT; 3132 } 3133 va = (uint32_t)(v * sendLevel); 3134 } 3135 3136 // Delegate volume control to effect in track effect chain if needed 3137 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3138 // Do not ramp volume if volume is controlled by effect 3139 param = AudioMixer::VOLUME; 3140 track->mHasVolumeController = true; 3141 } else { 3142 // force no volume ramp when volume controller was just disabled or removed 3143 // from effect chain to avoid volume spike 3144 if (track->mHasVolumeController) { 3145 param = AudioMixer::VOLUME; 3146 } 3147 track->mHasVolumeController = false; 3148 } 3149 3150 // Convert volumes from 8.24 to 4.12 format 3151 // This additional clamping is needed in case chain->setVolume_l() overshot 3152 vl = (vl + (1 << 11)) >> 12; 3153 if (vl > MAX_GAIN_INT) { 3154 vl = MAX_GAIN_INT; 3155 } 3156 vr = (vr + (1 << 11)) >> 12; 3157 if (vr > MAX_GAIN_INT) { 3158 vr = MAX_GAIN_INT; 3159 } 3160 3161 if (va > MAX_GAIN_INT) { 3162 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3163 } 3164 3165 // XXX: these things DON'T need to be done each time 3166 mAudioMixer->setBufferProvider(name, track); 3167 mAudioMixer->enable(name); 3168 3169 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3170 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3171 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3172 mAudioMixer->setParameter( 3173 name, 3174 AudioMixer::TRACK, 3175 AudioMixer::FORMAT, (void *)track->format()); 3176 mAudioMixer->setParameter( 3177 name, 3178 AudioMixer::TRACK, 3179 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3180 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3181 uint32_t maxSampleRate = mSampleRate * 2; 3182 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3183 if (reqSampleRate == 0) { 3184 reqSampleRate = mSampleRate; 3185 } else if (reqSampleRate > maxSampleRate) { 3186 reqSampleRate = maxSampleRate; 3187 } 3188 mAudioMixer->setParameter( 3189 name, 3190 AudioMixer::RESAMPLE, 3191 AudioMixer::SAMPLE_RATE, 3192 (void *)reqSampleRate); 3193 mAudioMixer->setParameter( 3194 name, 3195 AudioMixer::TRACK, 3196 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3197 mAudioMixer->setParameter( 3198 name, 3199 AudioMixer::TRACK, 3200 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3201 3202 // reset retry count 3203 track->mRetryCount = kMaxTrackRetries; 3204 3205 // If one track is ready, set the mixer ready if: 3206 // - the mixer was not ready during previous round OR 3207 // - no other track is not ready 3208 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3209 mixerStatus != MIXER_TRACKS_ENABLED) { 3210 mixerStatus = MIXER_TRACKS_READY; 3211 } 3212 } else { 3213 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3214 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3215 } 3216 // clear effect chain input buffer if an active track underruns to avoid sending 3217 // previous audio buffer again to effects 3218 chain = getEffectChain_l(track->sessionId()); 3219 if (chain != 0) { 3220 chain->clearInputBuffer(); 3221 } 3222 3223 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3224 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3225 track->isStopped() || track->isPaused()) { 3226 // We have consumed all the buffers of this track. 3227 // Remove it from the list of active tracks. 3228 // TODO: use actual buffer filling status instead of latency when available from 3229 // audio HAL 3230 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3231 size_t framesWritten = mBytesWritten / mFrameSize; 3232 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3233 if (track->isStopped()) { 3234 track->reset(); 3235 } 3236 tracksToRemove->add(track); 3237 } 3238 } else { 3239 // No buffers for this track. Give it a few chances to 3240 // fill a buffer, then remove it from active list. 3241 if (--(track->mRetryCount) <= 0) { 3242 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3243 tracksToRemove->add(track); 3244 // indicate to client process that the track was disabled because of underrun; 3245 // it will then automatically call start() when data is available 3246 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3247 // If one track is not ready, mark the mixer also not ready if: 3248 // - the mixer was ready during previous round OR 3249 // - no other track is ready 3250 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3251 mixerStatus != MIXER_TRACKS_READY) { 3252 mixerStatus = MIXER_TRACKS_ENABLED; 3253 } 3254 } 3255 mAudioMixer->disable(name); 3256 } 3257 3258 } // local variable scope to avoid goto warning 3259track_is_ready: ; 3260 3261 } 3262 3263 // Push the new FastMixer state if necessary 3264 bool pauseAudioWatchdog = false; 3265 if (didModify) { 3266 state->mFastTracksGen++; 3267 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3268 if (kUseFastMixer == FastMixer_Dynamic && 3269 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3270 state->mCommand = FastMixerState::COLD_IDLE; 3271 state->mColdFutexAddr = &mFastMixerFutex; 3272 state->mColdGen++; 3273 mFastMixerFutex = 0; 3274 if (kUseFastMixer == FastMixer_Dynamic) { 3275 mNormalSink = mOutputSink; 3276 } 3277 // If we go into cold idle, need to wait for acknowledgement 3278 // so that fast mixer stops doing I/O. 3279 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3280 pauseAudioWatchdog = true; 3281 } 3282 } 3283 if (sq != NULL) { 3284 sq->end(didModify); 3285 sq->push(block); 3286 } 3287#ifdef AUDIO_WATCHDOG 3288 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3289 mAudioWatchdog->pause(); 3290 } 3291#endif 3292 3293 // Now perform the deferred reset on fast tracks that have stopped 3294 while (resetMask != 0) { 3295 size_t i = __builtin_ctz(resetMask); 3296 ALOG_ASSERT(i < count); 3297 resetMask &= ~(1 << i); 3298 sp<Track> t = mActiveTracks[i].promote(); 3299 if (t == 0) { 3300 continue; 3301 } 3302 Track* track = t.get(); 3303 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3304 track->reset(); 3305 } 3306 3307 // remove all the tracks that need to be... 3308 removeTracks_l(*tracksToRemove); 3309 3310 // mix buffer must be cleared if all tracks are connected to an 3311 // effect chain as in this case the mixer will not write to 3312 // mix buffer and track effects will accumulate into it 3313 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3314 (mixedTracks == 0 && fastTracks > 0))) { 3315 // FIXME as a performance optimization, should remember previous zero status 3316 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3317 } 3318 3319 // if any fast tracks, then status is ready 3320 mMixerStatusIgnoringFastTracks = mixerStatus; 3321 if (fastTracks > 0) { 3322 mixerStatus = MIXER_TRACKS_READY; 3323 } 3324 return mixerStatus; 3325} 3326 3327// getTrackName_l() must be called with ThreadBase::mLock held 3328int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3329{ 3330 return mAudioMixer->getTrackName(channelMask, sessionId); 3331} 3332 3333// deleteTrackName_l() must be called with ThreadBase::mLock held 3334void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3335{ 3336 ALOGV("remove track (%d) and delete from mixer", name); 3337 mAudioMixer->deleteTrackName(name); 3338} 3339 3340// checkForNewParameters_l() must be called with ThreadBase::mLock held 3341bool AudioFlinger::MixerThread::checkForNewParameters_l() 3342{ 3343 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3344 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3345 bool reconfig = false; 3346 3347 while (!mNewParameters.isEmpty()) { 3348 3349 if (mFastMixer != NULL) { 3350 FastMixerStateQueue *sq = mFastMixer->sq(); 3351 FastMixerState *state = sq->begin(); 3352 if (!(state->mCommand & FastMixerState::IDLE)) { 3353 previousCommand = state->mCommand; 3354 state->mCommand = FastMixerState::HOT_IDLE; 3355 sq->end(); 3356 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3357 } else { 3358 sq->end(false /*didModify*/); 3359 } 3360 } 3361 3362 status_t status = NO_ERROR; 3363 String8 keyValuePair = mNewParameters[0]; 3364 AudioParameter param = AudioParameter(keyValuePair); 3365 int value; 3366 3367 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3368 reconfig = true; 3369 } 3370 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3371 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3372 status = BAD_VALUE; 3373 } else { 3374 // no need to save value, since it's constant 3375 reconfig = true; 3376 } 3377 } 3378 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3379 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3380 status = BAD_VALUE; 3381 } else { 3382 // no need to save value, since it's constant 3383 reconfig = true; 3384 } 3385 } 3386 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3387 // do not accept frame count changes if tracks are open as the track buffer 3388 // size depends on frame count and correct behavior would not be guaranteed 3389 // if frame count is changed after track creation 3390 if (!mTracks.isEmpty()) { 3391 status = INVALID_OPERATION; 3392 } else { 3393 reconfig = true; 3394 } 3395 } 3396 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3397#ifdef ADD_BATTERY_DATA 3398 // when changing the audio output device, call addBatteryData to notify 3399 // the change 3400 if (mOutDevice != value) { 3401 uint32_t params = 0; 3402 // check whether speaker is on 3403 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3404 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3405 } 3406 3407 audio_devices_t deviceWithoutSpeaker 3408 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3409 // check if any other device (except speaker) is on 3410 if (value & deviceWithoutSpeaker ) { 3411 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3412 } 3413 3414 if (params != 0) { 3415 addBatteryData(params); 3416 } 3417 } 3418#endif 3419 3420 // forward device change to effects that have requested to be 3421 // aware of attached audio device. 3422 if (value != AUDIO_DEVICE_NONE) { 3423 mOutDevice = value; 3424 for (size_t i = 0; i < mEffectChains.size(); i++) { 3425 mEffectChains[i]->setDevice_l(mOutDevice); 3426 } 3427 } 3428 } 3429 3430 if (status == NO_ERROR) { 3431 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3432 keyValuePair.string()); 3433 if (!mStandby && status == INVALID_OPERATION) { 3434 mOutput->stream->common.standby(&mOutput->stream->common); 3435 mStandby = true; 3436 mBytesWritten = 0; 3437 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3438 keyValuePair.string()); 3439 } 3440 if (status == NO_ERROR && reconfig) { 3441 readOutputParameters(); 3442 delete mAudioMixer; 3443 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3444 for (size_t i = 0; i < mTracks.size() ; i++) { 3445 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3446 if (name < 0) { 3447 break; 3448 } 3449 mTracks[i]->mName = name; 3450 } 3451 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3452 } 3453 } 3454 3455 mNewParameters.removeAt(0); 3456 3457 mParamStatus = status; 3458 mParamCond.signal(); 3459 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3460 // already timed out waiting for the status and will never signal the condition. 3461 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3462 } 3463 3464 if (!(previousCommand & FastMixerState::IDLE)) { 3465 ALOG_ASSERT(mFastMixer != NULL); 3466 FastMixerStateQueue *sq = mFastMixer->sq(); 3467 FastMixerState *state = sq->begin(); 3468 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3469 state->mCommand = previousCommand; 3470 sq->end(); 3471 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3472 } 3473 3474 return reconfig; 3475} 3476 3477 3478void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3479{ 3480 const size_t SIZE = 256; 3481 char buffer[SIZE]; 3482 String8 result; 3483 3484 PlaybackThread::dumpInternals(fd, args); 3485 3486 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3487 result.append(buffer); 3488 write(fd, result.string(), result.size()); 3489 3490 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3491 const FastMixerDumpState copy(mFastMixerDumpState); 3492 copy.dump(fd); 3493 3494#ifdef STATE_QUEUE_DUMP 3495 // Similar for state queue 3496 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3497 observerCopy.dump(fd); 3498 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3499 mutatorCopy.dump(fd); 3500#endif 3501 3502#ifdef TEE_SINK 3503 // Write the tee output to a .wav file 3504 dumpTee(fd, mTeeSource, mId); 3505#endif 3506 3507#ifdef AUDIO_WATCHDOG 3508 if (mAudioWatchdog != 0) { 3509 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3510 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3511 wdCopy.dump(fd); 3512 } 3513#endif 3514} 3515 3516uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3517{ 3518 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3519} 3520 3521uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3522{ 3523 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3524} 3525 3526void AudioFlinger::MixerThread::cacheParameters_l() 3527{ 3528 PlaybackThread::cacheParameters_l(); 3529 3530 // FIXME: Relaxed timing because of a certain device that can't meet latency 3531 // Should be reduced to 2x after the vendor fixes the driver issue 3532 // increase threshold again due to low power audio mode. The way this warning 3533 // threshold is calculated and its usefulness should be reconsidered anyway. 3534 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3535} 3536 3537// ---------------------------------------------------------------------------- 3538 3539AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3540 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3541 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3542 // mLeftVolFloat, mRightVolFloat 3543{ 3544} 3545 3546AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3547 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3548 ThreadBase::type_t type) 3549 : PlaybackThread(audioFlinger, output, id, device, type) 3550 // mLeftVolFloat, mRightVolFloat 3551{ 3552} 3553 3554AudioFlinger::DirectOutputThread::~DirectOutputThread() 3555{ 3556} 3557 3558void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3559{ 3560 audio_track_cblk_t* cblk = track->cblk(); 3561 float left, right; 3562 3563 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3564 left = right = 0; 3565 } else { 3566 float typeVolume = mStreamTypes[track->streamType()].volume; 3567 float v = mMasterVolume * typeVolume; 3568 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3569 uint32_t vlr = proxy->getVolumeLR(); 3570 float v_clamped = v * (vlr & 0xFFFF); 3571 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3572 left = v_clamped/MAX_GAIN; 3573 v_clamped = v * (vlr >> 16); 3574 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3575 right = v_clamped/MAX_GAIN; 3576 } 3577 3578 if (lastTrack) { 3579 if (left != mLeftVolFloat || right != mRightVolFloat) { 3580 mLeftVolFloat = left; 3581 mRightVolFloat = right; 3582 3583 // Convert volumes from float to 8.24 3584 uint32_t vl = (uint32_t)(left * (1 << 24)); 3585 uint32_t vr = (uint32_t)(right * (1 << 24)); 3586 3587 // Delegate volume control to effect in track effect chain if needed 3588 // only one effect chain can be present on DirectOutputThread, so if 3589 // there is one, the track is connected to it 3590 if (!mEffectChains.isEmpty()) { 3591 mEffectChains[0]->setVolume_l(&vl, &vr); 3592 left = (float)vl / (1 << 24); 3593 right = (float)vr / (1 << 24); 3594 } 3595 if (mOutput->stream->set_volume) { 3596 mOutput->stream->set_volume(mOutput->stream, left, right); 3597 } 3598 } 3599 } 3600} 3601 3602 3603AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3604 Vector< sp<Track> > *tracksToRemove 3605) 3606{ 3607 size_t count = mActiveTracks.size(); 3608 mixer_state mixerStatus = MIXER_IDLE; 3609 3610 // find out which tracks need to be processed 3611 for (size_t i = 0; i < count; i++) { 3612 sp<Track> t = mActiveTracks[i].promote(); 3613 // The track died recently 3614 if (t == 0) { 3615 continue; 3616 } 3617 3618 Track* const track = t.get(); 3619 audio_track_cblk_t* cblk = track->cblk(); 3620 // Only consider last track started for volume and mixer state control. 3621 // In theory an older track could underrun and restart after the new one starts 3622 // but as we only care about the transition phase between two tracks on a 3623 // direct output, it is not a problem to ignore the underrun case. 3624 sp<Track> l = mLatestActiveTrack.promote(); 3625 bool last = l.get() == track; 3626 3627 // The first time a track is added we wait 3628 // for all its buffers to be filled before processing it 3629 uint32_t minFrames; 3630 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3631 minFrames = mNormalFrameCount; 3632 } else { 3633 minFrames = 1; 3634 } 3635 3636 if ((track->framesReady() >= minFrames) && track->isReady() && 3637 !track->isPaused() && !track->isTerminated()) 3638 { 3639 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3640 3641 if (track->mFillingUpStatus == Track::FS_FILLED) { 3642 track->mFillingUpStatus = Track::FS_ACTIVE; 3643 // make sure processVolume_l() will apply new volume even if 0 3644 mLeftVolFloat = mRightVolFloat = -1.0; 3645 if (track->mState == TrackBase::RESUMING) { 3646 track->mState = TrackBase::ACTIVE; 3647 } 3648 } 3649 3650 // compute volume for this track 3651 processVolume_l(track, last); 3652 if (last) { 3653 // reset retry count 3654 track->mRetryCount = kMaxTrackRetriesDirect; 3655 mActiveTrack = t; 3656 mixerStatus = MIXER_TRACKS_READY; 3657 } 3658 } else { 3659 // clear effect chain input buffer if the last active track started underruns 3660 // to avoid sending previous audio buffer again to effects 3661 if (!mEffectChains.isEmpty() && last) { 3662 mEffectChains[0]->clearInputBuffer(); 3663 } 3664 3665 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3666 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3667 track->isStopped() || track->isPaused()) { 3668 // We have consumed all the buffers of this track. 3669 // Remove it from the list of active tracks. 3670 // TODO: implement behavior for compressed audio 3671 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3672 size_t framesWritten = mBytesWritten / mFrameSize; 3673 if (mStandby || !last || 3674 track->presentationComplete(framesWritten, audioHALFrames)) { 3675 if (track->isStopped()) { 3676 track->reset(); 3677 } 3678 tracksToRemove->add(track); 3679 } 3680 } else { 3681 // No buffers for this track. Give it a few chances to 3682 // fill a buffer, then remove it from active list. 3683 // Only consider last track started for mixer state control 3684 if (--(track->mRetryCount) <= 0) { 3685 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3686 tracksToRemove->add(track); 3687 // indicate to client process that the track was disabled because of underrun; 3688 // it will then automatically call start() when data is available 3689 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3690 } else if (last) { 3691 mixerStatus = MIXER_TRACKS_ENABLED; 3692 } 3693 } 3694 } 3695 } 3696 3697 // remove all the tracks that need to be... 3698 removeTracks_l(*tracksToRemove); 3699 3700 return mixerStatus; 3701} 3702 3703void AudioFlinger::DirectOutputThread::threadLoop_mix() 3704{ 3705 size_t frameCount = mFrameCount; 3706 int8_t *curBuf = (int8_t *)mMixBuffer; 3707 // output audio to hardware 3708 while (frameCount) { 3709 AudioBufferProvider::Buffer buffer; 3710 buffer.frameCount = frameCount; 3711 mActiveTrack->getNextBuffer(&buffer); 3712 if (buffer.raw == NULL) { 3713 memset(curBuf, 0, frameCount * mFrameSize); 3714 break; 3715 } 3716 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3717 frameCount -= buffer.frameCount; 3718 curBuf += buffer.frameCount * mFrameSize; 3719 mActiveTrack->releaseBuffer(&buffer); 3720 } 3721 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3722 sleepTime = 0; 3723 standbyTime = systemTime() + standbyDelay; 3724 mActiveTrack.clear(); 3725} 3726 3727void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3728{ 3729 if (sleepTime == 0) { 3730 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3731 sleepTime = activeSleepTime; 3732 } else { 3733 sleepTime = idleSleepTime; 3734 } 3735 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3736 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3737 sleepTime = 0; 3738 } 3739} 3740 3741// getTrackName_l() must be called with ThreadBase::mLock held 3742int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3743 int sessionId) 3744{ 3745 return 0; 3746} 3747 3748// deleteTrackName_l() must be called with ThreadBase::mLock held 3749void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3750{ 3751} 3752 3753// checkForNewParameters_l() must be called with ThreadBase::mLock held 3754bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3755{ 3756 bool reconfig = false; 3757 3758 while (!mNewParameters.isEmpty()) { 3759 status_t status = NO_ERROR; 3760 String8 keyValuePair = mNewParameters[0]; 3761 AudioParameter param = AudioParameter(keyValuePair); 3762 int value; 3763 3764 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3765 // do not accept frame count changes if tracks are open as the track buffer 3766 // size depends on frame count and correct behavior would not be garantied 3767 // if frame count is changed after track creation 3768 if (!mTracks.isEmpty()) { 3769 status = INVALID_OPERATION; 3770 } else { 3771 reconfig = true; 3772 } 3773 } 3774 if (status == NO_ERROR) { 3775 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3776 keyValuePair.string()); 3777 if (!mStandby && status == INVALID_OPERATION) { 3778 mOutput->stream->common.standby(&mOutput->stream->common); 3779 mStandby = true; 3780 mBytesWritten = 0; 3781 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3782 keyValuePair.string()); 3783 } 3784 if (status == NO_ERROR && reconfig) { 3785 readOutputParameters(); 3786 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3787 } 3788 } 3789 3790 mNewParameters.removeAt(0); 3791 3792 mParamStatus = status; 3793 mParamCond.signal(); 3794 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3795 // already timed out waiting for the status and will never signal the condition. 3796 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3797 } 3798 return reconfig; 3799} 3800 3801uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3802{ 3803 uint32_t time; 3804 if (audio_is_linear_pcm(mFormat)) { 3805 time = PlaybackThread::activeSleepTimeUs(); 3806 } else { 3807 time = 10000; 3808 } 3809 return time; 3810} 3811 3812uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3813{ 3814 uint32_t time; 3815 if (audio_is_linear_pcm(mFormat)) { 3816 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3817 } else { 3818 time = 10000; 3819 } 3820 return time; 3821} 3822 3823uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3824{ 3825 uint32_t time; 3826 if (audio_is_linear_pcm(mFormat)) { 3827 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3828 } else { 3829 time = 10000; 3830 } 3831 return time; 3832} 3833 3834void AudioFlinger::DirectOutputThread::cacheParameters_l() 3835{ 3836 PlaybackThread::cacheParameters_l(); 3837 3838 // use shorter standby delay as on normal output to release 3839 // hardware resources as soon as possible 3840 if (audio_is_linear_pcm(mFormat)) { 3841 standbyDelay = microseconds(activeSleepTime*2); 3842 } else { 3843 standbyDelay = kOffloadStandbyDelayNs; 3844 } 3845} 3846 3847// ---------------------------------------------------------------------------- 3848 3849AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3850 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3851 : Thread(false /*canCallJava*/), 3852 mPlaybackThread(playbackThread), 3853 mWriteAckSequence(0), 3854 mDrainSequence(0) 3855{ 3856} 3857 3858AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3859{ 3860} 3861 3862void AudioFlinger::AsyncCallbackThread::onFirstRef() 3863{ 3864 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3865} 3866 3867bool AudioFlinger::AsyncCallbackThread::threadLoop() 3868{ 3869 while (!exitPending()) { 3870 uint32_t writeAckSequence; 3871 uint32_t drainSequence; 3872 3873 { 3874 Mutex::Autolock _l(mLock); 3875 while (!((mWriteAckSequence & 1) || 3876 (mDrainSequence & 1) || 3877 exitPending())) { 3878 mWaitWorkCV.wait(mLock); 3879 } 3880 3881 if (exitPending()) { 3882 break; 3883 } 3884 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3885 mWriteAckSequence, mDrainSequence); 3886 writeAckSequence = mWriteAckSequence; 3887 mWriteAckSequence &= ~1; 3888 drainSequence = mDrainSequence; 3889 mDrainSequence &= ~1; 3890 } 3891 { 3892 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3893 if (playbackThread != 0) { 3894 if (writeAckSequence & 1) { 3895 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3896 } 3897 if (drainSequence & 1) { 3898 playbackThread->resetDraining(drainSequence >> 1); 3899 } 3900 } 3901 } 3902 } 3903 return false; 3904} 3905 3906void AudioFlinger::AsyncCallbackThread::exit() 3907{ 3908 ALOGV("AsyncCallbackThread::exit"); 3909 Mutex::Autolock _l(mLock); 3910 requestExit(); 3911 mWaitWorkCV.broadcast(); 3912} 3913 3914void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3915{ 3916 Mutex::Autolock _l(mLock); 3917 // bit 0 is cleared 3918 mWriteAckSequence = sequence << 1; 3919} 3920 3921void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3922{ 3923 Mutex::Autolock _l(mLock); 3924 // ignore unexpected callbacks 3925 if (mWriteAckSequence & 2) { 3926 mWriteAckSequence |= 1; 3927 mWaitWorkCV.signal(); 3928 } 3929} 3930 3931void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3932{ 3933 Mutex::Autolock _l(mLock); 3934 // bit 0 is cleared 3935 mDrainSequence = sequence << 1; 3936} 3937 3938void AudioFlinger::AsyncCallbackThread::resetDraining() 3939{ 3940 Mutex::Autolock _l(mLock); 3941 // ignore unexpected callbacks 3942 if (mDrainSequence & 2) { 3943 mDrainSequence |= 1; 3944 mWaitWorkCV.signal(); 3945 } 3946} 3947 3948 3949// ---------------------------------------------------------------------------- 3950AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3951 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3952 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3953 mHwPaused(false), 3954 mFlushPending(false), 3955 mPausedBytesRemaining(0) 3956{ 3957 //FIXME: mStandby should be set to true by ThreadBase constructor 3958 mStandby = true; 3959} 3960 3961void AudioFlinger::OffloadThread::threadLoop_exit() 3962{ 3963 if (mFlushPending || mHwPaused) { 3964 // If a flush is pending or track was paused, just discard buffered data 3965 flushHw_l(); 3966 } else { 3967 mMixerStatus = MIXER_DRAIN_ALL; 3968 threadLoop_drain(); 3969 } 3970 mCallbackThread->exit(); 3971 PlaybackThread::threadLoop_exit(); 3972} 3973 3974AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3975 Vector< sp<Track> > *tracksToRemove 3976) 3977{ 3978 size_t count = mActiveTracks.size(); 3979 3980 mixer_state mixerStatus = MIXER_IDLE; 3981 bool doHwPause = false; 3982 bool doHwResume = false; 3983 3984 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3985 3986 // find out which tracks need to be processed 3987 for (size_t i = 0; i < count; i++) { 3988 sp<Track> t = mActiveTracks[i].promote(); 3989 // The track died recently 3990 if (t == 0) { 3991 continue; 3992 } 3993 Track* const track = t.get(); 3994 audio_track_cblk_t* cblk = track->cblk(); 3995 // Only consider last track started for volume and mixer state control. 3996 // In theory an older track could underrun and restart after the new one starts 3997 // but as we only care about the transition phase between two tracks on a 3998 // direct output, it is not a problem to ignore the underrun case. 3999 sp<Track> l = mLatestActiveTrack.promote(); 4000 bool last = l.get() == track; 4001 4002 if (track->isPausing()) { 4003 track->setPaused(); 4004 if (last) { 4005 if (!mHwPaused) { 4006 doHwPause = true; 4007 mHwPaused = true; 4008 } 4009 // If we were part way through writing the mixbuffer to 4010 // the HAL we must save this until we resume 4011 // BUG - this will be wrong if a different track is made active, 4012 // in that case we want to discard the pending data in the 4013 // mixbuffer and tell the client to present it again when the 4014 // track is resumed 4015 mPausedWriteLength = mCurrentWriteLength; 4016 mPausedBytesRemaining = mBytesRemaining; 4017 mBytesRemaining = 0; // stop writing 4018 } 4019 tracksToRemove->add(track); 4020 } else if (track->framesReady() && track->isReady() && 4021 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4022 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4023 if (track->mFillingUpStatus == Track::FS_FILLED) { 4024 track->mFillingUpStatus = Track::FS_ACTIVE; 4025 // make sure processVolume_l() will apply new volume even if 0 4026 mLeftVolFloat = mRightVolFloat = -1.0; 4027 if (track->mState == TrackBase::RESUMING) { 4028 track->mState = TrackBase::ACTIVE; 4029 if (last) { 4030 if (mPausedBytesRemaining) { 4031 // Need to continue write that was interrupted 4032 mCurrentWriteLength = mPausedWriteLength; 4033 mBytesRemaining = mPausedBytesRemaining; 4034 mPausedBytesRemaining = 0; 4035 } 4036 if (mHwPaused) { 4037 doHwResume = true; 4038 mHwPaused = false; 4039 // threadLoop_mix() will handle the case that we need to 4040 // resume an interrupted write 4041 } 4042 // enable write to audio HAL 4043 sleepTime = 0; 4044 } 4045 } 4046 } 4047 4048 if (last) { 4049 sp<Track> previousTrack = mPreviousTrack.promote(); 4050 if (previousTrack != 0) { 4051 if (track != previousTrack.get()) { 4052 // Flush any data still being written from last track 4053 mBytesRemaining = 0; 4054 if (mPausedBytesRemaining) { 4055 // Last track was paused so we also need to flush saved 4056 // mixbuffer state and invalidate track so that it will 4057 // re-submit that unwritten data when it is next resumed 4058 mPausedBytesRemaining = 0; 4059 // Invalidate is a bit drastic - would be more efficient 4060 // to have a flag to tell client that some of the 4061 // previously written data was lost 4062 previousTrack->invalidate(); 4063 } 4064 // flush data already sent to the DSP if changing audio session as audio 4065 // comes from a different source. Also invalidate previous track to force a 4066 // seek when resuming. 4067 if (previousTrack->sessionId() != track->sessionId()) { 4068 previousTrack->invalidate(); 4069 mFlushPending = true; 4070 } 4071 } 4072 } 4073 mPreviousTrack = track; 4074 // reset retry count 4075 track->mRetryCount = kMaxTrackRetriesOffload; 4076 mActiveTrack = t; 4077 mixerStatus = MIXER_TRACKS_READY; 4078 } 4079 } else { 4080 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4081 if (track->isStopping_1()) { 4082 // Hardware buffer can hold a large amount of audio so we must 4083 // wait for all current track's data to drain before we say 4084 // that the track is stopped. 4085 if (mBytesRemaining == 0) { 4086 // Only start draining when all data in mixbuffer 4087 // has been written 4088 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4089 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4090 // do not drain if no data was ever sent to HAL (mStandby == true) 4091 if (last && !mStandby) { 4092 // do not modify drain sequence if we are already draining. This happens 4093 // when resuming from pause after drain. 4094 if ((mDrainSequence & 1) == 0) { 4095 sleepTime = 0; 4096 standbyTime = systemTime() + standbyDelay; 4097 mixerStatus = MIXER_DRAIN_TRACK; 4098 mDrainSequence += 2; 4099 } 4100 if (mHwPaused) { 4101 // It is possible to move from PAUSED to STOPPING_1 without 4102 // a resume so we must ensure hardware is running 4103 doHwResume = true; 4104 mHwPaused = false; 4105 } 4106 } 4107 } 4108 } else if (track->isStopping_2()) { 4109 // Drain has completed or we are in standby, signal presentation complete 4110 if (!(mDrainSequence & 1) || !last || mStandby) { 4111 track->mState = TrackBase::STOPPED; 4112 size_t audioHALFrames = 4113 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4114 size_t framesWritten = 4115 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4116 track->presentationComplete(framesWritten, audioHALFrames); 4117 track->reset(); 4118 tracksToRemove->add(track); 4119 } 4120 } else { 4121 // No buffers for this track. Give it a few chances to 4122 // fill a buffer, then remove it from active list. 4123 if (--(track->mRetryCount) <= 0) { 4124 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4125 track->name()); 4126 tracksToRemove->add(track); 4127 // indicate to client process that the track was disabled because of underrun; 4128 // it will then automatically call start() when data is available 4129 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4130 } else if (last){ 4131 mixerStatus = MIXER_TRACKS_ENABLED; 4132 } 4133 } 4134 } 4135 // compute volume for this track 4136 processVolume_l(track, last); 4137 } 4138 4139 // make sure the pause/flush/resume sequence is executed in the right order. 4140 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4141 // before flush and then resume HW. This can happen in case of pause/flush/resume 4142 // if resume is received before pause is executed. 4143 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4144 mOutput->stream->pause(mOutput->stream); 4145 if (!doHwPause) { 4146 doHwResume = true; 4147 } 4148 } 4149 if (mFlushPending) { 4150 flushHw_l(); 4151 mFlushPending = false; 4152 } 4153 if (!mStandby && doHwResume) { 4154 mOutput->stream->resume(mOutput->stream); 4155 } 4156 4157 // remove all the tracks that need to be... 4158 removeTracks_l(*tracksToRemove); 4159 4160 return mixerStatus; 4161} 4162 4163void AudioFlinger::OffloadThread::flushOutput_l() 4164{ 4165 mFlushPending = true; 4166} 4167 4168// must be called with thread mutex locked 4169bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4170{ 4171 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4172 mWriteAckSequence, mDrainSequence); 4173 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4174 return true; 4175 } 4176 return false; 4177} 4178 4179// must be called with thread mutex locked 4180bool AudioFlinger::OffloadThread::shouldStandby_l() 4181{ 4182 bool trackPaused = false; 4183 4184 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4185 // after a timeout and we will enter standby then. 4186 if (mTracks.size() > 0) { 4187 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4188 } 4189 4190 return !mStandby && !trackPaused; 4191} 4192 4193 4194bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4195{ 4196 Mutex::Autolock _l(mLock); 4197 return waitingAsyncCallback_l(); 4198} 4199 4200void AudioFlinger::OffloadThread::flushHw_l() 4201{ 4202 mOutput->stream->flush(mOutput->stream); 4203 // Flush anything still waiting in the mixbuffer 4204 mCurrentWriteLength = 0; 4205 mBytesRemaining = 0; 4206 mPausedWriteLength = 0; 4207 mPausedBytesRemaining = 0; 4208 if (mUseAsyncWrite) { 4209 // discard any pending drain or write ack by incrementing sequence 4210 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4211 mDrainSequence = (mDrainSequence + 2) & ~1; 4212 ALOG_ASSERT(mCallbackThread != 0); 4213 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4214 mCallbackThread->setDraining(mDrainSequence); 4215 } 4216} 4217 4218// ---------------------------------------------------------------------------- 4219 4220AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4221 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4222 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4223 DUPLICATING), 4224 mWaitTimeMs(UINT_MAX) 4225{ 4226 addOutputTrack(mainThread); 4227} 4228 4229AudioFlinger::DuplicatingThread::~DuplicatingThread() 4230{ 4231 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4232 mOutputTracks[i]->destroy(); 4233 } 4234} 4235 4236void AudioFlinger::DuplicatingThread::threadLoop_mix() 4237{ 4238 // mix buffers... 4239 if (outputsReady(outputTracks)) { 4240 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4241 } else { 4242 memset(mMixBuffer, 0, mixBufferSize); 4243 } 4244 sleepTime = 0; 4245 writeFrames = mNormalFrameCount; 4246 mCurrentWriteLength = mixBufferSize; 4247 standbyTime = systemTime() + standbyDelay; 4248} 4249 4250void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4251{ 4252 if (sleepTime == 0) { 4253 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4254 sleepTime = activeSleepTime; 4255 } else { 4256 sleepTime = idleSleepTime; 4257 } 4258 } else if (mBytesWritten != 0) { 4259 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4260 writeFrames = mNormalFrameCount; 4261 memset(mMixBuffer, 0, mixBufferSize); 4262 } else { 4263 // flush remaining overflow buffers in output tracks 4264 writeFrames = 0; 4265 } 4266 sleepTime = 0; 4267 } 4268} 4269 4270ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4271{ 4272 for (size_t i = 0; i < outputTracks.size(); i++) { 4273 outputTracks[i]->write(mMixBuffer, writeFrames); 4274 } 4275 mStandby = false; 4276 return (ssize_t)mixBufferSize; 4277} 4278 4279void AudioFlinger::DuplicatingThread::threadLoop_standby() 4280{ 4281 // DuplicatingThread implements standby by stopping all tracks 4282 for (size_t i = 0; i < outputTracks.size(); i++) { 4283 outputTracks[i]->stop(); 4284 } 4285} 4286 4287void AudioFlinger::DuplicatingThread::saveOutputTracks() 4288{ 4289 outputTracks = mOutputTracks; 4290} 4291 4292void AudioFlinger::DuplicatingThread::clearOutputTracks() 4293{ 4294 outputTracks.clear(); 4295} 4296 4297void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4298{ 4299 Mutex::Autolock _l(mLock); 4300 // FIXME explain this formula 4301 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4302 OutputTrack *outputTrack = new OutputTrack(thread, 4303 this, 4304 mSampleRate, 4305 mFormat, 4306 mChannelMask, 4307 frameCount, 4308 IPCThreadState::self()->getCallingUid()); 4309 if (outputTrack->cblk() != NULL) { 4310 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4311 mOutputTracks.add(outputTrack); 4312 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4313 updateWaitTime_l(); 4314 } 4315} 4316 4317void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4318{ 4319 Mutex::Autolock _l(mLock); 4320 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4321 if (mOutputTracks[i]->thread() == thread) { 4322 mOutputTracks[i]->destroy(); 4323 mOutputTracks.removeAt(i); 4324 updateWaitTime_l(); 4325 return; 4326 } 4327 } 4328 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4329} 4330 4331// caller must hold mLock 4332void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4333{ 4334 mWaitTimeMs = UINT_MAX; 4335 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4336 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4337 if (strong != 0) { 4338 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4339 if (waitTimeMs < mWaitTimeMs) { 4340 mWaitTimeMs = waitTimeMs; 4341 } 4342 } 4343 } 4344} 4345 4346 4347bool AudioFlinger::DuplicatingThread::outputsReady( 4348 const SortedVector< sp<OutputTrack> > &outputTracks) 4349{ 4350 for (size_t i = 0; i < outputTracks.size(); i++) { 4351 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4352 if (thread == 0) { 4353 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4354 outputTracks[i].get()); 4355 return false; 4356 } 4357 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4358 // see note at standby() declaration 4359 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4360 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4361 thread.get()); 4362 return false; 4363 } 4364 } 4365 return true; 4366} 4367 4368uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4369{ 4370 return (mWaitTimeMs * 1000) / 2; 4371} 4372 4373void AudioFlinger::DuplicatingThread::cacheParameters_l() 4374{ 4375 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4376 updateWaitTime_l(); 4377 4378 MixerThread::cacheParameters_l(); 4379} 4380 4381// ---------------------------------------------------------------------------- 4382// Record 4383// ---------------------------------------------------------------------------- 4384 4385AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4386 AudioStreamIn *input, 4387 uint32_t sampleRate, 4388 audio_channel_mask_t channelMask, 4389 audio_io_handle_t id, 4390 audio_devices_t outDevice, 4391 audio_devices_t inDevice 4392#ifdef TEE_SINK 4393 , const sp<NBAIO_Sink>& teeSink 4394#endif 4395 ) : 4396 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4397 mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4398 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4399 // are set by readInputParameters() 4400 // mRsmpInIndex LEGACY 4401 mReqChannelCount(popcount(channelMask)), 4402 mReqSampleRate(sampleRate) 4403 // mBytesRead is only meaningful while active, and so is cleared in start() 4404 // (but might be better to also clear here for dump?) 4405#ifdef TEE_SINK 4406 , mTeeSink(teeSink) 4407#endif 4408{ 4409 snprintf(mName, kNameLength, "AudioIn_%X", id); 4410 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4411 4412 readInputParameters(); 4413} 4414 4415 4416AudioFlinger::RecordThread::~RecordThread() 4417{ 4418 mAudioFlinger->unregisterWriter(mNBLogWriter); 4419 delete[] mRsmpInBuffer; 4420 delete mResampler; 4421 delete[] mRsmpOutBuffer; 4422} 4423 4424void AudioFlinger::RecordThread::onFirstRef() 4425{ 4426 run(mName, PRIORITY_URGENT_AUDIO); 4427} 4428 4429bool AudioFlinger::RecordThread::threadLoop() 4430{ 4431 nsecs_t lastWarning = 0; 4432 4433 inputStandBy(); 4434 4435 // used to verify we've read at least once before evaluating how many bytes were read 4436 bool readOnce = false; 4437 4438 // used to request a deferred sleep, to be executed later while mutex is unlocked 4439 bool doSleep = false; 4440 4441reacquire_wakelock: 4442 sp<RecordTrack> activeTrack; 4443 int activeTracksGen; 4444 { 4445 Mutex::Autolock _l(mLock); 4446 size_t size = mActiveTracks.size(); 4447 activeTracksGen = mActiveTracksGen; 4448 if (size > 0) { 4449 // FIXME an arbitrary choice 4450 activeTrack = mActiveTracks[0]; 4451 acquireWakeLock_l(activeTrack->uid()); 4452 if (size > 1) { 4453 SortedVector<int> tmp; 4454 for (size_t i = 0; i < size; i++) { 4455 tmp.add(mActiveTracks[i]->uid()); 4456 } 4457 updateWakeLockUids_l(tmp); 4458 } 4459 } else { 4460 acquireWakeLock_l(-1); 4461 } 4462 } 4463 4464 // start recording 4465 for (;;) { 4466 TrackBase::track_state activeTrackState; 4467 Vector< sp<EffectChain> > effectChains; 4468 4469 // sleep with mutex unlocked 4470 if (doSleep) { 4471 doSleep = false; 4472 usleep(kRecordThreadSleepUs); 4473 } 4474 4475 { // scope for mLock 4476 Mutex::Autolock _l(mLock); 4477 4478 processConfigEvents_l(); 4479 // return value 'reconfig' is currently unused 4480 bool reconfig = checkForNewParameters_l(); 4481 4482 // check exitPending here because checkForNewParameters_l() and 4483 // checkForNewParameters_l() can temporarily release mLock 4484 if (exitPending()) { 4485 break; 4486 } 4487 4488 // if no active track(s), then standby and release wakelock 4489 size_t size = mActiveTracks.size(); 4490 if (size == 0) { 4491 standbyIfNotAlreadyInStandby(); 4492 // exitPending() can't become true here 4493 releaseWakeLock_l(); 4494 ALOGV("RecordThread: loop stopping"); 4495 // go to sleep 4496 mWaitWorkCV.wait(mLock); 4497 ALOGV("RecordThread: loop starting"); 4498 goto reacquire_wakelock; 4499 } 4500 4501 if (mActiveTracksGen != activeTracksGen) { 4502 activeTracksGen = mActiveTracksGen; 4503 SortedVector<int> tmp; 4504 for (size_t i = 0; i < size; i++) { 4505 tmp.add(mActiveTracks[i]->uid()); 4506 } 4507 updateWakeLockUids_l(tmp); 4508 // FIXME an arbitrary choice 4509 activeTrack = mActiveTracks[0]; 4510 } 4511 4512 if (activeTrack->isTerminated()) { 4513 removeTrack_l(activeTrack); 4514 mActiveTracks.remove(activeTrack); 4515 mActiveTracksGen++; 4516 continue; 4517 } 4518 4519 activeTrackState = activeTrack->mState; 4520 switch (activeTrackState) { 4521 case TrackBase::PAUSING: 4522 standbyIfNotAlreadyInStandby(); 4523 mActiveTracks.remove(activeTrack); 4524 mActiveTracksGen++; 4525 mStartStopCond.broadcast(); 4526 doSleep = true; 4527 continue; 4528 4529 case TrackBase::RESUMING: 4530 mStandby = false; 4531 if (mReqChannelCount != activeTrack->channelCount()) { 4532 mActiveTracks.remove(activeTrack); 4533 mActiveTracksGen++; 4534 mStartStopCond.broadcast(); 4535 continue; 4536 } 4537 if (readOnce) { 4538 mStartStopCond.broadcast(); 4539 // record start succeeds only if first read from audio input succeeds 4540 if (mBytesRead < 0) { 4541 mActiveTracks.remove(activeTrack); 4542 mActiveTracksGen++; 4543 continue; 4544 } 4545 activeTrack->mState = TrackBase::ACTIVE; 4546 } 4547 break; 4548 4549 case TrackBase::ACTIVE: 4550 break; 4551 4552 case TrackBase::IDLE: 4553 doSleep = true; 4554 continue; 4555 4556 default: 4557 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4558 } 4559 4560 lockEffectChains_l(effectChains); 4561 } 4562 4563 // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable 4564 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4565 4566 for (size_t i = 0; i < effectChains.size(); i ++) { 4567 // thread mutex is not locked, but effect chain is locked 4568 effectChains[i]->process_l(); 4569 } 4570 4571 AudioBufferProvider::Buffer buffer; 4572 buffer.frameCount = mFrameCount; 4573 status_t status = activeTrack->getNextBuffer(&buffer); 4574 if (status == NO_ERROR) { 4575 readOnce = true; 4576 size_t framesOut = buffer.frameCount; 4577 if (mResampler == NULL) { 4578 // no resampling 4579 while (framesOut) { 4580 size_t framesIn = mFrameCount - mRsmpInIndex; 4581 if (framesIn > 0) { 4582 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4583 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4584 activeTrack->mFrameSize; 4585 if (framesIn > framesOut) { 4586 framesIn = framesOut; 4587 } 4588 mRsmpInIndex += framesIn; 4589 framesOut -= framesIn; 4590 if (mChannelCount == mReqChannelCount) { 4591 memcpy(dst, src, framesIn * mFrameSize); 4592 } else { 4593 if (mChannelCount == 1) { 4594 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4595 (int16_t *)src, framesIn); 4596 } else { 4597 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4598 (int16_t *)src, framesIn); 4599 } 4600 } 4601 } 4602 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4603 void *readInto; 4604 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4605 readInto = buffer.raw; 4606 framesOut = 0; 4607 } else { 4608 readInto = mRsmpInBuffer; 4609 mRsmpInIndex = 0; 4610 } 4611 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4612 mBufferSize); 4613 if (mBytesRead <= 0) { 4614 // TODO: verify that it's benign to use a stale track state 4615 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4616 { 4617 ALOGE("Error reading audio input"); 4618 // Force input into standby so that it tries to 4619 // recover at next read attempt 4620 inputStandBy(); 4621 doSleep = true; 4622 } 4623 mRsmpInIndex = mFrameCount; 4624 framesOut = 0; 4625 buffer.frameCount = 0; 4626 } 4627#ifdef TEE_SINK 4628 else if (mTeeSink != 0) { 4629 (void) mTeeSink->write(readInto, 4630 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4631 } 4632#endif 4633 } 4634 } 4635 } else { 4636 // resampling 4637 4638 // avoid busy-waiting if client doesn't keep up 4639 bool madeProgress = false; 4640 4641 // keep mRsmpInBuffer full so resampler always has sufficient input 4642 for (;;) { 4643 int32_t rear = mRsmpInRear; 4644 ssize_t filled = rear - mRsmpInFront; 4645 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4646 // exit once there is enough data in buffer for resampler 4647 if ((size_t) filled >= mRsmpInFrames) { 4648 break; 4649 } 4650 size_t avail = mRsmpInFramesP2 - filled; 4651 // Only try to read full HAL buffers. 4652 // But if the HAL read returns a partial buffer, use it. 4653 if (avail < mFrameCount) { 4654 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4655 avail, mFrameCount); 4656 break; 4657 } 4658 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4659 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4660 rear &= mRsmpInFramesP2 - 1; 4661 mBytesRead = mInput->stream->read(mInput->stream, 4662 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4663 if (mBytesRead <= 0) { 4664 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4665 break; 4666 } 4667 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4668 size_t framesRead = mBytesRead / mFrameSize; 4669 ALOG_ASSERT(framesRead > 0); 4670 madeProgress = true; 4671 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4672 size_t part1 = mRsmpInFramesP2 - rear; 4673 if (framesRead > part1) { 4674 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4675 (framesRead - part1) * mFrameSize); 4676 } 4677 mRsmpInRear += framesRead; 4678 } 4679 4680 if (!madeProgress) { 4681 ALOGV("Did not make progress"); 4682 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4683 } 4684 4685 // resampler accumulates, but we only have one source track 4686 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4687 mResampler->resample(mRsmpOutBuffer, framesOut, 4688 this /* AudioBufferProvider* */); 4689 // ditherAndClamp() works as long as all buffers returned by 4690 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4691 if (mReqChannelCount == 1) { 4692 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4693 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4694 // the resampler always outputs stereo samples: 4695 // do post stereo to mono conversion 4696 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4697 framesOut); 4698 } else { 4699 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4700 } 4701 // now done with mRsmpOutBuffer 4702 4703 } 4704 if (mFramestoDrop == 0) { 4705 activeTrack->releaseBuffer(&buffer); 4706 } else { 4707 if (mFramestoDrop > 0) { 4708 mFramestoDrop -= buffer.frameCount; 4709 if (mFramestoDrop <= 0) { 4710 clearSyncStartEvent(); 4711 } 4712 } else { 4713 mFramestoDrop += buffer.frameCount; 4714 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4715 mSyncStartEvent->isCancelled()) { 4716 ALOGW("Synced record %s, session %d, trigger session %d", 4717 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4718 activeTrack->sessionId(), 4719 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4720 clearSyncStartEvent(); 4721 } 4722 } 4723 } 4724 activeTrack->clearOverflow(); 4725 } 4726 // client isn't retrieving buffers fast enough 4727 else { 4728 if (!activeTrack->setOverflow()) { 4729 nsecs_t now = systemTime(); 4730 if ((now - lastWarning) > kWarningThrottleNs) { 4731 ALOGW("RecordThread: buffer overflow"); 4732 lastWarning = now; 4733 } 4734 } 4735 // Release the processor for a while before asking for a new buffer. 4736 // This will give the application more chance to read from the buffer and 4737 // clear the overflow. 4738 doSleep = true; 4739 } 4740 4741 // enable changes in effect chain 4742 unlockEffectChains(effectChains); 4743 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4744 } 4745 4746 standbyIfNotAlreadyInStandby(); 4747 4748 { 4749 Mutex::Autolock _l(mLock); 4750 for (size_t i = 0; i < mTracks.size(); i++) { 4751 sp<RecordTrack> track = mTracks[i]; 4752 track->invalidate(); 4753 } 4754 mActiveTracks.clear(); 4755 mActiveTracksGen++; 4756 mStartStopCond.broadcast(); 4757 } 4758 4759 releaseWakeLock(); 4760 4761 ALOGV("RecordThread %p exiting", this); 4762 return false; 4763} 4764 4765void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4766{ 4767 if (!mStandby) { 4768 inputStandBy(); 4769 mStandby = true; 4770 } 4771} 4772 4773void AudioFlinger::RecordThread::inputStandBy() 4774{ 4775 mInput->stream->common.standby(&mInput->stream->common); 4776} 4777 4778sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4779 const sp<AudioFlinger::Client>& client, 4780 uint32_t sampleRate, 4781 audio_format_t format, 4782 audio_channel_mask_t channelMask, 4783 size_t *pFrameCount, 4784 int sessionId, 4785 int uid, 4786 IAudioFlinger::track_flags_t *flags, 4787 pid_t tid, 4788 status_t *status) 4789{ 4790 size_t frameCount = *pFrameCount; 4791 sp<RecordTrack> track; 4792 status_t lStatus; 4793 4794 lStatus = initCheck(); 4795 if (lStatus != NO_ERROR) { 4796 ALOGE("createRecordTrack_l() audio driver not initialized"); 4797 goto Exit; 4798 } 4799 // client expresses a preference for FAST, but we get the final say 4800 if (*flags & IAudioFlinger::TRACK_FAST) { 4801 if ( 4802 // use case: callback handler and frame count is default or at least as large as HAL 4803 ( 4804 (tid != -1) && 4805 ((frameCount == 0) || 4806 (frameCount >= mFrameCount)) 4807 ) && 4808 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4809 // mono or stereo 4810 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4811 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4812 // hardware sample rate 4813 (sampleRate == mSampleRate) && 4814 // record thread has an associated fast recorder 4815 hasFastRecorder() 4816 // FIXME test that RecordThread for this fast track has a capable output HAL 4817 // FIXME add a permission test also? 4818 ) { 4819 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4820 if (frameCount == 0) { 4821 frameCount = mFrameCount * kFastTrackMultiplier; 4822 } 4823 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4824 frameCount, mFrameCount); 4825 } else { 4826 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4827 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4828 "hasFastRecorder=%d tid=%d", 4829 frameCount, mFrameCount, format, 4830 audio_is_linear_pcm(format), 4831 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4832 *flags &= ~IAudioFlinger::TRACK_FAST; 4833 // For compatibility with AudioRecord calculation, buffer depth is forced 4834 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4835 // This is probably too conservative, but legacy application code may depend on it. 4836 // If you change this calculation, also review the start threshold which is related. 4837 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4838 size_t mNormalFrameCount = 2048; // FIXME 4839 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4840 if (minBufCount < 2) { 4841 minBufCount = 2; 4842 } 4843 size_t minFrameCount = mNormalFrameCount * minBufCount; 4844 if (frameCount < minFrameCount) { 4845 frameCount = minFrameCount; 4846 } 4847 } 4848 } 4849 *pFrameCount = frameCount; 4850 4851 // FIXME use flags and tid similar to createTrack_l() 4852 4853 { // scope for mLock 4854 Mutex::Autolock _l(mLock); 4855 4856 track = new RecordTrack(this, client, sampleRate, 4857 format, channelMask, frameCount, sessionId, uid); 4858 4859 lStatus = track->initCheck(); 4860 if (lStatus != NO_ERROR) { 4861 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4862 track.clear(); 4863 goto Exit; 4864 } 4865 mTracks.add(track); 4866 4867 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4868 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4869 mAudioFlinger->btNrecIsOff(); 4870 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4871 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4872 4873 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4874 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4875 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4876 // so ask activity manager to do this on our behalf 4877 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4878 } 4879 } 4880 lStatus = NO_ERROR; 4881 4882Exit: 4883 *status = lStatus; 4884 return track; 4885} 4886 4887status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4888 AudioSystem::sync_event_t event, 4889 int triggerSession) 4890{ 4891 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4892 sp<ThreadBase> strongMe = this; 4893 status_t status = NO_ERROR; 4894 4895 if (event == AudioSystem::SYNC_EVENT_NONE) { 4896 clearSyncStartEvent(); 4897 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4898 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4899 triggerSession, 4900 recordTrack->sessionId(), 4901 syncStartEventCallback, 4902 this); 4903 // Sync event can be cancelled by the trigger session if the track is not in a 4904 // compatible state in which case we start record immediately 4905 if (mSyncStartEvent->isCancelled()) { 4906 clearSyncStartEvent(); 4907 } else { 4908 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4909 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4910 } 4911 } 4912 4913 { 4914 // This section is a rendezvous between binder thread executing start() and RecordThread 4915 AutoMutex lock(mLock); 4916 if (mActiveTracks.size() > 0) { 4917 // FIXME does not work for multiple active tracks 4918 if (mActiveTracks.indexOf(recordTrack) != 0) { 4919 status = -EBUSY; 4920 } else if (recordTrack->mState == TrackBase::PAUSING) { 4921 recordTrack->mState = TrackBase::ACTIVE; 4922 } 4923 return status; 4924 } 4925 4926 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4927 recordTrack->mState = TrackBase::IDLE; 4928 mActiveTracks.add(recordTrack); 4929 mActiveTracksGen++; 4930 mLock.unlock(); 4931 status_t status = AudioSystem::startInput(mId); 4932 mLock.lock(); 4933 // FIXME should verify that mActiveTrack is still == recordTrack 4934 if (status != NO_ERROR) { 4935 mActiveTracks.remove(recordTrack); 4936 mActiveTracksGen++; 4937 clearSyncStartEvent(); 4938 return status; 4939 } 4940 // FIXME LEGACY 4941 mRsmpInIndex = mFrameCount; 4942 mRsmpInFront = 0; 4943 mRsmpInRear = 0; 4944 mRsmpInUnrel = 0; 4945 mBytesRead = 0; 4946 if (mResampler != NULL) { 4947 mResampler->reset(); 4948 } 4949 // FIXME hijacking a playback track state name which was intended for start after pause; 4950 // here 'STARTING_2' would be more accurate 4951 recordTrack->mState = TrackBase::RESUMING; 4952 // signal thread to start 4953 ALOGV("Signal record thread"); 4954 mWaitWorkCV.broadcast(); 4955 // do not wait for mStartStopCond if exiting 4956 if (exitPending()) { 4957 mActiveTracks.remove(recordTrack); 4958 mActiveTracksGen++; 4959 status = INVALID_OPERATION; 4960 goto startError; 4961 } 4962 // FIXME incorrect usage of wait: no explicit predicate or loop 4963 mStartStopCond.wait(mLock); 4964 if (mActiveTracks.indexOf(recordTrack) < 0) { 4965 ALOGV("Record failed to start"); 4966 status = BAD_VALUE; 4967 goto startError; 4968 } 4969 ALOGV("Record started OK"); 4970 return status; 4971 } 4972 4973startError: 4974 AudioSystem::stopInput(mId); 4975 clearSyncStartEvent(); 4976 return status; 4977} 4978 4979void AudioFlinger::RecordThread::clearSyncStartEvent() 4980{ 4981 if (mSyncStartEvent != 0) { 4982 mSyncStartEvent->cancel(); 4983 } 4984 mSyncStartEvent.clear(); 4985 mFramestoDrop = 0; 4986} 4987 4988void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4989{ 4990 sp<SyncEvent> strongEvent = event.promote(); 4991 4992 if (strongEvent != 0) { 4993 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4994 me->handleSyncStartEvent(strongEvent); 4995 } 4996} 4997 4998void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4999{ 5000 if (event == mSyncStartEvent) { 5001 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5002 // from audio HAL 5003 mFramestoDrop = mFrameCount * 2; 5004 } 5005} 5006 5007bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5008 ALOGV("RecordThread::stop"); 5009 AutoMutex _l(mLock); 5010 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5011 return false; 5012 } 5013 // note that threadLoop may still be processing the track at this point [without lock] 5014 recordTrack->mState = TrackBase::PAUSING; 5015 // do not wait for mStartStopCond if exiting 5016 if (exitPending()) { 5017 return true; 5018 } 5019 // FIXME incorrect usage of wait: no explicit predicate or loop 5020 mStartStopCond.wait(mLock); 5021 // if we have been restarted, recordTrack is in mActiveTracks here 5022 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5023 ALOGV("Record stopped OK"); 5024 return true; 5025 } 5026 return false; 5027} 5028 5029bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 5030{ 5031 return false; 5032} 5033 5034status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5035{ 5036#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5037 if (!isValidSyncEvent(event)) { 5038 return BAD_VALUE; 5039 } 5040 5041 int eventSession = event->triggerSession(); 5042 status_t ret = NAME_NOT_FOUND; 5043 5044 Mutex::Autolock _l(mLock); 5045 5046 for (size_t i = 0; i < mTracks.size(); i++) { 5047 sp<RecordTrack> track = mTracks[i]; 5048 if (eventSession == track->sessionId()) { 5049 (void) track->setSyncEvent(event); 5050 ret = NO_ERROR; 5051 } 5052 } 5053 return ret; 5054#else 5055 return BAD_VALUE; 5056#endif 5057} 5058 5059// destroyTrack_l() must be called with ThreadBase::mLock held 5060void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5061{ 5062 track->terminate(); 5063 track->mState = TrackBase::STOPPED; 5064 // active tracks are removed by threadLoop() 5065 if (mActiveTracks.indexOf(track) < 0) { 5066 removeTrack_l(track); 5067 } 5068} 5069 5070void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5071{ 5072 mTracks.remove(track); 5073 // need anything related to effects here? 5074} 5075 5076void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5077{ 5078 dumpInternals(fd, args); 5079 dumpTracks(fd, args); 5080 dumpEffectChains(fd, args); 5081} 5082 5083void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5084{ 5085 const size_t SIZE = 256; 5086 char buffer[SIZE]; 5087 String8 result; 5088 5089 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5090 result.append(buffer); 5091 5092 if (mActiveTracks.size() > 0) { 5093 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5094 result.append(buffer); 5095 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5096 result.append(buffer); 5097 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5098 result.append(buffer); 5099 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5100 result.append(buffer); 5101 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5102 result.append(buffer); 5103 } else { 5104 result.append("No active record client\n"); 5105 } 5106 5107 write(fd, result.string(), result.size()); 5108 5109 dumpBase(fd, args); 5110} 5111 5112void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 5113{ 5114 const size_t SIZE = 256; 5115 char buffer[SIZE]; 5116 String8 result; 5117 5118 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5119 result.append(buffer); 5120 RecordTrack::appendDumpHeader(result); 5121 for (size_t i = 0; i < mTracks.size(); ++i) { 5122 sp<RecordTrack> track = mTracks[i]; 5123 if (track != 0) { 5124 track->dump(buffer, SIZE); 5125 result.append(buffer); 5126 } 5127 } 5128 5129 size_t size = mActiveTracks.size(); 5130 if (size > 0) { 5131 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5132 result.append(buffer); 5133 RecordTrack::appendDumpHeader(result); 5134 for (size_t i = 0; i < size; ++i) { 5135 sp<RecordTrack> track = mActiveTracks[i]; 5136 track->dump(buffer, SIZE); 5137 result.append(buffer); 5138 } 5139 5140 } 5141 write(fd, result.string(), result.size()); 5142} 5143 5144// AudioBufferProvider interface 5145status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5146{ 5147 int32_t rear = mRsmpInRear; 5148 int32_t front = mRsmpInFront; 5149 ssize_t filled = rear - front; 5150 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5151 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5152 front &= mRsmpInFramesP2 - 1; 5153 size_t part1 = mRsmpInFramesP2 - front; 5154 if (part1 > (size_t) filled) { 5155 part1 = filled; 5156 } 5157 size_t ask = buffer->frameCount; 5158 ALOG_ASSERT(ask > 0); 5159 if (part1 > ask) { 5160 part1 = ask; 5161 } 5162 if (part1 == 0) { 5163 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5164 ALOGE("RecordThread::getNextBuffer() starved"); 5165 buffer->raw = NULL; 5166 buffer->frameCount = 0; 5167 mRsmpInUnrel = 0; 5168 return NOT_ENOUGH_DATA; 5169 } 5170 5171 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5172 buffer->frameCount = part1; 5173 mRsmpInUnrel = part1; 5174 return NO_ERROR; 5175} 5176 5177// AudioBufferProvider interface 5178void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5179{ 5180 size_t stepCount = buffer->frameCount; 5181 if (stepCount == 0) { 5182 return; 5183 } 5184 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5185 mRsmpInUnrel -= stepCount; 5186 mRsmpInFront += stepCount; 5187 buffer->raw = NULL; 5188 buffer->frameCount = 0; 5189} 5190 5191bool AudioFlinger::RecordThread::checkForNewParameters_l() 5192{ 5193 bool reconfig = false; 5194 5195 while (!mNewParameters.isEmpty()) { 5196 status_t status = NO_ERROR; 5197 String8 keyValuePair = mNewParameters[0]; 5198 AudioParameter param = AudioParameter(keyValuePair); 5199 int value; 5200 audio_format_t reqFormat = mFormat; 5201 uint32_t reqSamplingRate = mReqSampleRate; 5202 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5203 5204 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5205 reqSamplingRate = value; 5206 reconfig = true; 5207 } 5208 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5209 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5210 status = BAD_VALUE; 5211 } else { 5212 reqFormat = (audio_format_t) value; 5213 reconfig = true; 5214 } 5215 } 5216 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5217 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5218 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5219 status = BAD_VALUE; 5220 } else { 5221 reqChannelMask = mask; 5222 reconfig = true; 5223 } 5224 } 5225 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5226 // do not accept frame count changes if tracks are open as the track buffer 5227 // size depends on frame count and correct behavior would not be guaranteed 5228 // if frame count is changed after track creation 5229 if (mActiveTracks.size() > 0) { 5230 status = INVALID_OPERATION; 5231 } else { 5232 reconfig = true; 5233 } 5234 } 5235 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5236 // forward device change to effects that have requested to be 5237 // aware of attached audio device. 5238 for (size_t i = 0; i < mEffectChains.size(); i++) { 5239 mEffectChains[i]->setDevice_l(value); 5240 } 5241 5242 // store input device and output device but do not forward output device to audio HAL. 5243 // Note that status is ignored by the caller for output device 5244 // (see AudioFlinger::setParameters() 5245 if (audio_is_output_devices(value)) { 5246 mOutDevice = value; 5247 status = BAD_VALUE; 5248 } else { 5249 mInDevice = value; 5250 // disable AEC and NS if the device is a BT SCO headset supporting those 5251 // pre processings 5252 if (mTracks.size() > 0) { 5253 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5254 mAudioFlinger->btNrecIsOff(); 5255 for (size_t i = 0; i < mTracks.size(); i++) { 5256 sp<RecordTrack> track = mTracks[i]; 5257 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5258 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5259 } 5260 } 5261 } 5262 } 5263 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5264 mAudioSource != (audio_source_t)value) { 5265 // forward device change to effects that have requested to be 5266 // aware of attached audio device. 5267 for (size_t i = 0; i < mEffectChains.size(); i++) { 5268 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5269 } 5270 mAudioSource = (audio_source_t)value; 5271 } 5272 5273 if (status == NO_ERROR) { 5274 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5275 keyValuePair.string()); 5276 if (status == INVALID_OPERATION) { 5277 inputStandBy(); 5278 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5279 keyValuePair.string()); 5280 } 5281 if (reconfig) { 5282 if (status == BAD_VALUE && 5283 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5284 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5285 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5286 <= (2 * reqSamplingRate)) && 5287 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5288 <= FCC_2 && 5289 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5290 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5291 status = NO_ERROR; 5292 } 5293 if (status == NO_ERROR) { 5294 readInputParameters(); 5295 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5296 } 5297 } 5298 } 5299 5300 mNewParameters.removeAt(0); 5301 5302 mParamStatus = status; 5303 mParamCond.signal(); 5304 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5305 // already timed out waiting for the status and will never signal the condition. 5306 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5307 } 5308 return reconfig; 5309} 5310 5311String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5312{ 5313 Mutex::Autolock _l(mLock); 5314 if (initCheck() != NO_ERROR) { 5315 return String8(); 5316 } 5317 5318 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5319 const String8 out_s8(s); 5320 free(s); 5321 return out_s8; 5322} 5323 5324void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5325 AudioSystem::OutputDescriptor desc; 5326 const void *param2 = NULL; 5327 5328 switch (event) { 5329 case AudioSystem::INPUT_OPENED: 5330 case AudioSystem::INPUT_CONFIG_CHANGED: 5331 desc.channelMask = mChannelMask; 5332 desc.samplingRate = mSampleRate; 5333 desc.format = mFormat; 5334 desc.frameCount = mFrameCount; 5335 desc.latency = 0; 5336 param2 = &desc; 5337 break; 5338 5339 case AudioSystem::INPUT_CLOSED: 5340 default: 5341 break; 5342 } 5343 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5344} 5345 5346void AudioFlinger::RecordThread::readInputParameters() 5347{ 5348 delete[] mRsmpInBuffer; 5349 // mRsmpInBuffer is always assigned a new[] below 5350 delete[] mRsmpOutBuffer; 5351 mRsmpOutBuffer = NULL; 5352 delete mResampler; 5353 mResampler = NULL; 5354 5355 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5356 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5357 mChannelCount = popcount(mChannelMask); 5358 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5359 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5360 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5361 } 5362 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5363 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5364 mFrameCount = mBufferSize / mFrameSize; 5365 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5366 // 1 full output buffer, regardless of the alignment of the available input. 5367 mRsmpInFrames = mFrameCount * 3; 5368 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5369 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5370 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5371 mRsmpInFront = 0; 5372 mRsmpInRear = 0; 5373 mRsmpInUnrel = 0; 5374 5375 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5376 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5377 mResampler->setSampleRate(mSampleRate); 5378 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5379 // resampler always outputs stereo 5380 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5381 } 5382 mRsmpInIndex = mFrameCount; 5383} 5384 5385uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5386{ 5387 Mutex::Autolock _l(mLock); 5388 if (initCheck() != NO_ERROR) { 5389 return 0; 5390 } 5391 5392 return mInput->stream->get_input_frames_lost(mInput->stream); 5393} 5394 5395uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5396{ 5397 Mutex::Autolock _l(mLock); 5398 uint32_t result = 0; 5399 if (getEffectChain_l(sessionId) != 0) { 5400 result = EFFECT_SESSION; 5401 } 5402 5403 for (size_t i = 0; i < mTracks.size(); ++i) { 5404 if (sessionId == mTracks[i]->sessionId()) { 5405 result |= TRACK_SESSION; 5406 break; 5407 } 5408 } 5409 5410 return result; 5411} 5412 5413KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5414{ 5415 KeyedVector<int, bool> ids; 5416 Mutex::Autolock _l(mLock); 5417 for (size_t j = 0; j < mTracks.size(); ++j) { 5418 sp<RecordThread::RecordTrack> track = mTracks[j]; 5419 int sessionId = track->sessionId(); 5420 if (ids.indexOfKey(sessionId) < 0) { 5421 ids.add(sessionId, true); 5422 } 5423 } 5424 return ids; 5425} 5426 5427AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5428{ 5429 Mutex::Autolock _l(mLock); 5430 AudioStreamIn *input = mInput; 5431 mInput = NULL; 5432 return input; 5433} 5434 5435// this method must always be called either with ThreadBase mLock held or inside the thread loop 5436audio_stream_t* AudioFlinger::RecordThread::stream() const 5437{ 5438 if (mInput == NULL) { 5439 return NULL; 5440 } 5441 return &mInput->stream->common; 5442} 5443 5444status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5445{ 5446 // only one chain per input thread 5447 if (mEffectChains.size() != 0) { 5448 return INVALID_OPERATION; 5449 } 5450 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5451 5452 chain->setInBuffer(NULL); 5453 chain->setOutBuffer(NULL); 5454 5455 checkSuspendOnAddEffectChain_l(chain); 5456 5457 mEffectChains.add(chain); 5458 5459 return NO_ERROR; 5460} 5461 5462size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5463{ 5464 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5465 ALOGW_IF(mEffectChains.size() != 1, 5466 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5467 chain.get(), mEffectChains.size(), this); 5468 if (mEffectChains.size() == 1) { 5469 mEffectChains.removeAt(0); 5470 } 5471 return 0; 5472} 5473 5474}; // namespace android 5475