Threads.cpp revision 74935e44734c1ec235c2b6677db3e0dbefa5ddb8
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 track.clear(); 1345 goto Exit; 1346 } 1347 1348 mTracks.add(track); 1349 1350 sp<EffectChain> chain = getEffectChain_l(sessionId); 1351 if (chain != 0) { 1352 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1353 track->setMainBuffer(chain->inBuffer()); 1354 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1355 chain->incTrackCnt(); 1356 } 1357 1358 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1359 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1360 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1361 // so ask activity manager to do this on our behalf 1362 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1363 } 1364 } 1365 1366 lStatus = NO_ERROR; 1367 1368Exit: 1369 *status = lStatus; 1370 return track; 1371} 1372 1373uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1374{ 1375 return latency; 1376} 1377 1378uint32_t AudioFlinger::PlaybackThread::latency() const 1379{ 1380 Mutex::Autolock _l(mLock); 1381 return latency_l(); 1382} 1383uint32_t AudioFlinger::PlaybackThread::latency_l() const 1384{ 1385 if (initCheck() == NO_ERROR) { 1386 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1387 } else { 1388 return 0; 1389 } 1390} 1391 1392void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1393{ 1394 Mutex::Autolock _l(mLock); 1395 // Don't apply master volume in SW if our HAL can do it for us. 1396 if (mOutput && mOutput->audioHwDev && 1397 mOutput->audioHwDev->canSetMasterVolume()) { 1398 mMasterVolume = 1.0; 1399 } else { 1400 mMasterVolume = value; 1401 } 1402} 1403 1404void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1405{ 1406 Mutex::Autolock _l(mLock); 1407 // Don't apply master mute in SW if our HAL can do it for us. 1408 if (mOutput && mOutput->audioHwDev && 1409 mOutput->audioHwDev->canSetMasterMute()) { 1410 mMasterMute = false; 1411 } else { 1412 mMasterMute = muted; 1413 } 1414} 1415 1416void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1417{ 1418 Mutex::Autolock _l(mLock); 1419 mStreamTypes[stream].volume = value; 1420 broadcast_l(); 1421} 1422 1423void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 mStreamTypes[stream].mute = muted; 1427 broadcast_l(); 1428} 1429 1430float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1431{ 1432 Mutex::Autolock _l(mLock); 1433 return mStreamTypes[stream].volume; 1434} 1435 1436// addTrack_l() must be called with ThreadBase::mLock held 1437status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1438{ 1439 status_t status = ALREADY_EXISTS; 1440 1441 // set retry count for buffer fill 1442 track->mRetryCount = kMaxTrackStartupRetries; 1443 if (mActiveTracks.indexOf(track) < 0) { 1444 // the track is newly added, make sure it fills up all its 1445 // buffers before playing. This is to ensure the client will 1446 // effectively get the latency it requested. 1447 if (!track->isOutputTrack()) { 1448 TrackBase::track_state state = track->mState; 1449 mLock.unlock(); 1450 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1451 mLock.lock(); 1452 // abort track was stopped/paused while we released the lock 1453 if (state != track->mState) { 1454 if (status == NO_ERROR) { 1455 mLock.unlock(); 1456 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1457 mLock.lock(); 1458 } 1459 return INVALID_OPERATION; 1460 } 1461 // abort if start is rejected by audio policy manager 1462 if (status != NO_ERROR) { 1463 return PERMISSION_DENIED; 1464 } 1465#ifdef ADD_BATTERY_DATA 1466 // to track the speaker usage 1467 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1468#endif 1469 } 1470 1471 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1472 track->mResetDone = false; 1473 track->mPresentationCompleteFrames = 0; 1474 mActiveTracks.add(track); 1475 mWakeLockUids.add(track->uid()); 1476 mActiveTracksGeneration++; 1477 mLatestActiveTrack = track; 1478 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1479 if (chain != 0) { 1480 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1481 track->sessionId()); 1482 chain->incActiveTrackCnt(); 1483 } 1484 1485 status = NO_ERROR; 1486 } 1487 1488 ALOGV("signal playback thread"); 1489 broadcast_l(); 1490 1491 return status; 1492} 1493 1494bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1495{ 1496 track->terminate(); 1497 // active tracks are removed by threadLoop() 1498 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1499 track->mState = TrackBase::STOPPED; 1500 if (!trackActive) { 1501 removeTrack_l(track); 1502 } else if (track->isFastTrack() || track->isOffloaded()) { 1503 track->mState = TrackBase::STOPPING_1; 1504 } 1505 1506 return trackActive; 1507} 1508 1509void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1510{ 1511 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1512 mTracks.remove(track); 1513 deleteTrackName_l(track->name()); 1514 // redundant as track is about to be destroyed, for dumpsys only 1515 track->mName = -1; 1516 if (track->isFastTrack()) { 1517 int index = track->mFastIndex; 1518 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1519 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1520 mFastTrackAvailMask |= 1 << index; 1521 // redundant as track is about to be destroyed, for dumpsys only 1522 track->mFastIndex = -1; 1523 } 1524 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1525 if (chain != 0) { 1526 chain->decTrackCnt(); 1527 } 1528} 1529 1530void AudioFlinger::PlaybackThread::broadcast_l() 1531{ 1532 // Thread could be blocked waiting for async 1533 // so signal it to handle state changes immediately 1534 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1535 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1536 mSignalPending = true; 1537 mWaitWorkCV.broadcast(); 1538} 1539 1540String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1541{ 1542 Mutex::Autolock _l(mLock); 1543 if (initCheck() != NO_ERROR) { 1544 return String8(); 1545 } 1546 1547 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1548 const String8 out_s8(s); 1549 free(s); 1550 return out_s8; 1551} 1552 1553// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1554void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1555 AudioSystem::OutputDescriptor desc; 1556 void *param2 = NULL; 1557 1558 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1559 param); 1560 1561 switch (event) { 1562 case AudioSystem::OUTPUT_OPENED: 1563 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1564 desc.channelMask = mChannelMask; 1565 desc.samplingRate = mSampleRate; 1566 desc.format = mFormat; 1567 desc.frameCount = mNormalFrameCount; // FIXME see 1568 // AudioFlinger::frameCount(audio_io_handle_t) 1569 desc.latency = latency(); 1570 param2 = &desc; 1571 break; 1572 1573 case AudioSystem::STREAM_CONFIG_CHANGED: 1574 param2 = ¶m; 1575 case AudioSystem::OUTPUT_CLOSED: 1576 default: 1577 break; 1578 } 1579 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1580} 1581 1582void AudioFlinger::PlaybackThread::writeCallback() 1583{ 1584 ALOG_ASSERT(mCallbackThread != 0); 1585 mCallbackThread->resetWriteBlocked(); 1586} 1587 1588void AudioFlinger::PlaybackThread::drainCallback() 1589{ 1590 ALOG_ASSERT(mCallbackThread != 0); 1591 mCallbackThread->resetDraining(); 1592} 1593 1594void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1595{ 1596 Mutex::Autolock _l(mLock); 1597 // reject out of sequence requests 1598 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1599 mWriteAckSequence &= ~1; 1600 mWaitWorkCV.signal(); 1601 } 1602} 1603 1604void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1605{ 1606 Mutex::Autolock _l(mLock); 1607 // reject out of sequence requests 1608 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1609 mDrainSequence &= ~1; 1610 mWaitWorkCV.signal(); 1611 } 1612} 1613 1614// static 1615int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1616 void *param, 1617 void *cookie) 1618{ 1619 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1620 ALOGV("asyncCallback() event %d", event); 1621 switch (event) { 1622 case STREAM_CBK_EVENT_WRITE_READY: 1623 me->writeCallback(); 1624 break; 1625 case STREAM_CBK_EVENT_DRAIN_READY: 1626 me->drainCallback(); 1627 break; 1628 default: 1629 ALOGW("asyncCallback() unknown event %d", event); 1630 break; 1631 } 1632 return 0; 1633} 1634 1635void AudioFlinger::PlaybackThread::readOutputParameters() 1636{ 1637 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1638 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1639 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1640 if (!audio_is_output_channel(mChannelMask)) { 1641 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1642 } 1643 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1644 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1645 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1646 } 1647 mChannelCount = popcount(mChannelMask); 1648 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1649 if (!audio_is_valid_format(mFormat)) { 1650 LOG_FATAL("HAL format %d not valid for output", mFormat); 1651 } 1652 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1653 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1654 mFormat); 1655 } 1656 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1657 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1658 mFrameCount = mBufferSize / mFrameSize; 1659 if (mFrameCount & 15) { 1660 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1661 mFrameCount); 1662 } 1663 1664 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1665 (mOutput->stream->set_callback != NULL)) { 1666 if (mOutput->stream->set_callback(mOutput->stream, 1667 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1668 mUseAsyncWrite = true; 1669 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1670 } 1671 } 1672 1673 // Calculate size of normal mix buffer relative to the HAL output buffer size 1674 double multiplier = 1.0; 1675 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1676 kUseFastMixer == FastMixer_Dynamic)) { 1677 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1678 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1679 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1680 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1681 maxNormalFrameCount = maxNormalFrameCount & ~15; 1682 if (maxNormalFrameCount < minNormalFrameCount) { 1683 maxNormalFrameCount = minNormalFrameCount; 1684 } 1685 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1686 if (multiplier <= 1.0) { 1687 multiplier = 1.0; 1688 } else if (multiplier <= 2.0) { 1689 if (2 * mFrameCount <= maxNormalFrameCount) { 1690 multiplier = 2.0; 1691 } else { 1692 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1693 } 1694 } else { 1695 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1696 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1697 // track, but we sometimes have to do this to satisfy the maximum frame count 1698 // constraint) 1699 // FIXME this rounding up should not be done if no HAL SRC 1700 uint32_t truncMult = (uint32_t) multiplier; 1701 if ((truncMult & 1)) { 1702 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1703 ++truncMult; 1704 } 1705 } 1706 multiplier = (double) truncMult; 1707 } 1708 } 1709 mNormalFrameCount = multiplier * mFrameCount; 1710 // round up to nearest 16 frames to satisfy AudioMixer 1711 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1712 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1713 mNormalFrameCount); 1714 1715 delete[] mMixBuffer; 1716 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1717 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1718 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1719 memset(mMixBuffer, 0, normalBufferSize); 1720 1721 // force reconfiguration of effect chains and engines to take new buffer size and audio 1722 // parameters into account 1723 // Note that mLock is not held when readOutputParameters() is called from the constructor 1724 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1725 // matter. 1726 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1727 Vector< sp<EffectChain> > effectChains = mEffectChains; 1728 for (size_t i = 0; i < effectChains.size(); i ++) { 1729 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1730 } 1731} 1732 1733 1734status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1735{ 1736 if (halFrames == NULL || dspFrames == NULL) { 1737 return BAD_VALUE; 1738 } 1739 Mutex::Autolock _l(mLock); 1740 if (initCheck() != NO_ERROR) { 1741 return INVALID_OPERATION; 1742 } 1743 size_t framesWritten = mBytesWritten / mFrameSize; 1744 *halFrames = framesWritten; 1745 1746 if (isSuspended()) { 1747 // return an estimation of rendered frames when the output is suspended 1748 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1749 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1750 return NO_ERROR; 1751 } else { 1752 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1753 } 1754} 1755 1756uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1757{ 1758 Mutex::Autolock _l(mLock); 1759 uint32_t result = 0; 1760 if (getEffectChain_l(sessionId) != 0) { 1761 result = EFFECT_SESSION; 1762 } 1763 1764 for (size_t i = 0; i < mTracks.size(); ++i) { 1765 sp<Track> track = mTracks[i]; 1766 if (sessionId == track->sessionId() && !track->isInvalid()) { 1767 result |= TRACK_SESSION; 1768 break; 1769 } 1770 } 1771 1772 return result; 1773} 1774 1775uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1776{ 1777 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1778 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1779 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1780 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1781 } 1782 for (size_t i = 0; i < mTracks.size(); i++) { 1783 sp<Track> track = mTracks[i]; 1784 if (sessionId == track->sessionId() && !track->isInvalid()) { 1785 return AudioSystem::getStrategyForStream(track->streamType()); 1786 } 1787 } 1788 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1789} 1790 1791 1792AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1793{ 1794 Mutex::Autolock _l(mLock); 1795 return mOutput; 1796} 1797 1798AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1799{ 1800 Mutex::Autolock _l(mLock); 1801 AudioStreamOut *output = mOutput; 1802 mOutput = NULL; 1803 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1804 // must push a NULL and wait for ack 1805 mOutputSink.clear(); 1806 mPipeSink.clear(); 1807 mNormalSink.clear(); 1808 return output; 1809} 1810 1811// this method must always be called either with ThreadBase mLock held or inside the thread loop 1812audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1813{ 1814 if (mOutput == NULL) { 1815 return NULL; 1816 } 1817 return &mOutput->stream->common; 1818} 1819 1820uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1821{ 1822 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1823} 1824 1825status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1826{ 1827 if (!isValidSyncEvent(event)) { 1828 return BAD_VALUE; 1829 } 1830 1831 Mutex::Autolock _l(mLock); 1832 1833 for (size_t i = 0; i < mTracks.size(); ++i) { 1834 sp<Track> track = mTracks[i]; 1835 if (event->triggerSession() == track->sessionId()) { 1836 (void) track->setSyncEvent(event); 1837 return NO_ERROR; 1838 } 1839 } 1840 1841 return NAME_NOT_FOUND; 1842} 1843 1844bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1845{ 1846 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1847} 1848 1849void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1850 const Vector< sp<Track> >& tracksToRemove) 1851{ 1852 size_t count = tracksToRemove.size(); 1853 if (count > 0) { 1854 for (size_t i = 0 ; i < count ; i++) { 1855 const sp<Track>& track = tracksToRemove.itemAt(i); 1856 if (!track->isOutputTrack()) { 1857 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1858#ifdef ADD_BATTERY_DATA 1859 // to track the speaker usage 1860 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1861#endif 1862 if (track->isTerminated()) { 1863 AudioSystem::releaseOutput(mId); 1864 } 1865 } 1866 } 1867 } 1868} 1869 1870void AudioFlinger::PlaybackThread::checkSilentMode_l() 1871{ 1872 if (!mMasterMute) { 1873 char value[PROPERTY_VALUE_MAX]; 1874 if (property_get("ro.audio.silent", value, "0") > 0) { 1875 char *endptr; 1876 unsigned long ul = strtoul(value, &endptr, 0); 1877 if (*endptr == '\0' && ul != 0) { 1878 ALOGD("Silence is golden"); 1879 // The setprop command will not allow a property to be changed after 1880 // the first time it is set, so we don't have to worry about un-muting. 1881 setMasterMute_l(true); 1882 } 1883 } 1884 } 1885} 1886 1887// shared by MIXER and DIRECT, overridden by DUPLICATING 1888ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1889{ 1890 // FIXME rewrite to reduce number of system calls 1891 mLastWriteTime = systemTime(); 1892 mInWrite = true; 1893 ssize_t bytesWritten; 1894 1895 // If an NBAIO sink is present, use it to write the normal mixer's submix 1896 if (mNormalSink != 0) { 1897#define mBitShift 2 // FIXME 1898 size_t count = mBytesRemaining >> mBitShift; 1899 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1900 ATRACE_BEGIN("write"); 1901 // update the setpoint when AudioFlinger::mScreenState changes 1902 uint32_t screenState = AudioFlinger::mScreenState; 1903 if (screenState != mScreenState) { 1904 mScreenState = screenState; 1905 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1906 if (pipe != NULL) { 1907 pipe->setAvgFrames((mScreenState & 1) ? 1908 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1909 } 1910 } 1911 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1912 ATRACE_END(); 1913 if (framesWritten > 0) { 1914 bytesWritten = framesWritten << mBitShift; 1915 } else { 1916 bytesWritten = framesWritten; 1917 } 1918 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1919 if (status == NO_ERROR) { 1920 size_t totalFramesWritten = mNormalSink->framesWritten(); 1921 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1922 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1923 mLatchDValid = true; 1924 } 1925 } 1926 // otherwise use the HAL / AudioStreamOut directly 1927 } else { 1928 // Direct output and offload threads 1929 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1930 if (mUseAsyncWrite) { 1931 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1932 mWriteAckSequence += 2; 1933 mWriteAckSequence |= 1; 1934 ALOG_ASSERT(mCallbackThread != 0); 1935 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1936 } 1937 // FIXME We should have an implementation of timestamps for direct output threads. 1938 // They are used e.g for multichannel PCM playback over HDMI. 1939 bytesWritten = mOutput->stream->write(mOutput->stream, 1940 (char *)mMixBuffer + offset, mBytesRemaining); 1941 if (mUseAsyncWrite && 1942 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1943 // do not wait for async callback in case of error of full write 1944 mWriteAckSequence &= ~1; 1945 ALOG_ASSERT(mCallbackThread != 0); 1946 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1947 } 1948 } 1949 1950 mNumWrites++; 1951 mInWrite = false; 1952 mStandby = false; 1953 return bytesWritten; 1954} 1955 1956void AudioFlinger::PlaybackThread::threadLoop_drain() 1957{ 1958 if (mOutput->stream->drain) { 1959 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1960 if (mUseAsyncWrite) { 1961 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1962 mDrainSequence |= 1; 1963 ALOG_ASSERT(mCallbackThread != 0); 1964 mCallbackThread->setDraining(mDrainSequence); 1965 } 1966 mOutput->stream->drain(mOutput->stream, 1967 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1968 : AUDIO_DRAIN_ALL); 1969 } 1970} 1971 1972void AudioFlinger::PlaybackThread::threadLoop_exit() 1973{ 1974 // Default implementation has nothing to do 1975} 1976 1977/* 1978The derived values that are cached: 1979 - mixBufferSize from frame count * frame size 1980 - activeSleepTime from activeSleepTimeUs() 1981 - idleSleepTime from idleSleepTimeUs() 1982 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1983 - maxPeriod from frame count and sample rate (MIXER only) 1984 1985The parameters that affect these derived values are: 1986 - frame count 1987 - frame size 1988 - sample rate 1989 - device type: A2DP or not 1990 - device latency 1991 - format: PCM or not 1992 - active sleep time 1993 - idle sleep time 1994*/ 1995 1996void AudioFlinger::PlaybackThread::cacheParameters_l() 1997{ 1998 mixBufferSize = mNormalFrameCount * mFrameSize; 1999 activeSleepTime = activeSleepTimeUs(); 2000 idleSleepTime = idleSleepTimeUs(); 2001} 2002 2003void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2004{ 2005 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2006 this, streamType, mTracks.size()); 2007 Mutex::Autolock _l(mLock); 2008 2009 size_t size = mTracks.size(); 2010 for (size_t i = 0; i < size; i++) { 2011 sp<Track> t = mTracks[i]; 2012 if (t->streamType() == streamType) { 2013 t->invalidate(); 2014 } 2015 } 2016} 2017 2018status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2019{ 2020 int session = chain->sessionId(); 2021 int16_t *buffer = mMixBuffer; 2022 bool ownsBuffer = false; 2023 2024 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2025 if (session > 0) { 2026 // Only one effect chain can be present in direct output thread and it uses 2027 // the mix buffer as input 2028 if (mType != DIRECT) { 2029 size_t numSamples = mNormalFrameCount * mChannelCount; 2030 buffer = new int16_t[numSamples]; 2031 memset(buffer, 0, numSamples * sizeof(int16_t)); 2032 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2033 ownsBuffer = true; 2034 } 2035 2036 // Attach all tracks with same session ID to this chain. 2037 for (size_t i = 0; i < mTracks.size(); ++i) { 2038 sp<Track> track = mTracks[i]; 2039 if (session == track->sessionId()) { 2040 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2041 buffer); 2042 track->setMainBuffer(buffer); 2043 chain->incTrackCnt(); 2044 } 2045 } 2046 2047 // indicate all active tracks in the chain 2048 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2049 sp<Track> track = mActiveTracks[i].promote(); 2050 if (track == 0) { 2051 continue; 2052 } 2053 if (session == track->sessionId()) { 2054 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2055 chain->incActiveTrackCnt(); 2056 } 2057 } 2058 } 2059 2060 chain->setInBuffer(buffer, ownsBuffer); 2061 chain->setOutBuffer(mMixBuffer); 2062 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2063 // chains list in order to be processed last as it contains output stage effects 2064 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2065 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2066 // after track specific effects and before output stage 2067 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2068 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2069 // Effect chain for other sessions are inserted at beginning of effect 2070 // chains list to be processed before output mix effects. Relative order between other 2071 // sessions is not important 2072 size_t size = mEffectChains.size(); 2073 size_t i = 0; 2074 for (i = 0; i < size; i++) { 2075 if (mEffectChains[i]->sessionId() < session) { 2076 break; 2077 } 2078 } 2079 mEffectChains.insertAt(chain, i); 2080 checkSuspendOnAddEffectChain_l(chain); 2081 2082 return NO_ERROR; 2083} 2084 2085size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2086{ 2087 int session = chain->sessionId(); 2088 2089 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2090 2091 for (size_t i = 0; i < mEffectChains.size(); i++) { 2092 if (chain == mEffectChains[i]) { 2093 mEffectChains.removeAt(i); 2094 // detach all active tracks from the chain 2095 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2096 sp<Track> track = mActiveTracks[i].promote(); 2097 if (track == 0) { 2098 continue; 2099 } 2100 if (session == track->sessionId()) { 2101 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2102 chain.get(), session); 2103 chain->decActiveTrackCnt(); 2104 } 2105 } 2106 2107 // detach all tracks with same session ID from this chain 2108 for (size_t i = 0; i < mTracks.size(); ++i) { 2109 sp<Track> track = mTracks[i]; 2110 if (session == track->sessionId()) { 2111 track->setMainBuffer(mMixBuffer); 2112 chain->decTrackCnt(); 2113 } 2114 } 2115 break; 2116 } 2117 } 2118 return mEffectChains.size(); 2119} 2120 2121status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2122 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2123{ 2124 Mutex::Autolock _l(mLock); 2125 return attachAuxEffect_l(track, EffectId); 2126} 2127 2128status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2129 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2130{ 2131 status_t status = NO_ERROR; 2132 2133 if (EffectId == 0) { 2134 track->setAuxBuffer(0, NULL); 2135 } else { 2136 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2137 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2138 if (effect != 0) { 2139 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2140 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2141 } else { 2142 status = INVALID_OPERATION; 2143 } 2144 } else { 2145 status = BAD_VALUE; 2146 } 2147 } 2148 return status; 2149} 2150 2151void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2152{ 2153 for (size_t i = 0; i < mTracks.size(); ++i) { 2154 sp<Track> track = mTracks[i]; 2155 if (track->auxEffectId() == effectId) { 2156 attachAuxEffect_l(track, 0); 2157 } 2158 } 2159} 2160 2161bool AudioFlinger::PlaybackThread::threadLoop() 2162{ 2163 Vector< sp<Track> > tracksToRemove; 2164 2165 standbyTime = systemTime(); 2166 2167 // MIXER 2168 nsecs_t lastWarning = 0; 2169 2170 // DUPLICATING 2171 // FIXME could this be made local to while loop? 2172 writeFrames = 0; 2173 2174 int lastGeneration = 0; 2175 2176 cacheParameters_l(); 2177 sleepTime = idleSleepTime; 2178 2179 if (mType == MIXER) { 2180 sleepTimeShift = 0; 2181 } 2182 2183 CpuStats cpuStats; 2184 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2185 2186 acquireWakeLock(); 2187 2188 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2189 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2190 // and then that string will be logged at the next convenient opportunity. 2191 const char *logString = NULL; 2192 2193 checkSilentMode_l(); 2194 2195 while (!exitPending()) 2196 { 2197 cpuStats.sample(myName); 2198 2199 Vector< sp<EffectChain> > effectChains; 2200 2201 processConfigEvents(); 2202 2203 { // scope for mLock 2204 2205 Mutex::Autolock _l(mLock); 2206 2207 if (logString != NULL) { 2208 mNBLogWriter->logTimestamp(); 2209 mNBLogWriter->log(logString); 2210 logString = NULL; 2211 } 2212 2213 if (mLatchDValid) { 2214 mLatchQ = mLatchD; 2215 mLatchDValid = false; 2216 mLatchQValid = true; 2217 } 2218 2219 if (checkForNewParameters_l()) { 2220 cacheParameters_l(); 2221 } 2222 2223 saveOutputTracks(); 2224 if (mSignalPending) { 2225 // A signal was raised while we were unlocked 2226 mSignalPending = false; 2227 } else if (waitingAsyncCallback_l()) { 2228 if (exitPending()) { 2229 break; 2230 } 2231 releaseWakeLock_l(); 2232 mWakeLockUids.clear(); 2233 mActiveTracksGeneration++; 2234 ALOGV("wait async completion"); 2235 mWaitWorkCV.wait(mLock); 2236 ALOGV("async completion/wake"); 2237 acquireWakeLock_l(); 2238 standbyTime = systemTime() + standbyDelay; 2239 sleepTime = 0; 2240 2241 continue; 2242 } 2243 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2244 isSuspended()) { 2245 // put audio hardware into standby after short delay 2246 if (shouldStandby_l()) { 2247 2248 threadLoop_standby(); 2249 2250 mStandby = true; 2251 } 2252 2253 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2254 // we're about to wait, flush the binder command buffer 2255 IPCThreadState::self()->flushCommands(); 2256 2257 clearOutputTracks(); 2258 2259 if (exitPending()) { 2260 break; 2261 } 2262 2263 releaseWakeLock_l(); 2264 mWakeLockUids.clear(); 2265 mActiveTracksGeneration++; 2266 // wait until we have something to do... 2267 ALOGV("%s going to sleep", myName.string()); 2268 mWaitWorkCV.wait(mLock); 2269 ALOGV("%s waking up", myName.string()); 2270 acquireWakeLock_l(); 2271 2272 mMixerStatus = MIXER_IDLE; 2273 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2274 mBytesWritten = 0; 2275 mBytesRemaining = 0; 2276 checkSilentMode_l(); 2277 2278 standbyTime = systemTime() + standbyDelay; 2279 sleepTime = idleSleepTime; 2280 if (mType == MIXER) { 2281 sleepTimeShift = 0; 2282 } 2283 2284 continue; 2285 } 2286 } 2287 // mMixerStatusIgnoringFastTracks is also updated internally 2288 mMixerStatus = prepareTracks_l(&tracksToRemove); 2289 2290 // compare with previously applied list 2291 if (lastGeneration != mActiveTracksGeneration) { 2292 // update wakelock 2293 updateWakeLockUids_l(mWakeLockUids); 2294 lastGeneration = mActiveTracksGeneration; 2295 } 2296 2297 // prevent any changes in effect chain list and in each effect chain 2298 // during mixing and effect process as the audio buffers could be deleted 2299 // or modified if an effect is created or deleted 2300 lockEffectChains_l(effectChains); 2301 } // mLock scope ends 2302 2303 if (mBytesRemaining == 0) { 2304 mCurrentWriteLength = 0; 2305 if (mMixerStatus == MIXER_TRACKS_READY) { 2306 // threadLoop_mix() sets mCurrentWriteLength 2307 threadLoop_mix(); 2308 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2309 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2310 // threadLoop_sleepTime sets sleepTime to 0 if data 2311 // must be written to HAL 2312 threadLoop_sleepTime(); 2313 if (sleepTime == 0) { 2314 mCurrentWriteLength = mixBufferSize; 2315 } 2316 } 2317 mBytesRemaining = mCurrentWriteLength; 2318 if (isSuspended()) { 2319 sleepTime = suspendSleepTimeUs(); 2320 // simulate write to HAL when suspended 2321 mBytesWritten += mixBufferSize; 2322 mBytesRemaining = 0; 2323 } 2324 2325 // only process effects if we're going to write 2326 if (sleepTime == 0 && mType != OFFLOAD) { 2327 for (size_t i = 0; i < effectChains.size(); i ++) { 2328 effectChains[i]->process_l(); 2329 } 2330 } 2331 } 2332 // Process effect chains for offloaded thread even if no audio 2333 // was read from audio track: process only updates effect state 2334 // and thus does have to be synchronized with audio writes but may have 2335 // to be called while waiting for async write callback 2336 if (mType == OFFLOAD) { 2337 for (size_t i = 0; i < effectChains.size(); i ++) { 2338 effectChains[i]->process_l(); 2339 } 2340 } 2341 2342 // enable changes in effect chain 2343 unlockEffectChains(effectChains); 2344 2345 if (!waitingAsyncCallback()) { 2346 // sleepTime == 0 means we must write to audio hardware 2347 if (sleepTime == 0) { 2348 if (mBytesRemaining) { 2349 ssize_t ret = threadLoop_write(); 2350 if (ret < 0) { 2351 mBytesRemaining = 0; 2352 } else { 2353 mBytesWritten += ret; 2354 mBytesRemaining -= ret; 2355 } 2356 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2357 (mMixerStatus == MIXER_DRAIN_ALL)) { 2358 threadLoop_drain(); 2359 } 2360if (mType == MIXER) { 2361 // write blocked detection 2362 nsecs_t now = systemTime(); 2363 nsecs_t delta = now - mLastWriteTime; 2364 if (!mStandby && delta > maxPeriod) { 2365 mNumDelayedWrites++; 2366 if ((now - lastWarning) > kWarningThrottleNs) { 2367 ATRACE_NAME("underrun"); 2368 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2369 ns2ms(delta), mNumDelayedWrites, this); 2370 lastWarning = now; 2371 } 2372 } 2373} 2374 2375 } else { 2376 usleep(sleepTime); 2377 } 2378 } 2379 2380 // Finally let go of removed track(s), without the lock held 2381 // since we can't guarantee the destructors won't acquire that 2382 // same lock. This will also mutate and push a new fast mixer state. 2383 threadLoop_removeTracks(tracksToRemove); 2384 tracksToRemove.clear(); 2385 2386 // FIXME I don't understand the need for this here; 2387 // it was in the original code but maybe the 2388 // assignment in saveOutputTracks() makes this unnecessary? 2389 clearOutputTracks(); 2390 2391 // Effect chains will be actually deleted here if they were removed from 2392 // mEffectChains list during mixing or effects processing 2393 effectChains.clear(); 2394 2395 // FIXME Note that the above .clear() is no longer necessary since effectChains 2396 // is now local to this block, but will keep it for now (at least until merge done). 2397 } 2398 2399 threadLoop_exit(); 2400 2401 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2402 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2403 // put output stream into standby mode 2404 if (!mStandby) { 2405 mOutput->stream->common.standby(&mOutput->stream->common); 2406 } 2407 } 2408 2409 releaseWakeLock(); 2410 mWakeLockUids.clear(); 2411 mActiveTracksGeneration++; 2412 2413 ALOGV("Thread %p type %d exiting", this, mType); 2414 return false; 2415} 2416 2417// removeTracks_l() must be called with ThreadBase::mLock held 2418void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2419{ 2420 size_t count = tracksToRemove.size(); 2421 if (count > 0) { 2422 for (size_t i=0 ; i<count ; i++) { 2423 const sp<Track>& track = tracksToRemove.itemAt(i); 2424 mActiveTracks.remove(track); 2425 mWakeLockUids.remove(track->uid()); 2426 mActiveTracksGeneration++; 2427 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2428 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2429 if (chain != 0) { 2430 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2431 track->sessionId()); 2432 chain->decActiveTrackCnt(); 2433 } 2434 if (track->isTerminated()) { 2435 removeTrack_l(track); 2436 } 2437 } 2438 } 2439 2440} 2441 2442status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2443{ 2444 if (mNormalSink != 0) { 2445 return mNormalSink->getTimestamp(timestamp); 2446 } 2447 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2448 uint64_t position64; 2449 int ret = mOutput->stream->get_presentation_position( 2450 mOutput->stream, &position64, ×tamp.mTime); 2451 if (ret == 0) { 2452 timestamp.mPosition = (uint32_t)position64; 2453 return NO_ERROR; 2454 } 2455 } 2456 return INVALID_OPERATION; 2457} 2458// ---------------------------------------------------------------------------- 2459 2460AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2461 audio_io_handle_t id, audio_devices_t device, type_t type) 2462 : PlaybackThread(audioFlinger, output, id, device, type), 2463 // mAudioMixer below 2464 // mFastMixer below 2465 mFastMixerFutex(0) 2466 // mOutputSink below 2467 // mPipeSink below 2468 // mNormalSink below 2469{ 2470 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2471 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2472 "mFrameCount=%d, mNormalFrameCount=%d", 2473 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2474 mNormalFrameCount); 2475 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2476 2477 // FIXME - Current mixer implementation only supports stereo output 2478 if (mChannelCount != FCC_2) { 2479 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2480 } 2481 2482 // create an NBAIO sink for the HAL output stream, and negotiate 2483 mOutputSink = new AudioStreamOutSink(output->stream); 2484 size_t numCounterOffers = 0; 2485 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2486 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2487 ALOG_ASSERT(index == 0); 2488 2489 // initialize fast mixer depending on configuration 2490 bool initFastMixer; 2491 switch (kUseFastMixer) { 2492 case FastMixer_Never: 2493 initFastMixer = false; 2494 break; 2495 case FastMixer_Always: 2496 initFastMixer = true; 2497 break; 2498 case FastMixer_Static: 2499 case FastMixer_Dynamic: 2500 initFastMixer = mFrameCount < mNormalFrameCount; 2501 break; 2502 } 2503 if (initFastMixer) { 2504 2505 // create a MonoPipe to connect our submix to FastMixer 2506 NBAIO_Format format = mOutputSink->format(); 2507 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2508 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2509 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2510 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2511 const NBAIO_Format offers[1] = {format}; 2512 size_t numCounterOffers = 0; 2513 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2514 ALOG_ASSERT(index == 0); 2515 monoPipe->setAvgFrames((mScreenState & 1) ? 2516 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2517 mPipeSink = monoPipe; 2518 2519#ifdef TEE_SINK 2520 if (mTeeSinkOutputEnabled) { 2521 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2522 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2523 numCounterOffers = 0; 2524 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2525 ALOG_ASSERT(index == 0); 2526 mTeeSink = teeSink; 2527 PipeReader *teeSource = new PipeReader(*teeSink); 2528 numCounterOffers = 0; 2529 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2530 ALOG_ASSERT(index == 0); 2531 mTeeSource = teeSource; 2532 } 2533#endif 2534 2535 // create fast mixer and configure it initially with just one fast track for our submix 2536 mFastMixer = new FastMixer(); 2537 FastMixerStateQueue *sq = mFastMixer->sq(); 2538#ifdef STATE_QUEUE_DUMP 2539 sq->setObserverDump(&mStateQueueObserverDump); 2540 sq->setMutatorDump(&mStateQueueMutatorDump); 2541#endif 2542 FastMixerState *state = sq->begin(); 2543 FastTrack *fastTrack = &state->mFastTracks[0]; 2544 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2545 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2546 fastTrack->mVolumeProvider = NULL; 2547 fastTrack->mGeneration++; 2548 state->mFastTracksGen++; 2549 state->mTrackMask = 1; 2550 // fast mixer will use the HAL output sink 2551 state->mOutputSink = mOutputSink.get(); 2552 state->mOutputSinkGen++; 2553 state->mFrameCount = mFrameCount; 2554 state->mCommand = FastMixerState::COLD_IDLE; 2555 // already done in constructor initialization list 2556 //mFastMixerFutex = 0; 2557 state->mColdFutexAddr = &mFastMixerFutex; 2558 state->mColdGen++; 2559 state->mDumpState = &mFastMixerDumpState; 2560#ifdef TEE_SINK 2561 state->mTeeSink = mTeeSink.get(); 2562#endif 2563 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2564 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2565 sq->end(); 2566 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2567 2568 // start the fast mixer 2569 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2570 pid_t tid = mFastMixer->getTid(); 2571 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2572 if (err != 0) { 2573 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2574 kPriorityFastMixer, getpid_cached, tid, err); 2575 } 2576 2577#ifdef AUDIO_WATCHDOG 2578 // create and start the watchdog 2579 mAudioWatchdog = new AudioWatchdog(); 2580 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2581 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2582 tid = mAudioWatchdog->getTid(); 2583 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2584 if (err != 0) { 2585 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2586 kPriorityFastMixer, getpid_cached, tid, err); 2587 } 2588#endif 2589 2590 } else { 2591 mFastMixer = NULL; 2592 } 2593 2594 switch (kUseFastMixer) { 2595 case FastMixer_Never: 2596 case FastMixer_Dynamic: 2597 mNormalSink = mOutputSink; 2598 break; 2599 case FastMixer_Always: 2600 mNormalSink = mPipeSink; 2601 break; 2602 case FastMixer_Static: 2603 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2604 break; 2605 } 2606} 2607 2608AudioFlinger::MixerThread::~MixerThread() 2609{ 2610 if (mFastMixer != NULL) { 2611 FastMixerStateQueue *sq = mFastMixer->sq(); 2612 FastMixerState *state = sq->begin(); 2613 if (state->mCommand == FastMixerState::COLD_IDLE) { 2614 int32_t old = android_atomic_inc(&mFastMixerFutex); 2615 if (old == -1) { 2616 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2617 } 2618 } 2619 state->mCommand = FastMixerState::EXIT; 2620 sq->end(); 2621 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2622 mFastMixer->join(); 2623 // Though the fast mixer thread has exited, it's state queue is still valid. 2624 // We'll use that extract the final state which contains one remaining fast track 2625 // corresponding to our sub-mix. 2626 state = sq->begin(); 2627 ALOG_ASSERT(state->mTrackMask == 1); 2628 FastTrack *fastTrack = &state->mFastTracks[0]; 2629 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2630 delete fastTrack->mBufferProvider; 2631 sq->end(false /*didModify*/); 2632 delete mFastMixer; 2633#ifdef AUDIO_WATCHDOG 2634 if (mAudioWatchdog != 0) { 2635 mAudioWatchdog->requestExit(); 2636 mAudioWatchdog->requestExitAndWait(); 2637 mAudioWatchdog.clear(); 2638 } 2639#endif 2640 } 2641 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2642 delete mAudioMixer; 2643} 2644 2645 2646uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2647{ 2648 if (mFastMixer != NULL) { 2649 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2650 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2651 } 2652 return latency; 2653} 2654 2655 2656void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2657{ 2658 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2659} 2660 2661ssize_t AudioFlinger::MixerThread::threadLoop_write() 2662{ 2663 // FIXME we should only do one push per cycle; confirm this is true 2664 // Start the fast mixer if it's not already running 2665 if (mFastMixer != NULL) { 2666 FastMixerStateQueue *sq = mFastMixer->sq(); 2667 FastMixerState *state = sq->begin(); 2668 if (state->mCommand != FastMixerState::MIX_WRITE && 2669 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2670 if (state->mCommand == FastMixerState::COLD_IDLE) { 2671 int32_t old = android_atomic_inc(&mFastMixerFutex); 2672 if (old == -1) { 2673 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2674 } 2675#ifdef AUDIO_WATCHDOG 2676 if (mAudioWatchdog != 0) { 2677 mAudioWatchdog->resume(); 2678 } 2679#endif 2680 } 2681 state->mCommand = FastMixerState::MIX_WRITE; 2682 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2683 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2684 sq->end(); 2685 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2686 if (kUseFastMixer == FastMixer_Dynamic) { 2687 mNormalSink = mPipeSink; 2688 } 2689 } else { 2690 sq->end(false /*didModify*/); 2691 } 2692 } 2693 return PlaybackThread::threadLoop_write(); 2694} 2695 2696void AudioFlinger::MixerThread::threadLoop_standby() 2697{ 2698 // Idle the fast mixer if it's currently running 2699 if (mFastMixer != NULL) { 2700 FastMixerStateQueue *sq = mFastMixer->sq(); 2701 FastMixerState *state = sq->begin(); 2702 if (!(state->mCommand & FastMixerState::IDLE)) { 2703 state->mCommand = FastMixerState::COLD_IDLE; 2704 state->mColdFutexAddr = &mFastMixerFutex; 2705 state->mColdGen++; 2706 mFastMixerFutex = 0; 2707 sq->end(); 2708 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2709 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2710 if (kUseFastMixer == FastMixer_Dynamic) { 2711 mNormalSink = mOutputSink; 2712 } 2713#ifdef AUDIO_WATCHDOG 2714 if (mAudioWatchdog != 0) { 2715 mAudioWatchdog->pause(); 2716 } 2717#endif 2718 } else { 2719 sq->end(false /*didModify*/); 2720 } 2721 } 2722 PlaybackThread::threadLoop_standby(); 2723} 2724 2725// Empty implementation for standard mixer 2726// Overridden for offloaded playback 2727void AudioFlinger::PlaybackThread::flushOutput_l() 2728{ 2729} 2730 2731bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2732{ 2733 return false; 2734} 2735 2736bool AudioFlinger::PlaybackThread::shouldStandby_l() 2737{ 2738 return !mStandby; 2739} 2740 2741bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2742{ 2743 Mutex::Autolock _l(mLock); 2744 return waitingAsyncCallback_l(); 2745} 2746 2747// shared by MIXER and DIRECT, overridden by DUPLICATING 2748void AudioFlinger::PlaybackThread::threadLoop_standby() 2749{ 2750 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2751 mOutput->stream->common.standby(&mOutput->stream->common); 2752 if (mUseAsyncWrite != 0) { 2753 // discard any pending drain or write ack by incrementing sequence 2754 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2755 mDrainSequence = (mDrainSequence + 2) & ~1; 2756 ALOG_ASSERT(mCallbackThread != 0); 2757 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2758 mCallbackThread->setDraining(mDrainSequence); 2759 } 2760} 2761 2762void AudioFlinger::MixerThread::threadLoop_mix() 2763{ 2764 // obtain the presentation timestamp of the next output buffer 2765 int64_t pts; 2766 status_t status = INVALID_OPERATION; 2767 2768 if (mNormalSink != 0) { 2769 status = mNormalSink->getNextWriteTimestamp(&pts); 2770 } else { 2771 status = mOutputSink->getNextWriteTimestamp(&pts); 2772 } 2773 2774 if (status != NO_ERROR) { 2775 pts = AudioBufferProvider::kInvalidPTS; 2776 } 2777 2778 // mix buffers... 2779 mAudioMixer->process(pts); 2780 mCurrentWriteLength = mixBufferSize; 2781 // increase sleep time progressively when application underrun condition clears. 2782 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2783 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2784 // such that we would underrun the audio HAL. 2785 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2786 sleepTimeShift--; 2787 } 2788 sleepTime = 0; 2789 standbyTime = systemTime() + standbyDelay; 2790 //TODO: delay standby when effects have a tail 2791} 2792 2793void AudioFlinger::MixerThread::threadLoop_sleepTime() 2794{ 2795 // If no tracks are ready, sleep once for the duration of an output 2796 // buffer size, then write 0s to the output 2797 if (sleepTime == 0) { 2798 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2799 sleepTime = activeSleepTime >> sleepTimeShift; 2800 if (sleepTime < kMinThreadSleepTimeUs) { 2801 sleepTime = kMinThreadSleepTimeUs; 2802 } 2803 // reduce sleep time in case of consecutive application underruns to avoid 2804 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2805 // duration we would end up writing less data than needed by the audio HAL if 2806 // the condition persists. 2807 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2808 sleepTimeShift++; 2809 } 2810 } else { 2811 sleepTime = idleSleepTime; 2812 } 2813 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2814 memset(mMixBuffer, 0, mixBufferSize); 2815 sleepTime = 0; 2816 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2817 "anticipated start"); 2818 } 2819 // TODO add standby time extension fct of effect tail 2820} 2821 2822// prepareTracks_l() must be called with ThreadBase::mLock held 2823AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2824 Vector< sp<Track> > *tracksToRemove) 2825{ 2826 2827 mixer_state mixerStatus = MIXER_IDLE; 2828 // find out which tracks need to be processed 2829 size_t count = mActiveTracks.size(); 2830 size_t mixedTracks = 0; 2831 size_t tracksWithEffect = 0; 2832 // counts only _active_ fast tracks 2833 size_t fastTracks = 0; 2834 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2835 2836 float masterVolume = mMasterVolume; 2837 bool masterMute = mMasterMute; 2838 2839 if (masterMute) { 2840 masterVolume = 0; 2841 } 2842 // Delegate master volume control to effect in output mix effect chain if needed 2843 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2844 if (chain != 0) { 2845 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2846 chain->setVolume_l(&v, &v); 2847 masterVolume = (float)((v + (1 << 23)) >> 24); 2848 chain.clear(); 2849 } 2850 2851 // prepare a new state to push 2852 FastMixerStateQueue *sq = NULL; 2853 FastMixerState *state = NULL; 2854 bool didModify = false; 2855 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2856 if (mFastMixer != NULL) { 2857 sq = mFastMixer->sq(); 2858 state = sq->begin(); 2859 } 2860 2861 for (size_t i=0 ; i<count ; i++) { 2862 const sp<Track> t = mActiveTracks[i].promote(); 2863 if (t == 0) { 2864 continue; 2865 } 2866 2867 // this const just means the local variable doesn't change 2868 Track* const track = t.get(); 2869 2870 // process fast tracks 2871 if (track->isFastTrack()) { 2872 2873 // It's theoretically possible (though unlikely) for a fast track to be created 2874 // and then removed within the same normal mix cycle. This is not a problem, as 2875 // the track never becomes active so it's fast mixer slot is never touched. 2876 // The converse, of removing an (active) track and then creating a new track 2877 // at the identical fast mixer slot within the same normal mix cycle, 2878 // is impossible because the slot isn't marked available until the end of each cycle. 2879 int j = track->mFastIndex; 2880 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2881 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2882 FastTrack *fastTrack = &state->mFastTracks[j]; 2883 2884 // Determine whether the track is currently in underrun condition, 2885 // and whether it had a recent underrun. 2886 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2887 FastTrackUnderruns underruns = ftDump->mUnderruns; 2888 uint32_t recentFull = (underruns.mBitFields.mFull - 2889 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2890 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2891 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2892 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2893 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2894 uint32_t recentUnderruns = recentPartial + recentEmpty; 2895 track->mObservedUnderruns = underruns; 2896 // don't count underruns that occur while stopping or pausing 2897 // or stopped which can occur when flush() is called while active 2898 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2899 recentUnderruns > 0) { 2900 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2901 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2902 } 2903 2904 // This is similar to the state machine for normal tracks, 2905 // with a few modifications for fast tracks. 2906 bool isActive = true; 2907 switch (track->mState) { 2908 case TrackBase::STOPPING_1: 2909 // track stays active in STOPPING_1 state until first underrun 2910 if (recentUnderruns > 0 || track->isTerminated()) { 2911 track->mState = TrackBase::STOPPING_2; 2912 } 2913 break; 2914 case TrackBase::PAUSING: 2915 // ramp down is not yet implemented 2916 track->setPaused(); 2917 break; 2918 case TrackBase::RESUMING: 2919 // ramp up is not yet implemented 2920 track->mState = TrackBase::ACTIVE; 2921 break; 2922 case TrackBase::ACTIVE: 2923 if (recentFull > 0 || recentPartial > 0) { 2924 // track has provided at least some frames recently: reset retry count 2925 track->mRetryCount = kMaxTrackRetries; 2926 } 2927 if (recentUnderruns == 0) { 2928 // no recent underruns: stay active 2929 break; 2930 } 2931 // there has recently been an underrun of some kind 2932 if (track->sharedBuffer() == 0) { 2933 // were any of the recent underruns "empty" (no frames available)? 2934 if (recentEmpty == 0) { 2935 // no, then ignore the partial underruns as they are allowed indefinitely 2936 break; 2937 } 2938 // there has recently been an "empty" underrun: decrement the retry counter 2939 if (--(track->mRetryCount) > 0) { 2940 break; 2941 } 2942 // indicate to client process that the track was disabled because of underrun; 2943 // it will then automatically call start() when data is available 2944 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2945 // remove from active list, but state remains ACTIVE [confusing but true] 2946 isActive = false; 2947 break; 2948 } 2949 // fall through 2950 case TrackBase::STOPPING_2: 2951 case TrackBase::PAUSED: 2952 case TrackBase::STOPPED: 2953 case TrackBase::FLUSHED: // flush() while active 2954 // Check for presentation complete if track is inactive 2955 // We have consumed all the buffers of this track. 2956 // This would be incomplete if we auto-paused on underrun 2957 { 2958 size_t audioHALFrames = 2959 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2960 size_t framesWritten = mBytesWritten / mFrameSize; 2961 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2962 // track stays in active list until presentation is complete 2963 break; 2964 } 2965 } 2966 if (track->isStopping_2()) { 2967 track->mState = TrackBase::STOPPED; 2968 } 2969 if (track->isStopped()) { 2970 // Can't reset directly, as fast mixer is still polling this track 2971 // track->reset(); 2972 // So instead mark this track as needing to be reset after push with ack 2973 resetMask |= 1 << i; 2974 } 2975 isActive = false; 2976 break; 2977 case TrackBase::IDLE: 2978 default: 2979 LOG_FATAL("unexpected track state %d", track->mState); 2980 } 2981 2982 if (isActive) { 2983 // was it previously inactive? 2984 if (!(state->mTrackMask & (1 << j))) { 2985 ExtendedAudioBufferProvider *eabp = track; 2986 VolumeProvider *vp = track; 2987 fastTrack->mBufferProvider = eabp; 2988 fastTrack->mVolumeProvider = vp; 2989 fastTrack->mSampleRate = track->mSampleRate; 2990 fastTrack->mChannelMask = track->mChannelMask; 2991 fastTrack->mGeneration++; 2992 state->mTrackMask |= 1 << j; 2993 didModify = true; 2994 // no acknowledgement required for newly active tracks 2995 } 2996 // cache the combined master volume and stream type volume for fast mixer; this 2997 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2998 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2999 ++fastTracks; 3000 } else { 3001 // was it previously active? 3002 if (state->mTrackMask & (1 << j)) { 3003 fastTrack->mBufferProvider = NULL; 3004 fastTrack->mGeneration++; 3005 state->mTrackMask &= ~(1 << j); 3006 didModify = true; 3007 // If any fast tracks were removed, we must wait for acknowledgement 3008 // because we're about to decrement the last sp<> on those tracks. 3009 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3010 } else { 3011 LOG_FATAL("fast track %d should have been active", j); 3012 } 3013 tracksToRemove->add(track); 3014 // Avoids a misleading display in dumpsys 3015 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3016 } 3017 continue; 3018 } 3019 3020 { // local variable scope to avoid goto warning 3021 3022 audio_track_cblk_t* cblk = track->cblk(); 3023 3024 // The first time a track is added we wait 3025 // for all its buffers to be filled before processing it 3026 int name = track->name(); 3027 // make sure that we have enough frames to mix one full buffer. 3028 // enforce this condition only once to enable draining the buffer in case the client 3029 // app does not call stop() and relies on underrun to stop: 3030 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3031 // during last round 3032 size_t desiredFrames; 3033 uint32_t sr = track->sampleRate(); 3034 if (sr == mSampleRate) { 3035 desiredFrames = mNormalFrameCount; 3036 } else { 3037 // +1 for rounding and +1 for additional sample needed for interpolation 3038 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3039 // add frames already consumed but not yet released by the resampler 3040 // because mAudioTrackServerProxy->framesReady() will include these frames 3041 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3042#if 0 3043 // the minimum track buffer size is normally twice the number of frames necessary 3044 // to fill one buffer and the resampler should not leave more than one buffer worth 3045 // of unreleased frames after each pass, but just in case... 3046 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3047#endif 3048 } 3049 uint32_t minFrames = 1; 3050 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3051 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3052 minFrames = desiredFrames; 3053 } 3054 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 3055 size_t framesReady; 3056 if (track->sharedBuffer() == 0) { 3057 framesReady = track->framesReady(); 3058 } else if (track->isStopped()) { 3059 framesReady = 0; 3060 } else { 3061 framesReady = 1; 3062 } 3063 if ((framesReady >= minFrames) && track->isReady() && 3064 !track->isPaused() && !track->isTerminated()) 3065 { 3066 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3067 3068 mixedTracks++; 3069 3070 // track->mainBuffer() != mMixBuffer means there is an effect chain 3071 // connected to the track 3072 chain.clear(); 3073 if (track->mainBuffer() != mMixBuffer) { 3074 chain = getEffectChain_l(track->sessionId()); 3075 // Delegate volume control to effect in track effect chain if needed 3076 if (chain != 0) { 3077 tracksWithEffect++; 3078 } else { 3079 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3080 "session %d", 3081 name, track->sessionId()); 3082 } 3083 } 3084 3085 3086 int param = AudioMixer::VOLUME; 3087 if (track->mFillingUpStatus == Track::FS_FILLED) { 3088 // no ramp for the first volume setting 3089 track->mFillingUpStatus = Track::FS_ACTIVE; 3090 if (track->mState == TrackBase::RESUMING) { 3091 track->mState = TrackBase::ACTIVE; 3092 param = AudioMixer::RAMP_VOLUME; 3093 } 3094 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3095 // FIXME should not make a decision based on mServer 3096 } else if (cblk->mServer != 0) { 3097 // If the track is stopped before the first frame was mixed, 3098 // do not apply ramp 3099 param = AudioMixer::RAMP_VOLUME; 3100 } 3101 3102 // compute volume for this track 3103 uint32_t vl, vr, va; 3104 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3105 vl = vr = va = 0; 3106 if (track->isPausing()) { 3107 track->setPaused(); 3108 } 3109 } else { 3110 3111 // read original volumes with volume control 3112 float typeVolume = mStreamTypes[track->streamType()].volume; 3113 float v = masterVolume * typeVolume; 3114 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3115 uint32_t vlr = proxy->getVolumeLR(); 3116 vl = vlr & 0xFFFF; 3117 vr = vlr >> 16; 3118 // track volumes come from shared memory, so can't be trusted and must be clamped 3119 if (vl > MAX_GAIN_INT) { 3120 ALOGV("Track left volume out of range: %04X", vl); 3121 vl = MAX_GAIN_INT; 3122 } 3123 if (vr > MAX_GAIN_INT) { 3124 ALOGV("Track right volume out of range: %04X", vr); 3125 vr = MAX_GAIN_INT; 3126 } 3127 // now apply the master volume and stream type volume 3128 vl = (uint32_t)(v * vl) << 12; 3129 vr = (uint32_t)(v * vr) << 12; 3130 // assuming master volume and stream type volume each go up to 1.0, 3131 // vl and vr are now in 8.24 format 3132 3133 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3134 // send level comes from shared memory and so may be corrupt 3135 if (sendLevel > MAX_GAIN_INT) { 3136 ALOGV("Track send level out of range: %04X", sendLevel); 3137 sendLevel = MAX_GAIN_INT; 3138 } 3139 va = (uint32_t)(v * sendLevel); 3140 } 3141 3142 // Delegate volume control to effect in track effect chain if needed 3143 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3144 // Do not ramp volume if volume is controlled by effect 3145 param = AudioMixer::VOLUME; 3146 track->mHasVolumeController = true; 3147 } else { 3148 // force no volume ramp when volume controller was just disabled or removed 3149 // from effect chain to avoid volume spike 3150 if (track->mHasVolumeController) { 3151 param = AudioMixer::VOLUME; 3152 } 3153 track->mHasVolumeController = false; 3154 } 3155 3156 // Convert volumes from 8.24 to 4.12 format 3157 // This additional clamping is needed in case chain->setVolume_l() overshot 3158 vl = (vl + (1 << 11)) >> 12; 3159 if (vl > MAX_GAIN_INT) { 3160 vl = MAX_GAIN_INT; 3161 } 3162 vr = (vr + (1 << 11)) >> 12; 3163 if (vr > MAX_GAIN_INT) { 3164 vr = MAX_GAIN_INT; 3165 } 3166 3167 if (va > MAX_GAIN_INT) { 3168 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3169 } 3170 3171 // XXX: these things DON'T need to be done each time 3172 mAudioMixer->setBufferProvider(name, track); 3173 mAudioMixer->enable(name); 3174 3175 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3176 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3177 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3178 mAudioMixer->setParameter( 3179 name, 3180 AudioMixer::TRACK, 3181 AudioMixer::FORMAT, (void *)track->format()); 3182 mAudioMixer->setParameter( 3183 name, 3184 AudioMixer::TRACK, 3185 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3186 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3187 uint32_t maxSampleRate = mSampleRate * 2; 3188 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3189 if (reqSampleRate == 0) { 3190 reqSampleRate = mSampleRate; 3191 } else if (reqSampleRate > maxSampleRate) { 3192 reqSampleRate = maxSampleRate; 3193 } 3194 mAudioMixer->setParameter( 3195 name, 3196 AudioMixer::RESAMPLE, 3197 AudioMixer::SAMPLE_RATE, 3198 (void *)reqSampleRate); 3199 mAudioMixer->setParameter( 3200 name, 3201 AudioMixer::TRACK, 3202 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3203 mAudioMixer->setParameter( 3204 name, 3205 AudioMixer::TRACK, 3206 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3207 3208 // reset retry count 3209 track->mRetryCount = kMaxTrackRetries; 3210 3211 // If one track is ready, set the mixer ready if: 3212 // - the mixer was not ready during previous round OR 3213 // - no other track is not ready 3214 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3215 mixerStatus != MIXER_TRACKS_ENABLED) { 3216 mixerStatus = MIXER_TRACKS_READY; 3217 } 3218 } else { 3219 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3220 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3221 } 3222 // clear effect chain input buffer if an active track underruns to avoid sending 3223 // previous audio buffer again to effects 3224 chain = getEffectChain_l(track->sessionId()); 3225 if (chain != 0) { 3226 chain->clearInputBuffer(); 3227 } 3228 3229 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3230 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3231 track->isStopped() || track->isPaused()) { 3232 // We have consumed all the buffers of this track. 3233 // Remove it from the list of active tracks. 3234 // TODO: use actual buffer filling status instead of latency when available from 3235 // audio HAL 3236 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3237 size_t framesWritten = mBytesWritten / mFrameSize; 3238 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3239 if (track->isStopped()) { 3240 track->reset(); 3241 } 3242 tracksToRemove->add(track); 3243 } 3244 } else { 3245 // No buffers for this track. Give it a few chances to 3246 // fill a buffer, then remove it from active list. 3247 if (--(track->mRetryCount) <= 0) { 3248 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3249 tracksToRemove->add(track); 3250 // indicate to client process that the track was disabled because of underrun; 3251 // it will then automatically call start() when data is available 3252 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3253 // If one track is not ready, mark the mixer also not ready if: 3254 // - the mixer was ready during previous round OR 3255 // - no other track is ready 3256 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3257 mixerStatus != MIXER_TRACKS_READY) { 3258 mixerStatus = MIXER_TRACKS_ENABLED; 3259 } 3260 } 3261 mAudioMixer->disable(name); 3262 } 3263 3264 } // local variable scope to avoid goto warning 3265track_is_ready: ; 3266 3267 } 3268 3269 // Push the new FastMixer state if necessary 3270 bool pauseAudioWatchdog = false; 3271 if (didModify) { 3272 state->mFastTracksGen++; 3273 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3274 if (kUseFastMixer == FastMixer_Dynamic && 3275 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3276 state->mCommand = FastMixerState::COLD_IDLE; 3277 state->mColdFutexAddr = &mFastMixerFutex; 3278 state->mColdGen++; 3279 mFastMixerFutex = 0; 3280 if (kUseFastMixer == FastMixer_Dynamic) { 3281 mNormalSink = mOutputSink; 3282 } 3283 // If we go into cold idle, need to wait for acknowledgement 3284 // so that fast mixer stops doing I/O. 3285 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3286 pauseAudioWatchdog = true; 3287 } 3288 } 3289 if (sq != NULL) { 3290 sq->end(didModify); 3291 sq->push(block); 3292 } 3293#ifdef AUDIO_WATCHDOG 3294 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3295 mAudioWatchdog->pause(); 3296 } 3297#endif 3298 3299 // Now perform the deferred reset on fast tracks that have stopped 3300 while (resetMask != 0) { 3301 size_t i = __builtin_ctz(resetMask); 3302 ALOG_ASSERT(i < count); 3303 resetMask &= ~(1 << i); 3304 sp<Track> t = mActiveTracks[i].promote(); 3305 if (t == 0) { 3306 continue; 3307 } 3308 Track* track = t.get(); 3309 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3310 track->reset(); 3311 } 3312 3313 // remove all the tracks that need to be... 3314 removeTracks_l(*tracksToRemove); 3315 3316 // mix buffer must be cleared if all tracks are connected to an 3317 // effect chain as in this case the mixer will not write to 3318 // mix buffer and track effects will accumulate into it 3319 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3320 (mixedTracks == 0 && fastTracks > 0))) { 3321 // FIXME as a performance optimization, should remember previous zero status 3322 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3323 } 3324 3325 // if any fast tracks, then status is ready 3326 mMixerStatusIgnoringFastTracks = mixerStatus; 3327 if (fastTracks > 0) { 3328 mixerStatus = MIXER_TRACKS_READY; 3329 } 3330 return mixerStatus; 3331} 3332 3333// getTrackName_l() must be called with ThreadBase::mLock held 3334int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3335{ 3336 return mAudioMixer->getTrackName(channelMask, sessionId); 3337} 3338 3339// deleteTrackName_l() must be called with ThreadBase::mLock held 3340void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3341{ 3342 ALOGV("remove track (%d) and delete from mixer", name); 3343 mAudioMixer->deleteTrackName(name); 3344} 3345 3346// checkForNewParameters_l() must be called with ThreadBase::mLock held 3347bool AudioFlinger::MixerThread::checkForNewParameters_l() 3348{ 3349 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3350 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3351 bool reconfig = false; 3352 3353 while (!mNewParameters.isEmpty()) { 3354 3355 if (mFastMixer != NULL) { 3356 FastMixerStateQueue *sq = mFastMixer->sq(); 3357 FastMixerState *state = sq->begin(); 3358 if (!(state->mCommand & FastMixerState::IDLE)) { 3359 previousCommand = state->mCommand; 3360 state->mCommand = FastMixerState::HOT_IDLE; 3361 sq->end(); 3362 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3363 } else { 3364 sq->end(false /*didModify*/); 3365 } 3366 } 3367 3368 status_t status = NO_ERROR; 3369 String8 keyValuePair = mNewParameters[0]; 3370 AudioParameter param = AudioParameter(keyValuePair); 3371 int value; 3372 3373 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3374 reconfig = true; 3375 } 3376 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3377 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3378 status = BAD_VALUE; 3379 } else { 3380 // no need to save value, since it's constant 3381 reconfig = true; 3382 } 3383 } 3384 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3385 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3386 status = BAD_VALUE; 3387 } else { 3388 // no need to save value, since it's constant 3389 reconfig = true; 3390 } 3391 } 3392 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3393 // do not accept frame count changes if tracks are open as the track buffer 3394 // size depends on frame count and correct behavior would not be guaranteed 3395 // if frame count is changed after track creation 3396 if (!mTracks.isEmpty()) { 3397 status = INVALID_OPERATION; 3398 } else { 3399 reconfig = true; 3400 } 3401 } 3402 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3403#ifdef ADD_BATTERY_DATA 3404 // when changing the audio output device, call addBatteryData to notify 3405 // the change 3406 if (mOutDevice != value) { 3407 uint32_t params = 0; 3408 // check whether speaker is on 3409 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3410 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3411 } 3412 3413 audio_devices_t deviceWithoutSpeaker 3414 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3415 // check if any other device (except speaker) is on 3416 if (value & deviceWithoutSpeaker ) { 3417 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3418 } 3419 3420 if (params != 0) { 3421 addBatteryData(params); 3422 } 3423 } 3424#endif 3425 3426 // forward device change to effects that have requested to be 3427 // aware of attached audio device. 3428 if (value != AUDIO_DEVICE_NONE) { 3429 mOutDevice = value; 3430 for (size_t i = 0; i < mEffectChains.size(); i++) { 3431 mEffectChains[i]->setDevice_l(mOutDevice); 3432 } 3433 } 3434 } 3435 3436 if (status == NO_ERROR) { 3437 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3438 keyValuePair.string()); 3439 if (!mStandby && status == INVALID_OPERATION) { 3440 mOutput->stream->common.standby(&mOutput->stream->common); 3441 mStandby = true; 3442 mBytesWritten = 0; 3443 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3444 keyValuePair.string()); 3445 } 3446 if (status == NO_ERROR && reconfig) { 3447 readOutputParameters(); 3448 delete mAudioMixer; 3449 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3450 for (size_t i = 0; i < mTracks.size() ; i++) { 3451 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3452 if (name < 0) { 3453 break; 3454 } 3455 mTracks[i]->mName = name; 3456 } 3457 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3458 } 3459 } 3460 3461 mNewParameters.removeAt(0); 3462 3463 mParamStatus = status; 3464 mParamCond.signal(); 3465 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3466 // already timed out waiting for the status and will never signal the condition. 3467 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3468 } 3469 3470 if (!(previousCommand & FastMixerState::IDLE)) { 3471 ALOG_ASSERT(mFastMixer != NULL); 3472 FastMixerStateQueue *sq = mFastMixer->sq(); 3473 FastMixerState *state = sq->begin(); 3474 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3475 state->mCommand = previousCommand; 3476 sq->end(); 3477 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3478 } 3479 3480 return reconfig; 3481} 3482 3483 3484void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3485{ 3486 const size_t SIZE = 256; 3487 char buffer[SIZE]; 3488 String8 result; 3489 3490 PlaybackThread::dumpInternals(fd, args); 3491 3492 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3493 result.append(buffer); 3494 write(fd, result.string(), result.size()); 3495 3496 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3497 const FastMixerDumpState copy(mFastMixerDumpState); 3498 copy.dump(fd); 3499 3500#ifdef STATE_QUEUE_DUMP 3501 // Similar for state queue 3502 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3503 observerCopy.dump(fd); 3504 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3505 mutatorCopy.dump(fd); 3506#endif 3507 3508#ifdef TEE_SINK 3509 // Write the tee output to a .wav file 3510 dumpTee(fd, mTeeSource, mId); 3511#endif 3512 3513#ifdef AUDIO_WATCHDOG 3514 if (mAudioWatchdog != 0) { 3515 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3516 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3517 wdCopy.dump(fd); 3518 } 3519#endif 3520} 3521 3522uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3523{ 3524 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3525} 3526 3527uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3528{ 3529 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3530} 3531 3532void AudioFlinger::MixerThread::cacheParameters_l() 3533{ 3534 PlaybackThread::cacheParameters_l(); 3535 3536 // FIXME: Relaxed timing because of a certain device that can't meet latency 3537 // Should be reduced to 2x after the vendor fixes the driver issue 3538 // increase threshold again due to low power audio mode. The way this warning 3539 // threshold is calculated and its usefulness should be reconsidered anyway. 3540 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3541} 3542 3543// ---------------------------------------------------------------------------- 3544 3545AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3546 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3547 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3548 // mLeftVolFloat, mRightVolFloat 3549{ 3550} 3551 3552AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3553 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3554 ThreadBase::type_t type) 3555 : PlaybackThread(audioFlinger, output, id, device, type) 3556 // mLeftVolFloat, mRightVolFloat 3557{ 3558} 3559 3560AudioFlinger::DirectOutputThread::~DirectOutputThread() 3561{ 3562} 3563 3564void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3565{ 3566 audio_track_cblk_t* cblk = track->cblk(); 3567 float left, right; 3568 3569 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3570 left = right = 0; 3571 } else { 3572 float typeVolume = mStreamTypes[track->streamType()].volume; 3573 float v = mMasterVolume * typeVolume; 3574 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3575 uint32_t vlr = proxy->getVolumeLR(); 3576 float v_clamped = v * (vlr & 0xFFFF); 3577 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3578 left = v_clamped/MAX_GAIN; 3579 v_clamped = v * (vlr >> 16); 3580 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3581 right = v_clamped/MAX_GAIN; 3582 } 3583 3584 if (lastTrack) { 3585 if (left != mLeftVolFloat || right != mRightVolFloat) { 3586 mLeftVolFloat = left; 3587 mRightVolFloat = right; 3588 3589 // Convert volumes from float to 8.24 3590 uint32_t vl = (uint32_t)(left * (1 << 24)); 3591 uint32_t vr = (uint32_t)(right * (1 << 24)); 3592 3593 // Delegate volume control to effect in track effect chain if needed 3594 // only one effect chain can be present on DirectOutputThread, so if 3595 // there is one, the track is connected to it 3596 if (!mEffectChains.isEmpty()) { 3597 mEffectChains[0]->setVolume_l(&vl, &vr); 3598 left = (float)vl / (1 << 24); 3599 right = (float)vr / (1 << 24); 3600 } 3601 if (mOutput->stream->set_volume) { 3602 mOutput->stream->set_volume(mOutput->stream, left, right); 3603 } 3604 } 3605 } 3606} 3607 3608 3609AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3610 Vector< sp<Track> > *tracksToRemove 3611) 3612{ 3613 size_t count = mActiveTracks.size(); 3614 mixer_state mixerStatus = MIXER_IDLE; 3615 3616 // find out which tracks need to be processed 3617 for (size_t i = 0; i < count; i++) { 3618 sp<Track> t = mActiveTracks[i].promote(); 3619 // The track died recently 3620 if (t == 0) { 3621 continue; 3622 } 3623 3624 Track* const track = t.get(); 3625 audio_track_cblk_t* cblk = track->cblk(); 3626 // Only consider last track started for volume and mixer state control. 3627 // In theory an older track could underrun and restart after the new one starts 3628 // but as we only care about the transition phase between two tracks on a 3629 // direct output, it is not a problem to ignore the underrun case. 3630 sp<Track> l = mLatestActiveTrack.promote(); 3631 bool last = l.get() == track; 3632 3633 // The first time a track is added we wait 3634 // for all its buffers to be filled before processing it 3635 uint32_t minFrames; 3636 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3637 minFrames = mNormalFrameCount; 3638 } else { 3639 minFrames = 1; 3640 } 3641 3642 if ((track->framesReady() >= minFrames) && track->isReady() && 3643 !track->isPaused() && !track->isTerminated()) 3644 { 3645 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3646 3647 if (track->mFillingUpStatus == Track::FS_FILLED) { 3648 track->mFillingUpStatus = Track::FS_ACTIVE; 3649 // make sure processVolume_l() will apply new volume even if 0 3650 mLeftVolFloat = mRightVolFloat = -1.0; 3651 if (track->mState == TrackBase::RESUMING) { 3652 track->mState = TrackBase::ACTIVE; 3653 } 3654 } 3655 3656 // compute volume for this track 3657 processVolume_l(track, last); 3658 if (last) { 3659 // reset retry count 3660 track->mRetryCount = kMaxTrackRetriesDirect; 3661 mActiveTrack = t; 3662 mixerStatus = MIXER_TRACKS_READY; 3663 } 3664 } else { 3665 // clear effect chain input buffer if the last active track started underruns 3666 // to avoid sending previous audio buffer again to effects 3667 if (!mEffectChains.isEmpty() && last) { 3668 mEffectChains[0]->clearInputBuffer(); 3669 } 3670 3671 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3672 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3673 track->isStopped() || track->isPaused()) { 3674 // We have consumed all the buffers of this track. 3675 // Remove it from the list of active tracks. 3676 // TODO: implement behavior for compressed audio 3677 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3678 size_t framesWritten = mBytesWritten / mFrameSize; 3679 if (mStandby || !last || 3680 track->presentationComplete(framesWritten, audioHALFrames)) { 3681 if (track->isStopped()) { 3682 track->reset(); 3683 } 3684 tracksToRemove->add(track); 3685 } 3686 } else { 3687 // No buffers for this track. Give it a few chances to 3688 // fill a buffer, then remove it from active list. 3689 // Only consider last track started for mixer state control 3690 if (--(track->mRetryCount) <= 0) { 3691 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3692 tracksToRemove->add(track); 3693 // indicate to client process that the track was disabled because of underrun; 3694 // it will then automatically call start() when data is available 3695 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3696 } else if (last) { 3697 mixerStatus = MIXER_TRACKS_ENABLED; 3698 } 3699 } 3700 } 3701 } 3702 3703 // remove all the tracks that need to be... 3704 removeTracks_l(*tracksToRemove); 3705 3706 return mixerStatus; 3707} 3708 3709void AudioFlinger::DirectOutputThread::threadLoop_mix() 3710{ 3711 size_t frameCount = mFrameCount; 3712 int8_t *curBuf = (int8_t *)mMixBuffer; 3713 // output audio to hardware 3714 while (frameCount) { 3715 AudioBufferProvider::Buffer buffer; 3716 buffer.frameCount = frameCount; 3717 mActiveTrack->getNextBuffer(&buffer); 3718 if (buffer.raw == NULL) { 3719 memset(curBuf, 0, frameCount * mFrameSize); 3720 break; 3721 } 3722 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3723 frameCount -= buffer.frameCount; 3724 curBuf += buffer.frameCount * mFrameSize; 3725 mActiveTrack->releaseBuffer(&buffer); 3726 } 3727 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3728 sleepTime = 0; 3729 standbyTime = systemTime() + standbyDelay; 3730 mActiveTrack.clear(); 3731} 3732 3733void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3734{ 3735 if (sleepTime == 0) { 3736 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3737 sleepTime = activeSleepTime; 3738 } else { 3739 sleepTime = idleSleepTime; 3740 } 3741 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3742 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3743 sleepTime = 0; 3744 } 3745} 3746 3747// getTrackName_l() must be called with ThreadBase::mLock held 3748int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3749 int sessionId) 3750{ 3751 return 0; 3752} 3753 3754// deleteTrackName_l() must be called with ThreadBase::mLock held 3755void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3756{ 3757} 3758 3759// checkForNewParameters_l() must be called with ThreadBase::mLock held 3760bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3761{ 3762 bool reconfig = false; 3763 3764 while (!mNewParameters.isEmpty()) { 3765 status_t status = NO_ERROR; 3766 String8 keyValuePair = mNewParameters[0]; 3767 AudioParameter param = AudioParameter(keyValuePair); 3768 int value; 3769 3770 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3771 // do not accept frame count changes if tracks are open as the track buffer 3772 // size depends on frame count and correct behavior would not be garantied 3773 // if frame count is changed after track creation 3774 if (!mTracks.isEmpty()) { 3775 status = INVALID_OPERATION; 3776 } else { 3777 reconfig = true; 3778 } 3779 } 3780 if (status == NO_ERROR) { 3781 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3782 keyValuePair.string()); 3783 if (!mStandby && status == INVALID_OPERATION) { 3784 mOutput->stream->common.standby(&mOutput->stream->common); 3785 mStandby = true; 3786 mBytesWritten = 0; 3787 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3788 keyValuePair.string()); 3789 } 3790 if (status == NO_ERROR && reconfig) { 3791 readOutputParameters(); 3792 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3793 } 3794 } 3795 3796 mNewParameters.removeAt(0); 3797 3798 mParamStatus = status; 3799 mParamCond.signal(); 3800 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3801 // already timed out waiting for the status and will never signal the condition. 3802 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3803 } 3804 return reconfig; 3805} 3806 3807uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3808{ 3809 uint32_t time; 3810 if (audio_is_linear_pcm(mFormat)) { 3811 time = PlaybackThread::activeSleepTimeUs(); 3812 } else { 3813 time = 10000; 3814 } 3815 return time; 3816} 3817 3818uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3819{ 3820 uint32_t time; 3821 if (audio_is_linear_pcm(mFormat)) { 3822 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3823 } else { 3824 time = 10000; 3825 } 3826 return time; 3827} 3828 3829uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3830{ 3831 uint32_t time; 3832 if (audio_is_linear_pcm(mFormat)) { 3833 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3834 } else { 3835 time = 10000; 3836 } 3837 return time; 3838} 3839 3840void AudioFlinger::DirectOutputThread::cacheParameters_l() 3841{ 3842 PlaybackThread::cacheParameters_l(); 3843 3844 // use shorter standby delay as on normal output to release 3845 // hardware resources as soon as possible 3846 if (audio_is_linear_pcm(mFormat)) { 3847 standbyDelay = microseconds(activeSleepTime*2); 3848 } else { 3849 standbyDelay = kOffloadStandbyDelayNs; 3850 } 3851} 3852 3853// ---------------------------------------------------------------------------- 3854 3855AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3856 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3857 : Thread(false /*canCallJava*/), 3858 mPlaybackThread(playbackThread), 3859 mWriteAckSequence(0), 3860 mDrainSequence(0) 3861{ 3862} 3863 3864AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3865{ 3866} 3867 3868void AudioFlinger::AsyncCallbackThread::onFirstRef() 3869{ 3870 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3871} 3872 3873bool AudioFlinger::AsyncCallbackThread::threadLoop() 3874{ 3875 while (!exitPending()) { 3876 uint32_t writeAckSequence; 3877 uint32_t drainSequence; 3878 3879 { 3880 Mutex::Autolock _l(mLock); 3881 while (!((mWriteAckSequence & 1) || 3882 (mDrainSequence & 1) || 3883 exitPending())) { 3884 mWaitWorkCV.wait(mLock); 3885 } 3886 3887 if (exitPending()) { 3888 break; 3889 } 3890 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3891 mWriteAckSequence, mDrainSequence); 3892 writeAckSequence = mWriteAckSequence; 3893 mWriteAckSequence &= ~1; 3894 drainSequence = mDrainSequence; 3895 mDrainSequence &= ~1; 3896 } 3897 { 3898 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3899 if (playbackThread != 0) { 3900 if (writeAckSequence & 1) { 3901 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3902 } 3903 if (drainSequence & 1) { 3904 playbackThread->resetDraining(drainSequence >> 1); 3905 } 3906 } 3907 } 3908 } 3909 return false; 3910} 3911 3912void AudioFlinger::AsyncCallbackThread::exit() 3913{ 3914 ALOGV("AsyncCallbackThread::exit"); 3915 Mutex::Autolock _l(mLock); 3916 requestExit(); 3917 mWaitWorkCV.broadcast(); 3918} 3919 3920void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3921{ 3922 Mutex::Autolock _l(mLock); 3923 // bit 0 is cleared 3924 mWriteAckSequence = sequence << 1; 3925} 3926 3927void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3928{ 3929 Mutex::Autolock _l(mLock); 3930 // ignore unexpected callbacks 3931 if (mWriteAckSequence & 2) { 3932 mWriteAckSequence |= 1; 3933 mWaitWorkCV.signal(); 3934 } 3935} 3936 3937void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3938{ 3939 Mutex::Autolock _l(mLock); 3940 // bit 0 is cleared 3941 mDrainSequence = sequence << 1; 3942} 3943 3944void AudioFlinger::AsyncCallbackThread::resetDraining() 3945{ 3946 Mutex::Autolock _l(mLock); 3947 // ignore unexpected callbacks 3948 if (mDrainSequence & 2) { 3949 mDrainSequence |= 1; 3950 mWaitWorkCV.signal(); 3951 } 3952} 3953 3954 3955// ---------------------------------------------------------------------------- 3956AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3957 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3958 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3959 mHwPaused(false), 3960 mFlushPending(false), 3961 mPausedBytesRemaining(0) 3962{ 3963 //FIXME: mStandby should be set to true by ThreadBase constructor 3964 mStandby = true; 3965} 3966 3967void AudioFlinger::OffloadThread::threadLoop_exit() 3968{ 3969 if (mFlushPending || mHwPaused) { 3970 // If a flush is pending or track was paused, just discard buffered data 3971 flushHw_l(); 3972 } else { 3973 mMixerStatus = MIXER_DRAIN_ALL; 3974 threadLoop_drain(); 3975 } 3976 mCallbackThread->exit(); 3977 PlaybackThread::threadLoop_exit(); 3978} 3979 3980AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3981 Vector< sp<Track> > *tracksToRemove 3982) 3983{ 3984 size_t count = mActiveTracks.size(); 3985 3986 mixer_state mixerStatus = MIXER_IDLE; 3987 bool doHwPause = false; 3988 bool doHwResume = false; 3989 3990 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3991 3992 // find out which tracks need to be processed 3993 for (size_t i = 0; i < count; i++) { 3994 sp<Track> t = mActiveTracks[i].promote(); 3995 // The track died recently 3996 if (t == 0) { 3997 continue; 3998 } 3999 Track* const track = t.get(); 4000 audio_track_cblk_t* cblk = track->cblk(); 4001 // Only consider last track started for volume and mixer state control. 4002 // In theory an older track could underrun and restart after the new one starts 4003 // but as we only care about the transition phase between two tracks on a 4004 // direct output, it is not a problem to ignore the underrun case. 4005 sp<Track> l = mLatestActiveTrack.promote(); 4006 bool last = l.get() == track; 4007 4008 if (track->isPausing()) { 4009 track->setPaused(); 4010 if (last) { 4011 if (!mHwPaused) { 4012 doHwPause = true; 4013 mHwPaused = true; 4014 } 4015 // If we were part way through writing the mixbuffer to 4016 // the HAL we must save this until we resume 4017 // BUG - this will be wrong if a different track is made active, 4018 // in that case we want to discard the pending data in the 4019 // mixbuffer and tell the client to present it again when the 4020 // track is resumed 4021 mPausedWriteLength = mCurrentWriteLength; 4022 mPausedBytesRemaining = mBytesRemaining; 4023 mBytesRemaining = 0; // stop writing 4024 } 4025 tracksToRemove->add(track); 4026 } else if (track->framesReady() && track->isReady() && 4027 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4028 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4029 if (track->mFillingUpStatus == Track::FS_FILLED) { 4030 track->mFillingUpStatus = Track::FS_ACTIVE; 4031 // make sure processVolume_l() will apply new volume even if 0 4032 mLeftVolFloat = mRightVolFloat = -1.0; 4033 if (track->mState == TrackBase::RESUMING) { 4034 track->mState = TrackBase::ACTIVE; 4035 if (last) { 4036 if (mPausedBytesRemaining) { 4037 // Need to continue write that was interrupted 4038 mCurrentWriteLength = mPausedWriteLength; 4039 mBytesRemaining = mPausedBytesRemaining; 4040 mPausedBytesRemaining = 0; 4041 } 4042 if (mHwPaused) { 4043 doHwResume = true; 4044 mHwPaused = false; 4045 // threadLoop_mix() will handle the case that we need to 4046 // resume an interrupted write 4047 } 4048 // enable write to audio HAL 4049 sleepTime = 0; 4050 } 4051 } 4052 } 4053 4054 if (last) { 4055 sp<Track> previousTrack = mPreviousTrack.promote(); 4056 if (previousTrack != 0) { 4057 if (track != previousTrack.get()) { 4058 // Flush any data still being written from last track 4059 mBytesRemaining = 0; 4060 if (mPausedBytesRemaining) { 4061 // Last track was paused so we also need to flush saved 4062 // mixbuffer state and invalidate track so that it will 4063 // re-submit that unwritten data when it is next resumed 4064 mPausedBytesRemaining = 0; 4065 // Invalidate is a bit drastic - would be more efficient 4066 // to have a flag to tell client that some of the 4067 // previously written data was lost 4068 previousTrack->invalidate(); 4069 } 4070 // flush data already sent to the DSP if changing audio session as audio 4071 // comes from a different source. Also invalidate previous track to force a 4072 // seek when resuming. 4073 if (previousTrack->sessionId() != track->sessionId()) { 4074 previousTrack->invalidate(); 4075 mFlushPending = true; 4076 } 4077 } 4078 } 4079 mPreviousTrack = track; 4080 // reset retry count 4081 track->mRetryCount = kMaxTrackRetriesOffload; 4082 mActiveTrack = t; 4083 mixerStatus = MIXER_TRACKS_READY; 4084 } 4085 } else { 4086 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4087 if (track->isStopping_1()) { 4088 // Hardware buffer can hold a large amount of audio so we must 4089 // wait for all current track's data to drain before we say 4090 // that the track is stopped. 4091 if (mBytesRemaining == 0) { 4092 // Only start draining when all data in mixbuffer 4093 // has been written 4094 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4095 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4096 // do not drain if no data was ever sent to HAL (mStandby == true) 4097 if (last && !mStandby) { 4098 // do not modify drain sequence if we are already draining. This happens 4099 // when resuming from pause after drain. 4100 if ((mDrainSequence & 1) == 0) { 4101 sleepTime = 0; 4102 standbyTime = systemTime() + standbyDelay; 4103 mixerStatus = MIXER_DRAIN_TRACK; 4104 mDrainSequence += 2; 4105 } 4106 if (mHwPaused) { 4107 // It is possible to move from PAUSED to STOPPING_1 without 4108 // a resume so we must ensure hardware is running 4109 doHwResume = true; 4110 mHwPaused = false; 4111 } 4112 } 4113 } 4114 } else if (track->isStopping_2()) { 4115 // Drain has completed or we are in standby, signal presentation complete 4116 if (!(mDrainSequence & 1) || !last || mStandby) { 4117 track->mState = TrackBase::STOPPED; 4118 size_t audioHALFrames = 4119 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4120 size_t framesWritten = 4121 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4122 track->presentationComplete(framesWritten, audioHALFrames); 4123 track->reset(); 4124 tracksToRemove->add(track); 4125 } 4126 } else { 4127 // No buffers for this track. Give it a few chances to 4128 // fill a buffer, then remove it from active list. 4129 if (--(track->mRetryCount) <= 0) { 4130 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4131 track->name()); 4132 tracksToRemove->add(track); 4133 // indicate to client process that the track was disabled because of underrun; 4134 // it will then automatically call start() when data is available 4135 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4136 } else if (last){ 4137 mixerStatus = MIXER_TRACKS_ENABLED; 4138 } 4139 } 4140 } 4141 // compute volume for this track 4142 processVolume_l(track, last); 4143 } 4144 4145 // make sure the pause/flush/resume sequence is executed in the right order. 4146 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4147 // before flush and then resume HW. This can happen in case of pause/flush/resume 4148 // if resume is received before pause is executed. 4149 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4150 mOutput->stream->pause(mOutput->stream); 4151 if (!doHwPause) { 4152 doHwResume = true; 4153 } 4154 } 4155 if (mFlushPending) { 4156 flushHw_l(); 4157 mFlushPending = false; 4158 } 4159 if (!mStandby && doHwResume) { 4160 mOutput->stream->resume(mOutput->stream); 4161 } 4162 4163 // remove all the tracks that need to be... 4164 removeTracks_l(*tracksToRemove); 4165 4166 return mixerStatus; 4167} 4168 4169void AudioFlinger::OffloadThread::flushOutput_l() 4170{ 4171 mFlushPending = true; 4172} 4173 4174// must be called with thread mutex locked 4175bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4176{ 4177 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4178 mWriteAckSequence, mDrainSequence); 4179 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4180 return true; 4181 } 4182 return false; 4183} 4184 4185// must be called with thread mutex locked 4186bool AudioFlinger::OffloadThread::shouldStandby_l() 4187{ 4188 bool trackPaused = false; 4189 4190 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4191 // after a timeout and we will enter standby then. 4192 if (mTracks.size() > 0) { 4193 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4194 } 4195 4196 return !mStandby && !trackPaused; 4197} 4198 4199 4200bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4201{ 4202 Mutex::Autolock _l(mLock); 4203 return waitingAsyncCallback_l(); 4204} 4205 4206void AudioFlinger::OffloadThread::flushHw_l() 4207{ 4208 mOutput->stream->flush(mOutput->stream); 4209 // Flush anything still waiting in the mixbuffer 4210 mCurrentWriteLength = 0; 4211 mBytesRemaining = 0; 4212 mPausedWriteLength = 0; 4213 mPausedBytesRemaining = 0; 4214 if (mUseAsyncWrite) { 4215 // discard any pending drain or write ack by incrementing sequence 4216 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4217 mDrainSequence = (mDrainSequence + 2) & ~1; 4218 ALOG_ASSERT(mCallbackThread != 0); 4219 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4220 mCallbackThread->setDraining(mDrainSequence); 4221 } 4222} 4223 4224// ---------------------------------------------------------------------------- 4225 4226AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4227 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4228 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4229 DUPLICATING), 4230 mWaitTimeMs(UINT_MAX) 4231{ 4232 addOutputTrack(mainThread); 4233} 4234 4235AudioFlinger::DuplicatingThread::~DuplicatingThread() 4236{ 4237 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4238 mOutputTracks[i]->destroy(); 4239 } 4240} 4241 4242void AudioFlinger::DuplicatingThread::threadLoop_mix() 4243{ 4244 // mix buffers... 4245 if (outputsReady(outputTracks)) { 4246 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4247 } else { 4248 memset(mMixBuffer, 0, mixBufferSize); 4249 } 4250 sleepTime = 0; 4251 writeFrames = mNormalFrameCount; 4252 mCurrentWriteLength = mixBufferSize; 4253 standbyTime = systemTime() + standbyDelay; 4254} 4255 4256void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4257{ 4258 if (sleepTime == 0) { 4259 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4260 sleepTime = activeSleepTime; 4261 } else { 4262 sleepTime = idleSleepTime; 4263 } 4264 } else if (mBytesWritten != 0) { 4265 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4266 writeFrames = mNormalFrameCount; 4267 memset(mMixBuffer, 0, mixBufferSize); 4268 } else { 4269 // flush remaining overflow buffers in output tracks 4270 writeFrames = 0; 4271 } 4272 sleepTime = 0; 4273 } 4274} 4275 4276ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4277{ 4278 for (size_t i = 0; i < outputTracks.size(); i++) { 4279 outputTracks[i]->write(mMixBuffer, writeFrames); 4280 } 4281 mStandby = false; 4282 return (ssize_t)mixBufferSize; 4283} 4284 4285void AudioFlinger::DuplicatingThread::threadLoop_standby() 4286{ 4287 // DuplicatingThread implements standby by stopping all tracks 4288 for (size_t i = 0; i < outputTracks.size(); i++) { 4289 outputTracks[i]->stop(); 4290 } 4291} 4292 4293void AudioFlinger::DuplicatingThread::saveOutputTracks() 4294{ 4295 outputTracks = mOutputTracks; 4296} 4297 4298void AudioFlinger::DuplicatingThread::clearOutputTracks() 4299{ 4300 outputTracks.clear(); 4301} 4302 4303void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4304{ 4305 Mutex::Autolock _l(mLock); 4306 // FIXME explain this formula 4307 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4308 OutputTrack *outputTrack = new OutputTrack(thread, 4309 this, 4310 mSampleRate, 4311 mFormat, 4312 mChannelMask, 4313 frameCount, 4314 IPCThreadState::self()->getCallingUid()); 4315 if (outputTrack->cblk() != NULL) { 4316 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4317 mOutputTracks.add(outputTrack); 4318 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4319 updateWaitTime_l(); 4320 } 4321} 4322 4323void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4324{ 4325 Mutex::Autolock _l(mLock); 4326 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4327 if (mOutputTracks[i]->thread() == thread) { 4328 mOutputTracks[i]->destroy(); 4329 mOutputTracks.removeAt(i); 4330 updateWaitTime_l(); 4331 return; 4332 } 4333 } 4334 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4335} 4336 4337// caller must hold mLock 4338void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4339{ 4340 mWaitTimeMs = UINT_MAX; 4341 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4342 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4343 if (strong != 0) { 4344 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4345 if (waitTimeMs < mWaitTimeMs) { 4346 mWaitTimeMs = waitTimeMs; 4347 } 4348 } 4349 } 4350} 4351 4352 4353bool AudioFlinger::DuplicatingThread::outputsReady( 4354 const SortedVector< sp<OutputTrack> > &outputTracks) 4355{ 4356 for (size_t i = 0; i < outputTracks.size(); i++) { 4357 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4358 if (thread == 0) { 4359 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4360 outputTracks[i].get()); 4361 return false; 4362 } 4363 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4364 // see note at standby() declaration 4365 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4366 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4367 thread.get()); 4368 return false; 4369 } 4370 } 4371 return true; 4372} 4373 4374uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4375{ 4376 return (mWaitTimeMs * 1000) / 2; 4377} 4378 4379void AudioFlinger::DuplicatingThread::cacheParameters_l() 4380{ 4381 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4382 updateWaitTime_l(); 4383 4384 MixerThread::cacheParameters_l(); 4385} 4386 4387// ---------------------------------------------------------------------------- 4388// Record 4389// ---------------------------------------------------------------------------- 4390 4391AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4392 AudioStreamIn *input, 4393 uint32_t sampleRate, 4394 audio_channel_mask_t channelMask, 4395 audio_io_handle_t id, 4396 audio_devices_t outDevice, 4397 audio_devices_t inDevice 4398#ifdef TEE_SINK 4399 , const sp<NBAIO_Sink>& teeSink 4400#endif 4401 ) : 4402 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4403 mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4404 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4405 // are set by readInputParameters() 4406 // mRsmpInIndex LEGACY 4407 mReqChannelCount(popcount(channelMask)), 4408 mReqSampleRate(sampleRate) 4409 // mBytesRead is only meaningful while active, and so is cleared in start() 4410 // (but might be better to also clear here for dump?) 4411#ifdef TEE_SINK 4412 , mTeeSink(teeSink) 4413#endif 4414{ 4415 snprintf(mName, kNameLength, "AudioIn_%X", id); 4416 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4417 4418 readInputParameters(); 4419} 4420 4421 4422AudioFlinger::RecordThread::~RecordThread() 4423{ 4424 mAudioFlinger->unregisterWriter(mNBLogWriter); 4425 delete[] mRsmpInBuffer; 4426 delete mResampler; 4427 delete[] mRsmpOutBuffer; 4428} 4429 4430void AudioFlinger::RecordThread::onFirstRef() 4431{ 4432 run(mName, PRIORITY_URGENT_AUDIO); 4433} 4434 4435bool AudioFlinger::RecordThread::threadLoop() 4436{ 4437 nsecs_t lastWarning = 0; 4438 4439 inputStandBy(); 4440 4441 // used to verify we've read at least once before evaluating how many bytes were read 4442 bool readOnce = false; 4443 4444 // used to request a deferred sleep, to be executed later while mutex is unlocked 4445 bool doSleep = false; 4446 4447reacquire_wakelock: 4448 sp<RecordTrack> activeTrack; 4449 int activeTracksGen; 4450 { 4451 Mutex::Autolock _l(mLock); 4452 size_t size = mActiveTracks.size(); 4453 activeTracksGen = mActiveTracksGen; 4454 if (size > 0) { 4455 // FIXME an arbitrary choice 4456 activeTrack = mActiveTracks[0]; 4457 acquireWakeLock_l(activeTrack->uid()); 4458 if (size > 1) { 4459 SortedVector<int> tmp; 4460 for (size_t i = 0; i < size; i++) { 4461 tmp.add(mActiveTracks[i]->uid()); 4462 } 4463 updateWakeLockUids_l(tmp); 4464 } 4465 } else { 4466 acquireWakeLock_l(-1); 4467 } 4468 } 4469 4470 // start recording 4471 for (;;) { 4472 TrackBase::track_state activeTrackState; 4473 Vector< sp<EffectChain> > effectChains; 4474 4475 // sleep with mutex unlocked 4476 if (doSleep) { 4477 doSleep = false; 4478 usleep(kRecordThreadSleepUs); 4479 } 4480 4481 { // scope for mLock 4482 Mutex::Autolock _l(mLock); 4483 if (exitPending()) { 4484 break; 4485 } 4486 processConfigEvents_l(); 4487 // return value 'reconfig' is currently unused 4488 bool reconfig = checkForNewParameters_l(); 4489 4490 // if no active track(s), then standby and release wakelock 4491 size_t size = mActiveTracks.size(); 4492 if (size == 0) { 4493 standbyIfNotAlreadyInStandby(); 4494 // exitPending() can't become true here 4495 releaseWakeLock_l(); 4496 ALOGV("RecordThread: loop stopping"); 4497 // go to sleep 4498 mWaitWorkCV.wait(mLock); 4499 ALOGV("RecordThread: loop starting"); 4500 goto reacquire_wakelock; 4501 } 4502 4503 if (mActiveTracksGen != activeTracksGen) { 4504 activeTracksGen = mActiveTracksGen; 4505 SortedVector<int> tmp; 4506 for (size_t i = 0; i < size; i++) { 4507 tmp.add(mActiveTracks[i]->uid()); 4508 } 4509 updateWakeLockUids_l(tmp); 4510 // FIXME an arbitrary choice 4511 activeTrack = mActiveTracks[0]; 4512 } 4513 4514 if (activeTrack->isTerminated()) { 4515 removeTrack_l(activeTrack); 4516 mActiveTracks.remove(activeTrack); 4517 mActiveTracksGen++; 4518 continue; 4519 } 4520 4521 activeTrackState = activeTrack->mState; 4522 switch (activeTrackState) { 4523 case TrackBase::PAUSING: 4524 standbyIfNotAlreadyInStandby(); 4525 mActiveTracks.remove(activeTrack); 4526 mActiveTracksGen++; 4527 mStartStopCond.broadcast(); 4528 doSleep = true; 4529 continue; 4530 4531 case TrackBase::RESUMING: 4532 mStandby = false; 4533 if (mReqChannelCount != activeTrack->channelCount()) { 4534 mActiveTracks.remove(activeTrack); 4535 mActiveTracksGen++; 4536 mStartStopCond.broadcast(); 4537 continue; 4538 } 4539 if (readOnce) { 4540 mStartStopCond.broadcast(); 4541 // record start succeeds only if first read from audio input succeeds 4542 if (mBytesRead < 0) { 4543 mActiveTracks.remove(activeTrack); 4544 mActiveTracksGen++; 4545 continue; 4546 } 4547 activeTrack->mState = TrackBase::ACTIVE; 4548 } 4549 break; 4550 4551 case TrackBase::ACTIVE: 4552 break; 4553 4554 case TrackBase::IDLE: 4555 doSleep = true; 4556 continue; 4557 4558 default: 4559 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4560 } 4561 4562 lockEffectChains_l(effectChains); 4563 } 4564 4565 // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable 4566 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4567 4568 for (size_t i = 0; i < effectChains.size(); i ++) { 4569 // thread mutex is not locked, but effect chain is locked 4570 effectChains[i]->process_l(); 4571 } 4572 4573 AudioBufferProvider::Buffer buffer; 4574 buffer.frameCount = mFrameCount; 4575 status_t status = activeTrack->getNextBuffer(&buffer); 4576 if (status == NO_ERROR) { 4577 readOnce = true; 4578 size_t framesOut = buffer.frameCount; 4579 if (mResampler == NULL) { 4580 // no resampling 4581 while (framesOut) { 4582 size_t framesIn = mFrameCount - mRsmpInIndex; 4583 if (framesIn > 0) { 4584 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4585 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4586 activeTrack->mFrameSize; 4587 if (framesIn > framesOut) { 4588 framesIn = framesOut; 4589 } 4590 mRsmpInIndex += framesIn; 4591 framesOut -= framesIn; 4592 if (mChannelCount == mReqChannelCount) { 4593 memcpy(dst, src, framesIn * mFrameSize); 4594 } else { 4595 if (mChannelCount == 1) { 4596 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4597 (int16_t *)src, framesIn); 4598 } else { 4599 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4600 (int16_t *)src, framesIn); 4601 } 4602 } 4603 } 4604 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4605 void *readInto; 4606 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4607 readInto = buffer.raw; 4608 framesOut = 0; 4609 } else { 4610 readInto = mRsmpInBuffer; 4611 mRsmpInIndex = 0; 4612 } 4613 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4614 mBufferSize); 4615 if (mBytesRead <= 0) { 4616 // TODO: verify that it's benign to use a stale track state 4617 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4618 { 4619 ALOGE("Error reading audio input"); 4620 // Force input into standby so that it tries to 4621 // recover at next read attempt 4622 inputStandBy(); 4623 doSleep = true; 4624 } 4625 mRsmpInIndex = mFrameCount; 4626 framesOut = 0; 4627 buffer.frameCount = 0; 4628 } 4629#ifdef TEE_SINK 4630 else if (mTeeSink != 0) { 4631 (void) mTeeSink->write(readInto, 4632 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4633 } 4634#endif 4635 } 4636 } 4637 } else { 4638 // resampling 4639 4640 // avoid busy-waiting if client doesn't keep up 4641 bool madeProgress = false; 4642 4643 // keep mRsmpInBuffer full so resampler always has sufficient input 4644 for (;;) { 4645 int32_t rear = mRsmpInRear; 4646 ssize_t filled = rear - mRsmpInFront; 4647 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4648 // exit once there is enough data in buffer for resampler 4649 if ((size_t) filled >= mRsmpInFrames) { 4650 break; 4651 } 4652 size_t avail = mRsmpInFramesP2 - filled; 4653 // Only try to read full HAL buffers. 4654 // But if the HAL read returns a partial buffer, use it. 4655 if (avail < mFrameCount) { 4656 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4657 avail, mFrameCount); 4658 break; 4659 } 4660 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4661 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4662 rear &= mRsmpInFramesP2 - 1; 4663 mBytesRead = mInput->stream->read(mInput->stream, 4664 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4665 if (mBytesRead <= 0) { 4666 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4667 break; 4668 } 4669 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4670 size_t framesRead = mBytesRead / mFrameSize; 4671 ALOG_ASSERT(framesRead > 0); 4672 madeProgress = true; 4673 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4674 size_t part1 = mRsmpInFramesP2 - rear; 4675 if (framesRead > part1) { 4676 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4677 (framesRead - part1) * mFrameSize); 4678 } 4679 mRsmpInRear += framesRead; 4680 } 4681 4682 if (!madeProgress) { 4683 ALOGV("Did not make progress"); 4684 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4685 } 4686 4687 // resampler accumulates, but we only have one source track 4688 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4689 mResampler->resample(mRsmpOutBuffer, framesOut, 4690 this /* AudioBufferProvider* */); 4691 // ditherAndClamp() works as long as all buffers returned by 4692 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4693 if (mReqChannelCount == 1) { 4694 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4695 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4696 // the resampler always outputs stereo samples: 4697 // do post stereo to mono conversion 4698 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4699 framesOut); 4700 } else { 4701 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4702 } 4703 // now done with mRsmpOutBuffer 4704 4705 } 4706 if (mFramestoDrop == 0) { 4707 activeTrack->releaseBuffer(&buffer); 4708 } else { 4709 if (mFramestoDrop > 0) { 4710 mFramestoDrop -= buffer.frameCount; 4711 if (mFramestoDrop <= 0) { 4712 clearSyncStartEvent(); 4713 } 4714 } else { 4715 mFramestoDrop += buffer.frameCount; 4716 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4717 mSyncStartEvent->isCancelled()) { 4718 ALOGW("Synced record %s, session %d, trigger session %d", 4719 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4720 activeTrack->sessionId(), 4721 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4722 clearSyncStartEvent(); 4723 } 4724 } 4725 } 4726 activeTrack->clearOverflow(); 4727 } 4728 // client isn't retrieving buffers fast enough 4729 else { 4730 if (!activeTrack->setOverflow()) { 4731 nsecs_t now = systemTime(); 4732 if ((now - lastWarning) > kWarningThrottleNs) { 4733 ALOGW("RecordThread: buffer overflow"); 4734 lastWarning = now; 4735 } 4736 } 4737 // Release the processor for a while before asking for a new buffer. 4738 // This will give the application more chance to read from the buffer and 4739 // clear the overflow. 4740 doSleep = true; 4741 } 4742 4743 // enable changes in effect chain 4744 unlockEffectChains(effectChains); 4745 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4746 } 4747 4748 standbyIfNotAlreadyInStandby(); 4749 4750 { 4751 Mutex::Autolock _l(mLock); 4752 for (size_t i = 0; i < mTracks.size(); i++) { 4753 sp<RecordTrack> track = mTracks[i]; 4754 track->invalidate(); 4755 } 4756 mActiveTracks.clear(); 4757 mActiveTracksGen++; 4758 mStartStopCond.broadcast(); 4759 } 4760 4761 releaseWakeLock(); 4762 4763 ALOGV("RecordThread %p exiting", this); 4764 return false; 4765} 4766 4767void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4768{ 4769 if (!mStandby) { 4770 inputStandBy(); 4771 mStandby = true; 4772 } 4773} 4774 4775void AudioFlinger::RecordThread::inputStandBy() 4776{ 4777 mInput->stream->common.standby(&mInput->stream->common); 4778} 4779 4780sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4781 const sp<AudioFlinger::Client>& client, 4782 uint32_t sampleRate, 4783 audio_format_t format, 4784 audio_channel_mask_t channelMask, 4785 size_t *pFrameCount, 4786 int sessionId, 4787 int uid, 4788 IAudioFlinger::track_flags_t *flags, 4789 pid_t tid, 4790 status_t *status) 4791{ 4792 size_t frameCount = *pFrameCount; 4793 sp<RecordTrack> track; 4794 status_t lStatus; 4795 4796 lStatus = initCheck(); 4797 if (lStatus != NO_ERROR) { 4798 ALOGE("createRecordTrack_l() audio driver not initialized"); 4799 goto Exit; 4800 } 4801 // client expresses a preference for FAST, but we get the final say 4802 if (*flags & IAudioFlinger::TRACK_FAST) { 4803 if ( 4804 // use case: callback handler and frame count is default or at least as large as HAL 4805 ( 4806 (tid != -1) && 4807 ((frameCount == 0) || 4808 (frameCount >= mFrameCount)) 4809 ) && 4810 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4811 // mono or stereo 4812 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4813 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4814 // hardware sample rate 4815 (sampleRate == mSampleRate) && 4816 // record thread has an associated fast recorder 4817 hasFastRecorder() 4818 // FIXME test that RecordThread for this fast track has a capable output HAL 4819 // FIXME add a permission test also? 4820 ) { 4821 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4822 if (frameCount == 0) { 4823 frameCount = mFrameCount * kFastTrackMultiplier; 4824 } 4825 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4826 frameCount, mFrameCount); 4827 } else { 4828 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4829 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4830 "hasFastRecorder=%d tid=%d", 4831 frameCount, mFrameCount, format, 4832 audio_is_linear_pcm(format), 4833 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4834 *flags &= ~IAudioFlinger::TRACK_FAST; 4835 // For compatibility with AudioRecord calculation, buffer depth is forced 4836 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4837 // This is probably too conservative, but legacy application code may depend on it. 4838 // If you change this calculation, also review the start threshold which is related. 4839 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4840 size_t mNormalFrameCount = 2048; // FIXME 4841 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4842 if (minBufCount < 2) { 4843 minBufCount = 2; 4844 } 4845 size_t minFrameCount = mNormalFrameCount * minBufCount; 4846 if (frameCount < minFrameCount) { 4847 frameCount = minFrameCount; 4848 } 4849 } 4850 } 4851 *pFrameCount = frameCount; 4852 4853 // FIXME use flags and tid similar to createTrack_l() 4854 4855 { // scope for mLock 4856 Mutex::Autolock _l(mLock); 4857 4858 track = new RecordTrack(this, client, sampleRate, 4859 format, channelMask, frameCount, sessionId, uid); 4860 4861 lStatus = track->initCheck(); 4862 if (lStatus != NO_ERROR) { 4863 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4864 track.clear(); 4865 goto Exit; 4866 } 4867 mTracks.add(track); 4868 4869 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4870 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4871 mAudioFlinger->btNrecIsOff(); 4872 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4873 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4874 4875 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4876 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4877 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4878 // so ask activity manager to do this on our behalf 4879 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4880 } 4881 } 4882 lStatus = NO_ERROR; 4883 4884Exit: 4885 *status = lStatus; 4886 return track; 4887} 4888 4889status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4890 AudioSystem::sync_event_t event, 4891 int triggerSession) 4892{ 4893 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4894 sp<ThreadBase> strongMe = this; 4895 status_t status = NO_ERROR; 4896 4897 if (event == AudioSystem::SYNC_EVENT_NONE) { 4898 clearSyncStartEvent(); 4899 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4900 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4901 triggerSession, 4902 recordTrack->sessionId(), 4903 syncStartEventCallback, 4904 this); 4905 // Sync event can be cancelled by the trigger session if the track is not in a 4906 // compatible state in which case we start record immediately 4907 if (mSyncStartEvent->isCancelled()) { 4908 clearSyncStartEvent(); 4909 } else { 4910 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4911 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4912 } 4913 } 4914 4915 { 4916 // This section is a rendezvous between binder thread executing start() and RecordThread 4917 AutoMutex lock(mLock); 4918 if (mActiveTracks.size() > 0) { 4919 // FIXME does not work for multiple active tracks 4920 if (mActiveTracks.indexOf(recordTrack) != 0) { 4921 status = -EBUSY; 4922 } else if (recordTrack->mState == TrackBase::PAUSING) { 4923 recordTrack->mState = TrackBase::ACTIVE; 4924 } 4925 return status; 4926 } 4927 4928 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4929 recordTrack->mState = TrackBase::IDLE; 4930 mActiveTracks.add(recordTrack); 4931 mActiveTracksGen++; 4932 mLock.unlock(); 4933 status_t status = AudioSystem::startInput(mId); 4934 mLock.lock(); 4935 // FIXME should verify that mActiveTrack is still == recordTrack 4936 if (status != NO_ERROR) { 4937 mActiveTracks.remove(recordTrack); 4938 mActiveTracksGen++; 4939 clearSyncStartEvent(); 4940 return status; 4941 } 4942 // FIXME LEGACY 4943 mRsmpInIndex = mFrameCount; 4944 mRsmpInFront = 0; 4945 mRsmpInRear = 0; 4946 mRsmpInUnrel = 0; 4947 mBytesRead = 0; 4948 if (mResampler != NULL) { 4949 mResampler->reset(); 4950 } 4951 // FIXME hijacking a playback track state name which was intended for start after pause; 4952 // here 'STARTING_2' would be more accurate 4953 recordTrack->mState = TrackBase::RESUMING; 4954 // signal thread to start 4955 ALOGV("Signal record thread"); 4956 mWaitWorkCV.broadcast(); 4957 // do not wait for mStartStopCond if exiting 4958 if (exitPending()) { 4959 mActiveTracks.remove(recordTrack); 4960 mActiveTracksGen++; 4961 status = INVALID_OPERATION; 4962 goto startError; 4963 } 4964 // FIXME incorrect usage of wait: no explicit predicate or loop 4965 mStartStopCond.wait(mLock); 4966 if (mActiveTracks.indexOf(recordTrack) < 0) { 4967 ALOGV("Record failed to start"); 4968 status = BAD_VALUE; 4969 goto startError; 4970 } 4971 ALOGV("Record started OK"); 4972 return status; 4973 } 4974 4975startError: 4976 AudioSystem::stopInput(mId); 4977 clearSyncStartEvent(); 4978 return status; 4979} 4980 4981void AudioFlinger::RecordThread::clearSyncStartEvent() 4982{ 4983 if (mSyncStartEvent != 0) { 4984 mSyncStartEvent->cancel(); 4985 } 4986 mSyncStartEvent.clear(); 4987 mFramestoDrop = 0; 4988} 4989 4990void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4991{ 4992 sp<SyncEvent> strongEvent = event.promote(); 4993 4994 if (strongEvent != 0) { 4995 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4996 me->handleSyncStartEvent(strongEvent); 4997 } 4998} 4999 5000void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5001{ 5002 if (event == mSyncStartEvent) { 5003 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5004 // from audio HAL 5005 mFramestoDrop = mFrameCount * 2; 5006 } 5007} 5008 5009bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5010 ALOGV("RecordThread::stop"); 5011 AutoMutex _l(mLock); 5012 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5013 return false; 5014 } 5015 // note that threadLoop may still be processing the track at this point [without lock] 5016 recordTrack->mState = TrackBase::PAUSING; 5017 // do not wait for mStartStopCond if exiting 5018 if (exitPending()) { 5019 return true; 5020 } 5021 // FIXME incorrect usage of wait: no explicit predicate or loop 5022 mStartStopCond.wait(mLock); 5023 // if we have been restarted, recordTrack is in mActiveTracks here 5024 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5025 ALOGV("Record stopped OK"); 5026 return true; 5027 } 5028 return false; 5029} 5030 5031bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 5032{ 5033 return false; 5034} 5035 5036status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5037{ 5038#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5039 if (!isValidSyncEvent(event)) { 5040 return BAD_VALUE; 5041 } 5042 5043 int eventSession = event->triggerSession(); 5044 status_t ret = NAME_NOT_FOUND; 5045 5046 Mutex::Autolock _l(mLock); 5047 5048 for (size_t i = 0; i < mTracks.size(); i++) { 5049 sp<RecordTrack> track = mTracks[i]; 5050 if (eventSession == track->sessionId()) { 5051 (void) track->setSyncEvent(event); 5052 ret = NO_ERROR; 5053 } 5054 } 5055 return ret; 5056#else 5057 return BAD_VALUE; 5058#endif 5059} 5060 5061// destroyTrack_l() must be called with ThreadBase::mLock held 5062void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5063{ 5064 track->terminate(); 5065 track->mState = TrackBase::STOPPED; 5066 // active tracks are removed by threadLoop() 5067 if (mActiveTracks.indexOf(track) < 0) { 5068 removeTrack_l(track); 5069 } 5070} 5071 5072void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5073{ 5074 mTracks.remove(track); 5075 // need anything related to effects here? 5076} 5077 5078void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5079{ 5080 dumpInternals(fd, args); 5081 dumpTracks(fd, args); 5082 dumpEffectChains(fd, args); 5083} 5084 5085void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5086{ 5087 const size_t SIZE = 256; 5088 char buffer[SIZE]; 5089 String8 result; 5090 5091 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5092 result.append(buffer); 5093 5094 if (mActiveTracks.size() > 0) { 5095 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5096 result.append(buffer); 5097 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5098 result.append(buffer); 5099 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5100 result.append(buffer); 5101 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5102 result.append(buffer); 5103 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5104 result.append(buffer); 5105 } else { 5106 result.append("No active record client\n"); 5107 } 5108 5109 write(fd, result.string(), result.size()); 5110 5111 dumpBase(fd, args); 5112} 5113 5114void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 5115{ 5116 const size_t SIZE = 256; 5117 char buffer[SIZE]; 5118 String8 result; 5119 5120 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5121 result.append(buffer); 5122 RecordTrack::appendDumpHeader(result); 5123 for (size_t i = 0; i < mTracks.size(); ++i) { 5124 sp<RecordTrack> track = mTracks[i]; 5125 if (track != 0) { 5126 track->dump(buffer, SIZE); 5127 result.append(buffer); 5128 } 5129 } 5130 5131 size_t size = mActiveTracks.size(); 5132 if (size > 0) { 5133 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5134 result.append(buffer); 5135 RecordTrack::appendDumpHeader(result); 5136 for (size_t i = 0; i < size; ++i) { 5137 sp<RecordTrack> track = mActiveTracks[i]; 5138 track->dump(buffer, SIZE); 5139 result.append(buffer); 5140 } 5141 5142 } 5143 write(fd, result.string(), result.size()); 5144} 5145 5146// AudioBufferProvider interface 5147status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5148{ 5149 int32_t rear = mRsmpInRear; 5150 int32_t front = mRsmpInFront; 5151 ssize_t filled = rear - front; 5152 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5153 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5154 front &= mRsmpInFramesP2 - 1; 5155 size_t part1 = mRsmpInFramesP2 - front; 5156 if (part1 > (size_t) filled) { 5157 part1 = filled; 5158 } 5159 size_t ask = buffer->frameCount; 5160 ALOG_ASSERT(ask > 0); 5161 if (part1 > ask) { 5162 part1 = ask; 5163 } 5164 if (part1 == 0) { 5165 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5166 ALOGE("RecordThread::getNextBuffer() starved"); 5167 buffer->raw = NULL; 5168 buffer->frameCount = 0; 5169 mRsmpInUnrel = 0; 5170 return NOT_ENOUGH_DATA; 5171 } 5172 5173 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5174 buffer->frameCount = part1; 5175 mRsmpInUnrel = part1; 5176 return NO_ERROR; 5177} 5178 5179// AudioBufferProvider interface 5180void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5181{ 5182 size_t stepCount = buffer->frameCount; 5183 if (stepCount == 0) { 5184 return; 5185 } 5186 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5187 mRsmpInUnrel -= stepCount; 5188 mRsmpInFront += stepCount; 5189 buffer->raw = NULL; 5190 buffer->frameCount = 0; 5191} 5192 5193bool AudioFlinger::RecordThread::checkForNewParameters_l() 5194{ 5195 bool reconfig = false; 5196 5197 while (!mNewParameters.isEmpty()) { 5198 status_t status = NO_ERROR; 5199 String8 keyValuePair = mNewParameters[0]; 5200 AudioParameter param = AudioParameter(keyValuePair); 5201 int value; 5202 audio_format_t reqFormat = mFormat; 5203 uint32_t reqSamplingRate = mReqSampleRate; 5204 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5205 5206 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5207 reqSamplingRate = value; 5208 reconfig = true; 5209 } 5210 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5211 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5212 status = BAD_VALUE; 5213 } else { 5214 reqFormat = (audio_format_t) value; 5215 reconfig = true; 5216 } 5217 } 5218 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5219 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5220 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5221 status = BAD_VALUE; 5222 } else { 5223 reqChannelMask = mask; 5224 reconfig = true; 5225 } 5226 } 5227 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5228 // do not accept frame count changes if tracks are open as the track buffer 5229 // size depends on frame count and correct behavior would not be guaranteed 5230 // if frame count is changed after track creation 5231 if (mActiveTracks.size() > 0) { 5232 status = INVALID_OPERATION; 5233 } else { 5234 reconfig = true; 5235 } 5236 } 5237 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5238 // forward device change to effects that have requested to be 5239 // aware of attached audio device. 5240 for (size_t i = 0; i < mEffectChains.size(); i++) { 5241 mEffectChains[i]->setDevice_l(value); 5242 } 5243 5244 // store input device and output device but do not forward output device to audio HAL. 5245 // Note that status is ignored by the caller for output device 5246 // (see AudioFlinger::setParameters() 5247 if (audio_is_output_devices(value)) { 5248 mOutDevice = value; 5249 status = BAD_VALUE; 5250 } else { 5251 mInDevice = value; 5252 // disable AEC and NS if the device is a BT SCO headset supporting those 5253 // pre processings 5254 if (mTracks.size() > 0) { 5255 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5256 mAudioFlinger->btNrecIsOff(); 5257 for (size_t i = 0; i < mTracks.size(); i++) { 5258 sp<RecordTrack> track = mTracks[i]; 5259 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5260 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5261 } 5262 } 5263 } 5264 } 5265 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5266 mAudioSource != (audio_source_t)value) { 5267 // forward device change to effects that have requested to be 5268 // aware of attached audio device. 5269 for (size_t i = 0; i < mEffectChains.size(); i++) { 5270 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5271 } 5272 mAudioSource = (audio_source_t)value; 5273 } 5274 5275 if (status == NO_ERROR) { 5276 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5277 keyValuePair.string()); 5278 if (status == INVALID_OPERATION) { 5279 inputStandBy(); 5280 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5281 keyValuePair.string()); 5282 } 5283 if (reconfig) { 5284 if (status == BAD_VALUE && 5285 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5286 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5287 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5288 <= (2 * reqSamplingRate)) && 5289 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5290 <= FCC_2 && 5291 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5292 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5293 status = NO_ERROR; 5294 } 5295 if (status == NO_ERROR) { 5296 readInputParameters(); 5297 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5298 } 5299 } 5300 } 5301 5302 mNewParameters.removeAt(0); 5303 5304 mParamStatus = status; 5305 mParamCond.signal(); 5306 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5307 // already timed out waiting for the status and will never signal the condition. 5308 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5309 } 5310 return reconfig; 5311} 5312 5313String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5314{ 5315 Mutex::Autolock _l(mLock); 5316 if (initCheck() != NO_ERROR) { 5317 return String8(); 5318 } 5319 5320 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5321 const String8 out_s8(s); 5322 free(s); 5323 return out_s8; 5324} 5325 5326void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5327 AudioSystem::OutputDescriptor desc; 5328 const void *param2 = NULL; 5329 5330 switch (event) { 5331 case AudioSystem::INPUT_OPENED: 5332 case AudioSystem::INPUT_CONFIG_CHANGED: 5333 desc.channelMask = mChannelMask; 5334 desc.samplingRate = mSampleRate; 5335 desc.format = mFormat; 5336 desc.frameCount = mFrameCount; 5337 desc.latency = 0; 5338 param2 = &desc; 5339 break; 5340 5341 case AudioSystem::INPUT_CLOSED: 5342 default: 5343 break; 5344 } 5345 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5346} 5347 5348void AudioFlinger::RecordThread::readInputParameters() 5349{ 5350 delete[] mRsmpInBuffer; 5351 // mRsmpInBuffer is always assigned a new[] below 5352 delete[] mRsmpOutBuffer; 5353 mRsmpOutBuffer = NULL; 5354 delete mResampler; 5355 mResampler = NULL; 5356 5357 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5358 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5359 mChannelCount = popcount(mChannelMask); 5360 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5361 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5362 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5363 } 5364 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5365 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5366 mFrameCount = mBufferSize / mFrameSize; 5367 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5368 // 1 full output buffer, regardless of the alignment of the available input. 5369 mRsmpInFrames = mFrameCount * 3; 5370 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5371 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5372 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5373 mRsmpInFront = 0; 5374 mRsmpInRear = 0; 5375 mRsmpInUnrel = 0; 5376 5377 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5378 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5379 mResampler->setSampleRate(mSampleRate); 5380 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5381 // resampler always outputs stereo 5382 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5383 } 5384 mRsmpInIndex = mFrameCount; 5385} 5386 5387unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5388{ 5389 Mutex::Autolock _l(mLock); 5390 if (initCheck() != NO_ERROR) { 5391 return 0; 5392 } 5393 5394 return mInput->stream->get_input_frames_lost(mInput->stream); 5395} 5396 5397uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5398{ 5399 Mutex::Autolock _l(mLock); 5400 uint32_t result = 0; 5401 if (getEffectChain_l(sessionId) != 0) { 5402 result = EFFECT_SESSION; 5403 } 5404 5405 for (size_t i = 0; i < mTracks.size(); ++i) { 5406 if (sessionId == mTracks[i]->sessionId()) { 5407 result |= TRACK_SESSION; 5408 break; 5409 } 5410 } 5411 5412 return result; 5413} 5414 5415KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5416{ 5417 KeyedVector<int, bool> ids; 5418 Mutex::Autolock _l(mLock); 5419 for (size_t j = 0; j < mTracks.size(); ++j) { 5420 sp<RecordThread::RecordTrack> track = mTracks[j]; 5421 int sessionId = track->sessionId(); 5422 if (ids.indexOfKey(sessionId) < 0) { 5423 ids.add(sessionId, true); 5424 } 5425 } 5426 return ids; 5427} 5428 5429AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5430{ 5431 Mutex::Autolock _l(mLock); 5432 AudioStreamIn *input = mInput; 5433 mInput = NULL; 5434 return input; 5435} 5436 5437// this method must always be called either with ThreadBase mLock held or inside the thread loop 5438audio_stream_t* AudioFlinger::RecordThread::stream() const 5439{ 5440 if (mInput == NULL) { 5441 return NULL; 5442 } 5443 return &mInput->stream->common; 5444} 5445 5446status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5447{ 5448 // only one chain per input thread 5449 if (mEffectChains.size() != 0) { 5450 return INVALID_OPERATION; 5451 } 5452 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5453 5454 chain->setInBuffer(NULL); 5455 chain->setOutBuffer(NULL); 5456 5457 checkSuspendOnAddEffectChain_l(chain); 5458 5459 mEffectChains.add(chain); 5460 5461 return NO_ERROR; 5462} 5463 5464size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5465{ 5466 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5467 ALOGW_IF(mEffectChains.size() != 1, 5468 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5469 chain.get(), mEffectChains.size(), this); 5470 if (mEffectChains.size() == 1) { 5471 mEffectChains.removeAt(0); 5472 } 5473 return 0; 5474} 5475 5476}; // namespace android 5477