Threads.cpp revision 579dd27d96497022e534e859c6ebbec675ee07aa
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 double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 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 = effect->addHandle(handle.get()); 850 if (enabled != NULL) { 851 *enabled = (int)effect->isEnabled(); 852 } 853 } 854 855Exit: 856 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 857 Mutex::Autolock _l(mLock); 858 if (effectCreated) { 859 chain->removeEffect_l(effect); 860 } 861 if (effectRegistered) { 862 AudioSystem::unregisterEffect(effect->id()); 863 } 864 if (chainCreated) { 865 removeEffectChain_l(chain); 866 } 867 handle.clear(); 868 } 869 870 *status = lStatus; 871 return handle; 872} 873 874sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 875{ 876 Mutex::Autolock _l(mLock); 877 return getEffect_l(sessionId, effectId); 878} 879 880sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 881{ 882 sp<EffectChain> chain = getEffectChain_l(sessionId); 883 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 884} 885 886// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 887// PlaybackThread::mLock held 888status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 889{ 890 // check for existing effect chain with the requested audio session 891 int sessionId = effect->sessionId(); 892 sp<EffectChain> chain = getEffectChain_l(sessionId); 893 bool chainCreated = false; 894 895 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 896 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 897 this, effect->desc().name, effect->desc().flags); 898 899 if (chain == 0) { 900 // create a new chain for this session 901 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 902 chain = new EffectChain(this, sessionId); 903 addEffectChain_l(chain); 904 chain->setStrategy(getStrategyForSession_l(sessionId)); 905 chainCreated = true; 906 } 907 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 908 909 if (chain->getEffectFromId_l(effect->id()) != 0) { 910 ALOGW("addEffect_l() %p effect %s already present in chain %p", 911 this, effect->desc().name, chain.get()); 912 return BAD_VALUE; 913 } 914 915 effect->setOffloaded(mType == OFFLOAD, mId); 916 917 status_t status = chain->addEffect_l(effect); 918 if (status != NO_ERROR) { 919 if (chainCreated) { 920 removeEffectChain_l(chain); 921 } 922 return status; 923 } 924 925 effect->setDevice(mOutDevice); 926 effect->setDevice(mInDevice); 927 effect->setMode(mAudioFlinger->getMode()); 928 effect->setAudioSource(mAudioSource); 929 return NO_ERROR; 930} 931 932void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 933 934 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 935 effect_descriptor_t desc = effect->desc(); 936 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 937 detachAuxEffect_l(effect->id()); 938 } 939 940 sp<EffectChain> chain = effect->chain().promote(); 941 if (chain != 0) { 942 // remove effect chain if removing last effect 943 if (chain->removeEffect_l(effect) == 0) { 944 removeEffectChain_l(chain); 945 } 946 } else { 947 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 948 } 949} 950 951void AudioFlinger::ThreadBase::lockEffectChains_l( 952 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 953{ 954 effectChains = mEffectChains; 955 for (size_t i = 0; i < mEffectChains.size(); i++) { 956 mEffectChains[i]->lock(); 957 } 958} 959 960void AudioFlinger::ThreadBase::unlockEffectChains( 961 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 962{ 963 for (size_t i = 0; i < effectChains.size(); i++) { 964 effectChains[i]->unlock(); 965 } 966} 967 968sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 969{ 970 Mutex::Autolock _l(mLock); 971 return getEffectChain_l(sessionId); 972} 973 974sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 975{ 976 size_t size = mEffectChains.size(); 977 for (size_t i = 0; i < size; i++) { 978 if (mEffectChains[i]->sessionId() == sessionId) { 979 return mEffectChains[i]; 980 } 981 } 982 return 0; 983} 984 985void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 986{ 987 Mutex::Autolock _l(mLock); 988 size_t size = mEffectChains.size(); 989 for (size_t i = 0; i < size; i++) { 990 mEffectChains[i]->setMode_l(mode); 991 } 992} 993 994void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 995 EffectHandle *handle, 996 bool unpinIfLast) { 997 998 Mutex::Autolock _l(mLock); 999 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1000 // delete the effect module if removing last handle on it 1001 if (effect->removeHandle(handle) == 0) { 1002 if (!effect->isPinned() || unpinIfLast) { 1003 removeEffect_l(effect); 1004 AudioSystem::unregisterEffect(effect->id()); 1005 } 1006 } 1007} 1008 1009// ---------------------------------------------------------------------------- 1010// Playback 1011// ---------------------------------------------------------------------------- 1012 1013AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1014 AudioStreamOut* output, 1015 audio_io_handle_t id, 1016 audio_devices_t device, 1017 type_t type) 1018 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1019 mNormalFrameCount(0), mMixBuffer(NULL), 1020 mSuspended(0), mBytesWritten(0), 1021 mActiveTracksGeneration(0), 1022 // mStreamTypes[] initialized in constructor body 1023 mOutput(output), 1024 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1025 mMixerStatus(MIXER_IDLE), 1026 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1027 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1028 mBytesRemaining(0), 1029 mCurrentWriteLength(0), 1030 mUseAsyncWrite(false), 1031 mWriteAckSequence(0), 1032 mDrainSequence(0), 1033 mSignalPending(false), 1034 mScreenState(AudioFlinger::mScreenState), 1035 // index 0 is reserved for normal mixer's submix 1036 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1037 // mLatchD, mLatchQ, 1038 mLatchDValid(false), mLatchQValid(false) 1039{ 1040 snprintf(mName, kNameLength, "AudioOut_%X", id); 1041 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1042 1043 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1044 // it would be safer to explicitly pass initial masterVolume/masterMute as 1045 // parameter. 1046 // 1047 // If the HAL we are using has support for master volume or master mute, 1048 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1049 // and the mute set to false). 1050 mMasterVolume = audioFlinger->masterVolume_l(); 1051 mMasterMute = audioFlinger->masterMute_l(); 1052 if (mOutput && mOutput->audioHwDev) { 1053 if (mOutput->audioHwDev->canSetMasterVolume()) { 1054 mMasterVolume = 1.0; 1055 } 1056 1057 if (mOutput->audioHwDev->canSetMasterMute()) { 1058 mMasterMute = false; 1059 } 1060 } 1061 1062 readOutputParameters(); 1063 1064 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1065 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1066 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1067 stream = (audio_stream_type_t) (stream + 1)) { 1068 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1069 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1070 } 1071 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1072 // because mAudioFlinger doesn't have one to copy from 1073} 1074 1075AudioFlinger::PlaybackThread::~PlaybackThread() 1076{ 1077 mAudioFlinger->unregisterWriter(mNBLogWriter); 1078 delete[] mMixBuffer; 1079} 1080 1081void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1082{ 1083 dumpInternals(fd, args); 1084 dumpTracks(fd, args); 1085 dumpEffectChains(fd, args); 1086} 1087 1088void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1089{ 1090 const size_t SIZE = 256; 1091 char buffer[SIZE]; 1092 String8 result; 1093 1094 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1095 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1096 const stream_type_t *st = &mStreamTypes[i]; 1097 if (i > 0) { 1098 result.appendFormat(", "); 1099 } 1100 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1101 if (st->mute) { 1102 result.append("M"); 1103 } 1104 } 1105 result.append("\n"); 1106 write(fd, result.string(), result.length()); 1107 result.clear(); 1108 1109 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1110 result.append(buffer); 1111 Track::appendDumpHeader(result); 1112 for (size_t i = 0; i < mTracks.size(); ++i) { 1113 sp<Track> track = mTracks[i]; 1114 if (track != 0) { 1115 track->dump(buffer, SIZE); 1116 result.append(buffer); 1117 } 1118 } 1119 1120 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1121 result.append(buffer); 1122 Track::appendDumpHeader(result); 1123 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1124 sp<Track> track = mActiveTracks[i].promote(); 1125 if (track != 0) { 1126 track->dump(buffer, SIZE); 1127 result.append(buffer); 1128 } 1129 } 1130 write(fd, result.string(), result.size()); 1131 1132 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1133 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1134 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1135 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1136} 1137 1138void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1139{ 1140 const size_t SIZE = 256; 1141 char buffer[SIZE]; 1142 String8 result; 1143 1144 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1145 result.append(buffer); 1146 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1147 result.append(buffer); 1148 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1149 ns2ms(systemTime() - mLastWriteTime)); 1150 result.append(buffer); 1151 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1152 result.append(buffer); 1153 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1154 result.append(buffer); 1155 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1156 result.append(buffer); 1157 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1158 result.append(buffer); 1159 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1160 result.append(buffer); 1161 write(fd, result.string(), result.size()); 1162 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1163 1164 dumpBase(fd, args); 1165} 1166 1167// Thread virtuals 1168 1169void AudioFlinger::PlaybackThread::onFirstRef() 1170{ 1171 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1172} 1173 1174// ThreadBase virtuals 1175void AudioFlinger::PlaybackThread::preExit() 1176{ 1177 ALOGV(" preExit()"); 1178 // FIXME this is using hard-coded strings but in the future, this functionality will be 1179 // converted to use audio HAL extensions required to support tunneling 1180 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1181} 1182 1183// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1184sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1185 const sp<AudioFlinger::Client>& client, 1186 audio_stream_type_t streamType, 1187 uint32_t sampleRate, 1188 audio_format_t format, 1189 audio_channel_mask_t channelMask, 1190 size_t frameCount, 1191 const sp<IMemory>& sharedBuffer, 1192 int sessionId, 1193 IAudioFlinger::track_flags_t *flags, 1194 pid_t tid, 1195 int uid, 1196 status_t *status) 1197{ 1198 sp<Track> track; 1199 status_t lStatus; 1200 1201 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1202 1203 // client expresses a preference for FAST, but we get the final say 1204 if (*flags & IAudioFlinger::TRACK_FAST) { 1205 if ( 1206 // not timed 1207 (!isTimed) && 1208 // either of these use cases: 1209 ( 1210 // use case 1: shared buffer with any frame count 1211 ( 1212 (sharedBuffer != 0) 1213 ) || 1214 // use case 2: callback handler and frame count is default or at least as large as HAL 1215 ( 1216 (tid != -1) && 1217 ((frameCount == 0) || 1218 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1219 ) 1220 ) && 1221 // PCM data 1222 audio_is_linear_pcm(format) && 1223 // mono or stereo 1224 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1225 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1226#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1227 // hardware sample rate 1228 (sampleRate == mSampleRate) && 1229#endif 1230 // normal mixer has an associated fast mixer 1231 hasFastMixer() && 1232 // there are sufficient fast track slots available 1233 (mFastTrackAvailMask != 0) 1234 // FIXME test that MixerThread for this fast track has a capable output HAL 1235 // FIXME add a permission test also? 1236 ) { 1237 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1238 if (frameCount == 0) { 1239 frameCount = mFrameCount * kFastTrackMultiplier; 1240 } 1241 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1242 frameCount, mFrameCount); 1243 } else { 1244 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1245 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1246 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1247 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1248 audio_is_linear_pcm(format), 1249 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1250 *flags &= ~IAudioFlinger::TRACK_FAST; 1251 // For compatibility with AudioTrack calculation, buffer depth is forced 1252 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1253 // This is probably too conservative, but legacy application code may depend on it. 1254 // If you change this calculation, also review the start threshold which is related. 1255 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1256 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1257 if (minBufCount < 2) { 1258 minBufCount = 2; 1259 } 1260 size_t minFrameCount = mNormalFrameCount * minBufCount; 1261 if (frameCount < minFrameCount) { 1262 frameCount = minFrameCount; 1263 } 1264 } 1265 } 1266 1267 if (mType == DIRECT) { 1268 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1269 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1270 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1271 "for output %p with format %d", 1272 sampleRate, format, channelMask, mOutput, mFormat); 1273 lStatus = BAD_VALUE; 1274 goto Exit; 1275 } 1276 } 1277 } else if (mType == OFFLOAD) { 1278 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1279 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1280 "for output %p with format %d", 1281 sampleRate, format, channelMask, mOutput, mFormat); 1282 lStatus = BAD_VALUE; 1283 goto Exit; 1284 } 1285 } else { 1286 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1287 ALOGE("createTrack_l() Bad parameter: format %d \"" 1288 "for output %p with format %d", 1289 format, mOutput, mFormat); 1290 lStatus = BAD_VALUE; 1291 goto Exit; 1292 } 1293 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1294 if (sampleRate > mSampleRate*2) { 1295 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1296 lStatus = BAD_VALUE; 1297 goto Exit; 1298 } 1299 } 1300 1301 lStatus = initCheck(); 1302 if (lStatus != NO_ERROR) { 1303 ALOGE("Audio driver not initialized."); 1304 goto Exit; 1305 } 1306 1307 { // scope for mLock 1308 Mutex::Autolock _l(mLock); 1309 1310 // all tracks in same audio session must share the same routing strategy otherwise 1311 // conflicts will happen when tracks are moved from one output to another by audio policy 1312 // manager 1313 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1314 for (size_t i = 0; i < mTracks.size(); ++i) { 1315 sp<Track> t = mTracks[i]; 1316 if (t != 0 && !t->isOutputTrack()) { 1317 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1318 if (sessionId == t->sessionId() && strategy != actual) { 1319 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1320 strategy, actual); 1321 lStatus = BAD_VALUE; 1322 goto Exit; 1323 } 1324 } 1325 } 1326 1327 if (!isTimed) { 1328 track = new Track(this, client, streamType, sampleRate, format, 1329 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1330 } else { 1331 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1332 channelMask, frameCount, sharedBuffer, sessionId, uid); 1333 } 1334 1335 // new Track always returns non-NULL, 1336 // but TimedTrack::create() is a factory that could fail by returning NULL 1337 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1338 if (lStatus != NO_ERROR) { 1339 track.clear(); 1340 goto Exit; 1341 } 1342 1343 mTracks.add(track); 1344 1345 sp<EffectChain> chain = getEffectChain_l(sessionId); 1346 if (chain != 0) { 1347 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1348 track->setMainBuffer(chain->inBuffer()); 1349 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1350 chain->incTrackCnt(); 1351 } 1352 1353 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1354 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1355 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1356 // so ask activity manager to do this on our behalf 1357 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1358 } 1359 } 1360 1361 lStatus = NO_ERROR; 1362 1363Exit: 1364 *status = lStatus; 1365 return track; 1366} 1367 1368uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1369{ 1370 return latency; 1371} 1372 1373uint32_t AudioFlinger::PlaybackThread::latency() const 1374{ 1375 Mutex::Autolock _l(mLock); 1376 return latency_l(); 1377} 1378uint32_t AudioFlinger::PlaybackThread::latency_l() const 1379{ 1380 if (initCheck() == NO_ERROR) { 1381 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1382 } else { 1383 return 0; 1384 } 1385} 1386 1387void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1388{ 1389 Mutex::Autolock _l(mLock); 1390 // Don't apply master volume in SW if our HAL can do it for us. 1391 if (mOutput && mOutput->audioHwDev && 1392 mOutput->audioHwDev->canSetMasterVolume()) { 1393 mMasterVolume = 1.0; 1394 } else { 1395 mMasterVolume = value; 1396 } 1397} 1398 1399void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1400{ 1401 Mutex::Autolock _l(mLock); 1402 // Don't apply master mute in SW if our HAL can do it for us. 1403 if (mOutput && mOutput->audioHwDev && 1404 mOutput->audioHwDev->canSetMasterMute()) { 1405 mMasterMute = false; 1406 } else { 1407 mMasterMute = muted; 1408 } 1409} 1410 1411void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1412{ 1413 Mutex::Autolock _l(mLock); 1414 mStreamTypes[stream].volume = value; 1415 broadcast_l(); 1416} 1417 1418void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1419{ 1420 Mutex::Autolock _l(mLock); 1421 mStreamTypes[stream].mute = muted; 1422 broadcast_l(); 1423} 1424 1425float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1426{ 1427 Mutex::Autolock _l(mLock); 1428 return mStreamTypes[stream].volume; 1429} 1430 1431// addTrack_l() must be called with ThreadBase::mLock held 1432status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1433{ 1434 status_t status = ALREADY_EXISTS; 1435 1436 // set retry count for buffer fill 1437 track->mRetryCount = kMaxTrackStartupRetries; 1438 if (mActiveTracks.indexOf(track) < 0) { 1439 // the track is newly added, make sure it fills up all its 1440 // buffers before playing. This is to ensure the client will 1441 // effectively get the latency it requested. 1442 if (!track->isOutputTrack()) { 1443 TrackBase::track_state state = track->mState; 1444 mLock.unlock(); 1445 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1446 mLock.lock(); 1447 // abort track was stopped/paused while we released the lock 1448 if (state != track->mState) { 1449 if (status == NO_ERROR) { 1450 mLock.unlock(); 1451 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1452 mLock.lock(); 1453 } 1454 return INVALID_OPERATION; 1455 } 1456 // abort if start is rejected by audio policy manager 1457 if (status != NO_ERROR) { 1458 return PERMISSION_DENIED; 1459 } 1460#ifdef ADD_BATTERY_DATA 1461 // to track the speaker usage 1462 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1463#endif 1464 } 1465 1466 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1467 track->mResetDone = false; 1468 track->mPresentationCompleteFrames = 0; 1469 mActiveTracks.add(track); 1470 mWakeLockUids.add(track->uid()); 1471 mActiveTracksGeneration++; 1472 mLatestActiveTrack = track; 1473 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1474 if (chain != 0) { 1475 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1476 track->sessionId()); 1477 chain->incActiveTrackCnt(); 1478 } 1479 1480 status = NO_ERROR; 1481 } 1482 1483 ALOGV("signal playback thread"); 1484 broadcast_l(); 1485 1486 return status; 1487} 1488 1489bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1490{ 1491 track->terminate(); 1492 // active tracks are removed by threadLoop() 1493 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1494 track->mState = TrackBase::STOPPED; 1495 if (!trackActive) { 1496 removeTrack_l(track); 1497 } else if (track->isFastTrack() || track->isOffloaded()) { 1498 track->mState = TrackBase::STOPPING_1; 1499 } 1500 1501 return trackActive; 1502} 1503 1504void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1505{ 1506 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1507 mTracks.remove(track); 1508 deleteTrackName_l(track->name()); 1509 // redundant as track is about to be destroyed, for dumpsys only 1510 track->mName = -1; 1511 if (track->isFastTrack()) { 1512 int index = track->mFastIndex; 1513 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1514 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1515 mFastTrackAvailMask |= 1 << index; 1516 // redundant as track is about to be destroyed, for dumpsys only 1517 track->mFastIndex = -1; 1518 } 1519 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1520 if (chain != 0) { 1521 chain->decTrackCnt(); 1522 } 1523} 1524 1525void AudioFlinger::PlaybackThread::broadcast_l() 1526{ 1527 // Thread could be blocked waiting for async 1528 // so signal it to handle state changes immediately 1529 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1530 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1531 mSignalPending = true; 1532 mWaitWorkCV.broadcast(); 1533} 1534 1535String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1536{ 1537 Mutex::Autolock _l(mLock); 1538 if (initCheck() != NO_ERROR) { 1539 return String8(); 1540 } 1541 1542 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1543 const String8 out_s8(s); 1544 free(s); 1545 return out_s8; 1546} 1547 1548// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1549void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1550 AudioSystem::OutputDescriptor desc; 1551 void *param2 = NULL; 1552 1553 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1554 param); 1555 1556 switch (event) { 1557 case AudioSystem::OUTPUT_OPENED: 1558 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1559 desc.channelMask = mChannelMask; 1560 desc.samplingRate = mSampleRate; 1561 desc.format = mFormat; 1562 desc.frameCount = mNormalFrameCount; // FIXME see 1563 // AudioFlinger::frameCount(audio_io_handle_t) 1564 desc.latency = latency(); 1565 param2 = &desc; 1566 break; 1567 1568 case AudioSystem::STREAM_CONFIG_CHANGED: 1569 param2 = ¶m; 1570 case AudioSystem::OUTPUT_CLOSED: 1571 default: 1572 break; 1573 } 1574 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1575} 1576 1577void AudioFlinger::PlaybackThread::writeCallback() 1578{ 1579 ALOG_ASSERT(mCallbackThread != 0); 1580 mCallbackThread->resetWriteBlocked(); 1581} 1582 1583void AudioFlinger::PlaybackThread::drainCallback() 1584{ 1585 ALOG_ASSERT(mCallbackThread != 0); 1586 mCallbackThread->resetDraining(); 1587} 1588 1589void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1590{ 1591 Mutex::Autolock _l(mLock); 1592 // reject out of sequence requests 1593 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1594 mWriteAckSequence &= ~1; 1595 mWaitWorkCV.signal(); 1596 } 1597} 1598 1599void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1600{ 1601 Mutex::Autolock _l(mLock); 1602 // reject out of sequence requests 1603 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1604 mDrainSequence &= ~1; 1605 mWaitWorkCV.signal(); 1606 } 1607} 1608 1609// static 1610int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1611 void *param, 1612 void *cookie) 1613{ 1614 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1615 ALOGV("asyncCallback() event %d", event); 1616 switch (event) { 1617 case STREAM_CBK_EVENT_WRITE_READY: 1618 me->writeCallback(); 1619 break; 1620 case STREAM_CBK_EVENT_DRAIN_READY: 1621 me->drainCallback(); 1622 break; 1623 default: 1624 ALOGW("asyncCallback() unknown event %d", event); 1625 break; 1626 } 1627 return 0; 1628} 1629 1630void AudioFlinger::PlaybackThread::readOutputParameters() 1631{ 1632 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1633 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1634 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1635 if (!audio_is_output_channel(mChannelMask)) { 1636 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1637 } 1638 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1639 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1640 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1641 } 1642 mChannelCount = popcount(mChannelMask); 1643 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1644 if (!audio_is_valid_format(mFormat)) { 1645 LOG_FATAL("HAL format %d not valid for output", mFormat); 1646 } 1647 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1648 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1649 mFormat); 1650 } 1651 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1652 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1653 mFrameCount = mBufferSize / mFrameSize; 1654 if (mFrameCount & 15) { 1655 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1656 mFrameCount); 1657 } 1658 1659 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1660 (mOutput->stream->set_callback != NULL)) { 1661 if (mOutput->stream->set_callback(mOutput->stream, 1662 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1663 mUseAsyncWrite = true; 1664 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1665 } 1666 } 1667 1668 // Calculate size of normal mix buffer relative to the HAL output buffer size 1669 double multiplier = 1.0; 1670 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1671 kUseFastMixer == FastMixer_Dynamic)) { 1672 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1673 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1674 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1675 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1676 maxNormalFrameCount = maxNormalFrameCount & ~15; 1677 if (maxNormalFrameCount < minNormalFrameCount) { 1678 maxNormalFrameCount = minNormalFrameCount; 1679 } 1680 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1681 if (multiplier <= 1.0) { 1682 multiplier = 1.0; 1683 } else if (multiplier <= 2.0) { 1684 if (2 * mFrameCount <= maxNormalFrameCount) { 1685 multiplier = 2.0; 1686 } else { 1687 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1688 } 1689 } else { 1690 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1691 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1692 // track, but we sometimes have to do this to satisfy the maximum frame count 1693 // constraint) 1694 // FIXME this rounding up should not be done if no HAL SRC 1695 uint32_t truncMult = (uint32_t) multiplier; 1696 if ((truncMult & 1)) { 1697 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1698 ++truncMult; 1699 } 1700 } 1701 multiplier = (double) truncMult; 1702 } 1703 } 1704 mNormalFrameCount = multiplier * mFrameCount; 1705 // round up to nearest 16 frames to satisfy AudioMixer 1706 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1707 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1708 mNormalFrameCount); 1709 1710 delete[] mMixBuffer; 1711 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1712 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1713 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1714 memset(mMixBuffer, 0, normalBufferSize); 1715 1716 // force reconfiguration of effect chains and engines to take new buffer size and audio 1717 // parameters into account 1718 // Note that mLock is not held when readOutputParameters() is called from the constructor 1719 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1720 // matter. 1721 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1722 Vector< sp<EffectChain> > effectChains = mEffectChains; 1723 for (size_t i = 0; i < effectChains.size(); i ++) { 1724 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1725 } 1726} 1727 1728 1729status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1730{ 1731 if (halFrames == NULL || dspFrames == NULL) { 1732 return BAD_VALUE; 1733 } 1734 Mutex::Autolock _l(mLock); 1735 if (initCheck() != NO_ERROR) { 1736 return INVALID_OPERATION; 1737 } 1738 size_t framesWritten = mBytesWritten / mFrameSize; 1739 *halFrames = framesWritten; 1740 1741 if (isSuspended()) { 1742 // return an estimation of rendered frames when the output is suspended 1743 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1744 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1745 return NO_ERROR; 1746 } else { 1747 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1748 } 1749} 1750 1751uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1752{ 1753 Mutex::Autolock _l(mLock); 1754 uint32_t result = 0; 1755 if (getEffectChain_l(sessionId) != 0) { 1756 result = EFFECT_SESSION; 1757 } 1758 1759 for (size_t i = 0; i < mTracks.size(); ++i) { 1760 sp<Track> track = mTracks[i]; 1761 if (sessionId == track->sessionId() && !track->isInvalid()) { 1762 result |= TRACK_SESSION; 1763 break; 1764 } 1765 } 1766 1767 return result; 1768} 1769 1770uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1771{ 1772 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1773 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1774 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1775 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1776 } 1777 for (size_t i = 0; i < mTracks.size(); i++) { 1778 sp<Track> track = mTracks[i]; 1779 if (sessionId == track->sessionId() && !track->isInvalid()) { 1780 return AudioSystem::getStrategyForStream(track->streamType()); 1781 } 1782 } 1783 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1784} 1785 1786 1787AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1788{ 1789 Mutex::Autolock _l(mLock); 1790 return mOutput; 1791} 1792 1793AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1794{ 1795 Mutex::Autolock _l(mLock); 1796 AudioStreamOut *output = mOutput; 1797 mOutput = NULL; 1798 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1799 // must push a NULL and wait for ack 1800 mOutputSink.clear(); 1801 mPipeSink.clear(); 1802 mNormalSink.clear(); 1803 return output; 1804} 1805 1806// this method must always be called either with ThreadBase mLock held or inside the thread loop 1807audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1808{ 1809 if (mOutput == NULL) { 1810 return NULL; 1811 } 1812 return &mOutput->stream->common; 1813} 1814 1815uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1816{ 1817 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1818} 1819 1820status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1821{ 1822 if (!isValidSyncEvent(event)) { 1823 return BAD_VALUE; 1824 } 1825 1826 Mutex::Autolock _l(mLock); 1827 1828 for (size_t i = 0; i < mTracks.size(); ++i) { 1829 sp<Track> track = mTracks[i]; 1830 if (event->triggerSession() == track->sessionId()) { 1831 (void) track->setSyncEvent(event); 1832 return NO_ERROR; 1833 } 1834 } 1835 1836 return NAME_NOT_FOUND; 1837} 1838 1839bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1840{ 1841 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1842} 1843 1844void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1845 const Vector< sp<Track> >& tracksToRemove) 1846{ 1847 size_t count = tracksToRemove.size(); 1848 if (count > 0) { 1849 for (size_t i = 0 ; i < count ; i++) { 1850 const sp<Track>& track = tracksToRemove.itemAt(i); 1851 if (!track->isOutputTrack()) { 1852 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1853#ifdef ADD_BATTERY_DATA 1854 // to track the speaker usage 1855 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1856#endif 1857 if (track->isTerminated()) { 1858 AudioSystem::releaseOutput(mId); 1859 } 1860 } 1861 } 1862 } 1863} 1864 1865void AudioFlinger::PlaybackThread::checkSilentMode_l() 1866{ 1867 if (!mMasterMute) { 1868 char value[PROPERTY_VALUE_MAX]; 1869 if (property_get("ro.audio.silent", value, "0") > 0) { 1870 char *endptr; 1871 unsigned long ul = strtoul(value, &endptr, 0); 1872 if (*endptr == '\0' && ul != 0) { 1873 ALOGD("Silence is golden"); 1874 // The setprop command will not allow a property to be changed after 1875 // the first time it is set, so we don't have to worry about un-muting. 1876 setMasterMute_l(true); 1877 } 1878 } 1879 } 1880} 1881 1882// shared by MIXER and DIRECT, overridden by DUPLICATING 1883ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1884{ 1885 // FIXME rewrite to reduce number of system calls 1886 mLastWriteTime = systemTime(); 1887 mInWrite = true; 1888 ssize_t bytesWritten; 1889 1890 // If an NBAIO sink is present, use it to write the normal mixer's submix 1891 if (mNormalSink != 0) { 1892#define mBitShift 2 // FIXME 1893 size_t count = mBytesRemaining >> mBitShift; 1894 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1895 ATRACE_BEGIN("write"); 1896 // update the setpoint when AudioFlinger::mScreenState changes 1897 uint32_t screenState = AudioFlinger::mScreenState; 1898 if (screenState != mScreenState) { 1899 mScreenState = screenState; 1900 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1901 if (pipe != NULL) { 1902 pipe->setAvgFrames((mScreenState & 1) ? 1903 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1904 } 1905 } 1906 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1907 ATRACE_END(); 1908 if (framesWritten > 0) { 1909 bytesWritten = framesWritten << mBitShift; 1910 } else { 1911 bytesWritten = framesWritten; 1912 } 1913 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1914 if (status == NO_ERROR) { 1915 size_t totalFramesWritten = mNormalSink->framesWritten(); 1916 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1917 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1918 mLatchDValid = true; 1919 } 1920 } 1921 // otherwise use the HAL / AudioStreamOut directly 1922 } else { 1923 // Direct output and offload threads 1924 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1925 if (mUseAsyncWrite) { 1926 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1927 mWriteAckSequence += 2; 1928 mWriteAckSequence |= 1; 1929 ALOG_ASSERT(mCallbackThread != 0); 1930 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1931 } 1932 // FIXME We should have an implementation of timestamps for direct output threads. 1933 // They are used e.g for multichannel PCM playback over HDMI. 1934 bytesWritten = mOutput->stream->write(mOutput->stream, 1935 mMixBuffer + offset, mBytesRemaining); 1936 if (mUseAsyncWrite && 1937 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1938 // do not wait for async callback in case of error of full write 1939 mWriteAckSequence &= ~1; 1940 ALOG_ASSERT(mCallbackThread != 0); 1941 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1942 } 1943 } 1944 1945 mNumWrites++; 1946 mInWrite = false; 1947 mStandby = false; 1948 return bytesWritten; 1949} 1950 1951void AudioFlinger::PlaybackThread::threadLoop_drain() 1952{ 1953 if (mOutput->stream->drain) { 1954 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1955 if (mUseAsyncWrite) { 1956 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1957 mDrainSequence |= 1; 1958 ALOG_ASSERT(mCallbackThread != 0); 1959 mCallbackThread->setDraining(mDrainSequence); 1960 } 1961 mOutput->stream->drain(mOutput->stream, 1962 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1963 : AUDIO_DRAIN_ALL); 1964 } 1965} 1966 1967void AudioFlinger::PlaybackThread::threadLoop_exit() 1968{ 1969 // Default implementation has nothing to do 1970} 1971 1972/* 1973The derived values that are cached: 1974 - mixBufferSize from frame count * frame size 1975 - activeSleepTime from activeSleepTimeUs() 1976 - idleSleepTime from idleSleepTimeUs() 1977 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1978 - maxPeriod from frame count and sample rate (MIXER only) 1979 1980The parameters that affect these derived values are: 1981 - frame count 1982 - frame size 1983 - sample rate 1984 - device type: A2DP or not 1985 - device latency 1986 - format: PCM or not 1987 - active sleep time 1988 - idle sleep time 1989*/ 1990 1991void AudioFlinger::PlaybackThread::cacheParameters_l() 1992{ 1993 mixBufferSize = mNormalFrameCount * mFrameSize; 1994 activeSleepTime = activeSleepTimeUs(); 1995 idleSleepTime = idleSleepTimeUs(); 1996} 1997 1998void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1999{ 2000 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2001 this, streamType, mTracks.size()); 2002 Mutex::Autolock _l(mLock); 2003 2004 size_t size = mTracks.size(); 2005 for (size_t i = 0; i < size; i++) { 2006 sp<Track> t = mTracks[i]; 2007 if (t->streamType() == streamType) { 2008 t->invalidate(); 2009 } 2010 } 2011} 2012 2013status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2014{ 2015 int session = chain->sessionId(); 2016 int16_t *buffer = mMixBuffer; 2017 bool ownsBuffer = false; 2018 2019 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2020 if (session > 0) { 2021 // Only one effect chain can be present in direct output thread and it uses 2022 // the mix buffer as input 2023 if (mType != DIRECT) { 2024 size_t numSamples = mNormalFrameCount * mChannelCount; 2025 buffer = new int16_t[numSamples]; 2026 memset(buffer, 0, numSamples * sizeof(int16_t)); 2027 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2028 ownsBuffer = true; 2029 } 2030 2031 // Attach all tracks with same session ID to this chain. 2032 for (size_t i = 0; i < mTracks.size(); ++i) { 2033 sp<Track> track = mTracks[i]; 2034 if (session == track->sessionId()) { 2035 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2036 buffer); 2037 track->setMainBuffer(buffer); 2038 chain->incTrackCnt(); 2039 } 2040 } 2041 2042 // indicate all active tracks in the chain 2043 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2044 sp<Track> track = mActiveTracks[i].promote(); 2045 if (track == 0) { 2046 continue; 2047 } 2048 if (session == track->sessionId()) { 2049 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2050 chain->incActiveTrackCnt(); 2051 } 2052 } 2053 } 2054 2055 chain->setInBuffer(buffer, ownsBuffer); 2056 chain->setOutBuffer(mMixBuffer); 2057 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2058 // chains list in order to be processed last as it contains output stage effects 2059 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2060 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2061 // after track specific effects and before output stage 2062 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2063 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2064 // Effect chain for other sessions are inserted at beginning of effect 2065 // chains list to be processed before output mix effects. Relative order between other 2066 // sessions is not important 2067 size_t size = mEffectChains.size(); 2068 size_t i = 0; 2069 for (i = 0; i < size; i++) { 2070 if (mEffectChains[i]->sessionId() < session) { 2071 break; 2072 } 2073 } 2074 mEffectChains.insertAt(chain, i); 2075 checkSuspendOnAddEffectChain_l(chain); 2076 2077 return NO_ERROR; 2078} 2079 2080size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2081{ 2082 int session = chain->sessionId(); 2083 2084 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2085 2086 for (size_t i = 0; i < mEffectChains.size(); i++) { 2087 if (chain == mEffectChains[i]) { 2088 mEffectChains.removeAt(i); 2089 // detach all active tracks from the chain 2090 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2091 sp<Track> track = mActiveTracks[i].promote(); 2092 if (track == 0) { 2093 continue; 2094 } 2095 if (session == track->sessionId()) { 2096 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2097 chain.get(), session); 2098 chain->decActiveTrackCnt(); 2099 } 2100 } 2101 2102 // detach all tracks with same session ID from this chain 2103 for (size_t i = 0; i < mTracks.size(); ++i) { 2104 sp<Track> track = mTracks[i]; 2105 if (session == track->sessionId()) { 2106 track->setMainBuffer(mMixBuffer); 2107 chain->decTrackCnt(); 2108 } 2109 } 2110 break; 2111 } 2112 } 2113 return mEffectChains.size(); 2114} 2115 2116status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2117 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2118{ 2119 Mutex::Autolock _l(mLock); 2120 return attachAuxEffect_l(track, EffectId); 2121} 2122 2123status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2124 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2125{ 2126 status_t status = NO_ERROR; 2127 2128 if (EffectId == 0) { 2129 track->setAuxBuffer(0, NULL); 2130 } else { 2131 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2132 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2133 if (effect != 0) { 2134 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2135 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2136 } else { 2137 status = INVALID_OPERATION; 2138 } 2139 } else { 2140 status = BAD_VALUE; 2141 } 2142 } 2143 return status; 2144} 2145 2146void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2147{ 2148 for (size_t i = 0; i < mTracks.size(); ++i) { 2149 sp<Track> track = mTracks[i]; 2150 if (track->auxEffectId() == effectId) { 2151 attachAuxEffect_l(track, 0); 2152 } 2153 } 2154} 2155 2156bool AudioFlinger::PlaybackThread::threadLoop() 2157{ 2158 Vector< sp<Track> > tracksToRemove; 2159 2160 standbyTime = systemTime(); 2161 2162 // MIXER 2163 nsecs_t lastWarning = 0; 2164 2165 // DUPLICATING 2166 // FIXME could this be made local to while loop? 2167 writeFrames = 0; 2168 2169 int lastGeneration = 0; 2170 2171 cacheParameters_l(); 2172 sleepTime = idleSleepTime; 2173 2174 if (mType == MIXER) { 2175 sleepTimeShift = 0; 2176 } 2177 2178 CpuStats cpuStats; 2179 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2180 2181 acquireWakeLock(); 2182 2183 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2184 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2185 // and then that string will be logged at the next convenient opportunity. 2186 const char *logString = NULL; 2187 2188 checkSilentMode_l(); 2189 2190 while (!exitPending()) 2191 { 2192 cpuStats.sample(myName); 2193 2194 Vector< sp<EffectChain> > effectChains; 2195 2196 processConfigEvents(); 2197 2198 { // scope for mLock 2199 2200 Mutex::Autolock _l(mLock); 2201 2202 if (logString != NULL) { 2203 mNBLogWriter->logTimestamp(); 2204 mNBLogWriter->log(logString); 2205 logString = NULL; 2206 } 2207 2208 if (mLatchDValid) { 2209 mLatchQ = mLatchD; 2210 mLatchDValid = false; 2211 mLatchQValid = true; 2212 } 2213 2214 if (checkForNewParameters_l()) { 2215 cacheParameters_l(); 2216 } 2217 2218 saveOutputTracks(); 2219 if (mSignalPending) { 2220 // A signal was raised while we were unlocked 2221 mSignalPending = false; 2222 } else if (waitingAsyncCallback_l()) { 2223 if (exitPending()) { 2224 break; 2225 } 2226 releaseWakeLock_l(); 2227 mWakeLockUids.clear(); 2228 mActiveTracksGeneration++; 2229 ALOGV("wait async completion"); 2230 mWaitWorkCV.wait(mLock); 2231 ALOGV("async completion/wake"); 2232 acquireWakeLock_l(); 2233 standbyTime = systemTime() + standbyDelay; 2234 sleepTime = 0; 2235 2236 continue; 2237 } 2238 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2239 isSuspended()) { 2240 // put audio hardware into standby after short delay 2241 if (shouldStandby_l()) { 2242 2243 threadLoop_standby(); 2244 2245 mStandby = true; 2246 } 2247 2248 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2249 // we're about to wait, flush the binder command buffer 2250 IPCThreadState::self()->flushCommands(); 2251 2252 clearOutputTracks(); 2253 2254 if (exitPending()) { 2255 break; 2256 } 2257 2258 releaseWakeLock_l(); 2259 mWakeLockUids.clear(); 2260 mActiveTracksGeneration++; 2261 // wait until we have something to do... 2262 ALOGV("%s going to sleep", myName.string()); 2263 mWaitWorkCV.wait(mLock); 2264 ALOGV("%s waking up", myName.string()); 2265 acquireWakeLock_l(); 2266 2267 mMixerStatus = MIXER_IDLE; 2268 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2269 mBytesWritten = 0; 2270 mBytesRemaining = 0; 2271 checkSilentMode_l(); 2272 2273 standbyTime = systemTime() + standbyDelay; 2274 sleepTime = idleSleepTime; 2275 if (mType == MIXER) { 2276 sleepTimeShift = 0; 2277 } 2278 2279 continue; 2280 } 2281 } 2282 // mMixerStatusIgnoringFastTracks is also updated internally 2283 mMixerStatus = prepareTracks_l(&tracksToRemove); 2284 2285 // compare with previously applied list 2286 if (lastGeneration != mActiveTracksGeneration) { 2287 // update wakelock 2288 updateWakeLockUids_l(mWakeLockUids); 2289 lastGeneration = mActiveTracksGeneration; 2290 } 2291 2292 // prevent any changes in effect chain list and in each effect chain 2293 // during mixing and effect process as the audio buffers could be deleted 2294 // or modified if an effect is created or deleted 2295 lockEffectChains_l(effectChains); 2296 } // mLock scope ends 2297 2298 if (mBytesRemaining == 0) { 2299 mCurrentWriteLength = 0; 2300 if (mMixerStatus == MIXER_TRACKS_READY) { 2301 // threadLoop_mix() sets mCurrentWriteLength 2302 threadLoop_mix(); 2303 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2304 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2305 // threadLoop_sleepTime sets sleepTime to 0 if data 2306 // must be written to HAL 2307 threadLoop_sleepTime(); 2308 if (sleepTime == 0) { 2309 mCurrentWriteLength = mixBufferSize; 2310 } 2311 } 2312 mBytesRemaining = mCurrentWriteLength; 2313 if (isSuspended()) { 2314 sleepTime = suspendSleepTimeUs(); 2315 // simulate write to HAL when suspended 2316 mBytesWritten += mixBufferSize; 2317 mBytesRemaining = 0; 2318 } 2319 2320 // only process effects if we're going to write 2321 if (sleepTime == 0 && mType != OFFLOAD) { 2322 for (size_t i = 0; i < effectChains.size(); i ++) { 2323 effectChains[i]->process_l(); 2324 } 2325 } 2326 } 2327 // Process effect chains for offloaded thread even if no audio 2328 // was read from audio track: process only updates effect state 2329 // and thus does have to be synchronized with audio writes but may have 2330 // to be called while waiting for async write callback 2331 if (mType == OFFLOAD) { 2332 for (size_t i = 0; i < effectChains.size(); i ++) { 2333 effectChains[i]->process_l(); 2334 } 2335 } 2336 2337 // enable changes in effect chain 2338 unlockEffectChains(effectChains); 2339 2340 if (!waitingAsyncCallback()) { 2341 // sleepTime == 0 means we must write to audio hardware 2342 if (sleepTime == 0) { 2343 if (mBytesRemaining) { 2344 ssize_t ret = threadLoop_write(); 2345 if (ret < 0) { 2346 mBytesRemaining = 0; 2347 } else { 2348 mBytesWritten += ret; 2349 mBytesRemaining -= ret; 2350 } 2351 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2352 (mMixerStatus == MIXER_DRAIN_ALL)) { 2353 threadLoop_drain(); 2354 } 2355if (mType == MIXER) { 2356 // write blocked detection 2357 nsecs_t now = systemTime(); 2358 nsecs_t delta = now - mLastWriteTime; 2359 if (!mStandby && delta > maxPeriod) { 2360 mNumDelayedWrites++; 2361 if ((now - lastWarning) > kWarningThrottleNs) { 2362 ATRACE_NAME("underrun"); 2363 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2364 ns2ms(delta), mNumDelayedWrites, this); 2365 lastWarning = now; 2366 } 2367 } 2368} 2369 2370 } else { 2371 usleep(sleepTime); 2372 } 2373 } 2374 2375 // Finally let go of removed track(s), without the lock held 2376 // since we can't guarantee the destructors won't acquire that 2377 // same lock. This will also mutate and push a new fast mixer state. 2378 threadLoop_removeTracks(tracksToRemove); 2379 tracksToRemove.clear(); 2380 2381 // FIXME I don't understand the need for this here; 2382 // it was in the original code but maybe the 2383 // assignment in saveOutputTracks() makes this unnecessary? 2384 clearOutputTracks(); 2385 2386 // Effect chains will be actually deleted here if they were removed from 2387 // mEffectChains list during mixing or effects processing 2388 effectChains.clear(); 2389 2390 // FIXME Note that the above .clear() is no longer necessary since effectChains 2391 // is now local to this block, but will keep it for now (at least until merge done). 2392 } 2393 2394 threadLoop_exit(); 2395 2396 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2397 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2398 // put output stream into standby mode 2399 if (!mStandby) { 2400 mOutput->stream->common.standby(&mOutput->stream->common); 2401 } 2402 } 2403 2404 releaseWakeLock(); 2405 mWakeLockUids.clear(); 2406 mActiveTracksGeneration++; 2407 2408 ALOGV("Thread %p type %d exiting", this, mType); 2409 return false; 2410} 2411 2412// removeTracks_l() must be called with ThreadBase::mLock held 2413void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2414{ 2415 size_t count = tracksToRemove.size(); 2416 if (count > 0) { 2417 for (size_t i=0 ; i<count ; i++) { 2418 const sp<Track>& track = tracksToRemove.itemAt(i); 2419 mActiveTracks.remove(track); 2420 mWakeLockUids.remove(track->uid()); 2421 mActiveTracksGeneration++; 2422 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2423 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2424 if (chain != 0) { 2425 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2426 track->sessionId()); 2427 chain->decActiveTrackCnt(); 2428 } 2429 if (track->isTerminated()) { 2430 removeTrack_l(track); 2431 } 2432 } 2433 } 2434 2435} 2436 2437status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2438{ 2439 if (mNormalSink != 0) { 2440 return mNormalSink->getTimestamp(timestamp); 2441 } 2442 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2443 uint64_t position64; 2444 int ret = mOutput->stream->get_presentation_position( 2445 mOutput->stream, &position64, ×tamp.mTime); 2446 if (ret == 0) { 2447 timestamp.mPosition = (uint32_t)position64; 2448 return NO_ERROR; 2449 } 2450 } 2451 return INVALID_OPERATION; 2452} 2453// ---------------------------------------------------------------------------- 2454 2455AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2456 audio_io_handle_t id, audio_devices_t device, type_t type) 2457 : PlaybackThread(audioFlinger, output, id, device, type), 2458 // mAudioMixer below 2459 // mFastMixer below 2460 mFastMixerFutex(0) 2461 // mOutputSink below 2462 // mPipeSink below 2463 // mNormalSink below 2464{ 2465 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2466 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2467 "mFrameCount=%d, mNormalFrameCount=%d", 2468 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2469 mNormalFrameCount); 2470 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2471 2472 // FIXME - Current mixer implementation only supports stereo output 2473 if (mChannelCount != FCC_2) { 2474 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2475 } 2476 2477 // create an NBAIO sink for the HAL output stream, and negotiate 2478 mOutputSink = new AudioStreamOutSink(output->stream); 2479 size_t numCounterOffers = 0; 2480 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2481 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2482 ALOG_ASSERT(index == 0); 2483 2484 // initialize fast mixer depending on configuration 2485 bool initFastMixer; 2486 switch (kUseFastMixer) { 2487 case FastMixer_Never: 2488 initFastMixer = false; 2489 break; 2490 case FastMixer_Always: 2491 initFastMixer = true; 2492 break; 2493 case FastMixer_Static: 2494 case FastMixer_Dynamic: 2495 initFastMixer = mFrameCount < mNormalFrameCount; 2496 break; 2497 } 2498 if (initFastMixer) { 2499 2500 // create a MonoPipe to connect our submix to FastMixer 2501 NBAIO_Format format = mOutputSink->format(); 2502 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2503 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2504 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2505 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2506 const NBAIO_Format offers[1] = {format}; 2507 size_t numCounterOffers = 0; 2508 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2509 ALOG_ASSERT(index == 0); 2510 monoPipe->setAvgFrames((mScreenState & 1) ? 2511 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2512 mPipeSink = monoPipe; 2513 2514#ifdef TEE_SINK 2515 if (mTeeSinkOutputEnabled) { 2516 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2517 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2518 numCounterOffers = 0; 2519 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2520 ALOG_ASSERT(index == 0); 2521 mTeeSink = teeSink; 2522 PipeReader *teeSource = new PipeReader(*teeSink); 2523 numCounterOffers = 0; 2524 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2525 ALOG_ASSERT(index == 0); 2526 mTeeSource = teeSource; 2527 } 2528#endif 2529 2530 // create fast mixer and configure it initially with just one fast track for our submix 2531 mFastMixer = new FastMixer(); 2532 FastMixerStateQueue *sq = mFastMixer->sq(); 2533#ifdef STATE_QUEUE_DUMP 2534 sq->setObserverDump(&mStateQueueObserverDump); 2535 sq->setMutatorDump(&mStateQueueMutatorDump); 2536#endif 2537 FastMixerState *state = sq->begin(); 2538 FastTrack *fastTrack = &state->mFastTracks[0]; 2539 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2540 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2541 fastTrack->mVolumeProvider = NULL; 2542 fastTrack->mGeneration++; 2543 state->mFastTracksGen++; 2544 state->mTrackMask = 1; 2545 // fast mixer will use the HAL output sink 2546 state->mOutputSink = mOutputSink.get(); 2547 state->mOutputSinkGen++; 2548 state->mFrameCount = mFrameCount; 2549 state->mCommand = FastMixerState::COLD_IDLE; 2550 // already done in constructor initialization list 2551 //mFastMixerFutex = 0; 2552 state->mColdFutexAddr = &mFastMixerFutex; 2553 state->mColdGen++; 2554 state->mDumpState = &mFastMixerDumpState; 2555#ifdef TEE_SINK 2556 state->mTeeSink = mTeeSink.get(); 2557#endif 2558 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2559 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2560 sq->end(); 2561 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2562 2563 // start the fast mixer 2564 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2565 pid_t tid = mFastMixer->getTid(); 2566 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2567 if (err != 0) { 2568 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2569 kPriorityFastMixer, getpid_cached, tid, err); 2570 } 2571 2572#ifdef AUDIO_WATCHDOG 2573 // create and start the watchdog 2574 mAudioWatchdog = new AudioWatchdog(); 2575 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2576 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2577 tid = mAudioWatchdog->getTid(); 2578 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2579 if (err != 0) { 2580 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2581 kPriorityFastMixer, getpid_cached, tid, err); 2582 } 2583#endif 2584 2585 } else { 2586 mFastMixer = NULL; 2587 } 2588 2589 switch (kUseFastMixer) { 2590 case FastMixer_Never: 2591 case FastMixer_Dynamic: 2592 mNormalSink = mOutputSink; 2593 break; 2594 case FastMixer_Always: 2595 mNormalSink = mPipeSink; 2596 break; 2597 case FastMixer_Static: 2598 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2599 break; 2600 } 2601} 2602 2603AudioFlinger::MixerThread::~MixerThread() 2604{ 2605 if (mFastMixer != NULL) { 2606 FastMixerStateQueue *sq = mFastMixer->sq(); 2607 FastMixerState *state = sq->begin(); 2608 if (state->mCommand == FastMixerState::COLD_IDLE) { 2609 int32_t old = android_atomic_inc(&mFastMixerFutex); 2610 if (old == -1) { 2611 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2612 } 2613 } 2614 state->mCommand = FastMixerState::EXIT; 2615 sq->end(); 2616 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2617 mFastMixer->join(); 2618 // Though the fast mixer thread has exited, it's state queue is still valid. 2619 // We'll use that extract the final state which contains one remaining fast track 2620 // corresponding to our sub-mix. 2621 state = sq->begin(); 2622 ALOG_ASSERT(state->mTrackMask == 1); 2623 FastTrack *fastTrack = &state->mFastTracks[0]; 2624 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2625 delete fastTrack->mBufferProvider; 2626 sq->end(false /*didModify*/); 2627 delete mFastMixer; 2628#ifdef AUDIO_WATCHDOG 2629 if (mAudioWatchdog != 0) { 2630 mAudioWatchdog->requestExit(); 2631 mAudioWatchdog->requestExitAndWait(); 2632 mAudioWatchdog.clear(); 2633 } 2634#endif 2635 } 2636 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2637 delete mAudioMixer; 2638} 2639 2640 2641uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2642{ 2643 if (mFastMixer != NULL) { 2644 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2645 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2646 } 2647 return latency; 2648} 2649 2650 2651void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2652{ 2653 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2654} 2655 2656ssize_t AudioFlinger::MixerThread::threadLoop_write() 2657{ 2658 // FIXME we should only do one push per cycle; confirm this is true 2659 // Start the fast mixer if it's not already running 2660 if (mFastMixer != NULL) { 2661 FastMixerStateQueue *sq = mFastMixer->sq(); 2662 FastMixerState *state = sq->begin(); 2663 if (state->mCommand != FastMixerState::MIX_WRITE && 2664 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2665 if (state->mCommand == FastMixerState::COLD_IDLE) { 2666 int32_t old = android_atomic_inc(&mFastMixerFutex); 2667 if (old == -1) { 2668 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2669 } 2670#ifdef AUDIO_WATCHDOG 2671 if (mAudioWatchdog != 0) { 2672 mAudioWatchdog->resume(); 2673 } 2674#endif 2675 } 2676 state->mCommand = FastMixerState::MIX_WRITE; 2677 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2678 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2679 sq->end(); 2680 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2681 if (kUseFastMixer == FastMixer_Dynamic) { 2682 mNormalSink = mPipeSink; 2683 } 2684 } else { 2685 sq->end(false /*didModify*/); 2686 } 2687 } 2688 return PlaybackThread::threadLoop_write(); 2689} 2690 2691void AudioFlinger::MixerThread::threadLoop_standby() 2692{ 2693 // Idle the fast mixer if it's currently running 2694 if (mFastMixer != NULL) { 2695 FastMixerStateQueue *sq = mFastMixer->sq(); 2696 FastMixerState *state = sq->begin(); 2697 if (!(state->mCommand & FastMixerState::IDLE)) { 2698 state->mCommand = FastMixerState::COLD_IDLE; 2699 state->mColdFutexAddr = &mFastMixerFutex; 2700 state->mColdGen++; 2701 mFastMixerFutex = 0; 2702 sq->end(); 2703 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2704 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2705 if (kUseFastMixer == FastMixer_Dynamic) { 2706 mNormalSink = mOutputSink; 2707 } 2708#ifdef AUDIO_WATCHDOG 2709 if (mAudioWatchdog != 0) { 2710 mAudioWatchdog->pause(); 2711 } 2712#endif 2713 } else { 2714 sq->end(false /*didModify*/); 2715 } 2716 } 2717 PlaybackThread::threadLoop_standby(); 2718} 2719 2720// Empty implementation for standard mixer 2721// Overridden for offloaded playback 2722void AudioFlinger::PlaybackThread::flushOutput_l() 2723{ 2724} 2725 2726bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2727{ 2728 return false; 2729} 2730 2731bool AudioFlinger::PlaybackThread::shouldStandby_l() 2732{ 2733 return !mStandby; 2734} 2735 2736bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2737{ 2738 Mutex::Autolock _l(mLock); 2739 return waitingAsyncCallback_l(); 2740} 2741 2742// shared by MIXER and DIRECT, overridden by DUPLICATING 2743void AudioFlinger::PlaybackThread::threadLoop_standby() 2744{ 2745 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2746 mOutput->stream->common.standby(&mOutput->stream->common); 2747 if (mUseAsyncWrite != 0) { 2748 // discard any pending drain or write ack by incrementing sequence 2749 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2750 mDrainSequence = (mDrainSequence + 2) & ~1; 2751 ALOG_ASSERT(mCallbackThread != 0); 2752 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2753 mCallbackThread->setDraining(mDrainSequence); 2754 } 2755} 2756 2757void AudioFlinger::MixerThread::threadLoop_mix() 2758{ 2759 // obtain the presentation timestamp of the next output buffer 2760 int64_t pts; 2761 status_t status = INVALID_OPERATION; 2762 2763 if (mNormalSink != 0) { 2764 status = mNormalSink->getNextWriteTimestamp(&pts); 2765 } else { 2766 status = mOutputSink->getNextWriteTimestamp(&pts); 2767 } 2768 2769 if (status != NO_ERROR) { 2770 pts = AudioBufferProvider::kInvalidPTS; 2771 } 2772 2773 // mix buffers... 2774 mAudioMixer->process(pts); 2775 mCurrentWriteLength = mixBufferSize; 2776 // increase sleep time progressively when application underrun condition clears. 2777 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2778 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2779 // such that we would underrun the audio HAL. 2780 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2781 sleepTimeShift--; 2782 } 2783 sleepTime = 0; 2784 standbyTime = systemTime() + standbyDelay; 2785 //TODO: delay standby when effects have a tail 2786} 2787 2788void AudioFlinger::MixerThread::threadLoop_sleepTime() 2789{ 2790 // If no tracks are ready, sleep once for the duration of an output 2791 // buffer size, then write 0s to the output 2792 if (sleepTime == 0) { 2793 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2794 sleepTime = activeSleepTime >> sleepTimeShift; 2795 if (sleepTime < kMinThreadSleepTimeUs) { 2796 sleepTime = kMinThreadSleepTimeUs; 2797 } 2798 // reduce sleep time in case of consecutive application underruns to avoid 2799 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2800 // duration we would end up writing less data than needed by the audio HAL if 2801 // the condition persists. 2802 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2803 sleepTimeShift++; 2804 } 2805 } else { 2806 sleepTime = idleSleepTime; 2807 } 2808 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2809 memset(mMixBuffer, 0, mixBufferSize); 2810 sleepTime = 0; 2811 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2812 "anticipated start"); 2813 } 2814 // TODO add standby time extension fct of effect tail 2815} 2816 2817// prepareTracks_l() must be called with ThreadBase::mLock held 2818AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2819 Vector< sp<Track> > *tracksToRemove) 2820{ 2821 2822 mixer_state mixerStatus = MIXER_IDLE; 2823 // find out which tracks need to be processed 2824 size_t count = mActiveTracks.size(); 2825 size_t mixedTracks = 0; 2826 size_t tracksWithEffect = 0; 2827 // counts only _active_ fast tracks 2828 size_t fastTracks = 0; 2829 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2830 2831 float masterVolume = mMasterVolume; 2832 bool masterMute = mMasterMute; 2833 2834 if (masterMute) { 2835 masterVolume = 0; 2836 } 2837 // Delegate master volume control to effect in output mix effect chain if needed 2838 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2839 if (chain != 0) { 2840 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2841 chain->setVolume_l(&v, &v); 2842 masterVolume = (float)((v + (1 << 23)) >> 24); 2843 chain.clear(); 2844 } 2845 2846 // prepare a new state to push 2847 FastMixerStateQueue *sq = NULL; 2848 FastMixerState *state = NULL; 2849 bool didModify = false; 2850 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2851 if (mFastMixer != NULL) { 2852 sq = mFastMixer->sq(); 2853 state = sq->begin(); 2854 } 2855 2856 for (size_t i=0 ; i<count ; i++) { 2857 const sp<Track> t = mActiveTracks[i].promote(); 2858 if (t == 0) { 2859 continue; 2860 } 2861 2862 // this const just means the local variable doesn't change 2863 Track* const track = t.get(); 2864 2865 // process fast tracks 2866 if (track->isFastTrack()) { 2867 2868 // It's theoretically possible (though unlikely) for a fast track to be created 2869 // and then removed within the same normal mix cycle. This is not a problem, as 2870 // the track never becomes active so it's fast mixer slot is never touched. 2871 // The converse, of removing an (active) track and then creating a new track 2872 // at the identical fast mixer slot within the same normal mix cycle, 2873 // is impossible because the slot isn't marked available until the end of each cycle. 2874 int j = track->mFastIndex; 2875 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2876 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2877 FastTrack *fastTrack = &state->mFastTracks[j]; 2878 2879 // Determine whether the track is currently in underrun condition, 2880 // and whether it had a recent underrun. 2881 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2882 FastTrackUnderruns underruns = ftDump->mUnderruns; 2883 uint32_t recentFull = (underruns.mBitFields.mFull - 2884 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2885 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2886 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2887 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2888 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2889 uint32_t recentUnderruns = recentPartial + recentEmpty; 2890 track->mObservedUnderruns = underruns; 2891 // don't count underruns that occur while stopping or pausing 2892 // or stopped which can occur when flush() is called while active 2893 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2894 recentUnderruns > 0) { 2895 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2896 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2897 } 2898 2899 // This is similar to the state machine for normal tracks, 2900 // with a few modifications for fast tracks. 2901 bool isActive = true; 2902 switch (track->mState) { 2903 case TrackBase::STOPPING_1: 2904 // track stays active in STOPPING_1 state until first underrun 2905 if (recentUnderruns > 0 || track->isTerminated()) { 2906 track->mState = TrackBase::STOPPING_2; 2907 } 2908 break; 2909 case TrackBase::PAUSING: 2910 // ramp down is not yet implemented 2911 track->setPaused(); 2912 break; 2913 case TrackBase::RESUMING: 2914 // ramp up is not yet implemented 2915 track->mState = TrackBase::ACTIVE; 2916 break; 2917 case TrackBase::ACTIVE: 2918 if (recentFull > 0 || recentPartial > 0) { 2919 // track has provided at least some frames recently: reset retry count 2920 track->mRetryCount = kMaxTrackRetries; 2921 } 2922 if (recentUnderruns == 0) { 2923 // no recent underruns: stay active 2924 break; 2925 } 2926 // there has recently been an underrun of some kind 2927 if (track->sharedBuffer() == 0) { 2928 // were any of the recent underruns "empty" (no frames available)? 2929 if (recentEmpty == 0) { 2930 // no, then ignore the partial underruns as they are allowed indefinitely 2931 break; 2932 } 2933 // there has recently been an "empty" underrun: decrement the retry counter 2934 if (--(track->mRetryCount) > 0) { 2935 break; 2936 } 2937 // indicate to client process that the track was disabled because of underrun; 2938 // it will then automatically call start() when data is available 2939 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2940 // remove from active list, but state remains ACTIVE [confusing but true] 2941 isActive = false; 2942 break; 2943 } 2944 // fall through 2945 case TrackBase::STOPPING_2: 2946 case TrackBase::PAUSED: 2947 case TrackBase::STOPPED: 2948 case TrackBase::FLUSHED: // flush() while active 2949 // Check for presentation complete if track is inactive 2950 // We have consumed all the buffers of this track. 2951 // This would be incomplete if we auto-paused on underrun 2952 { 2953 size_t audioHALFrames = 2954 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2955 size_t framesWritten = mBytesWritten / mFrameSize; 2956 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2957 // track stays in active list until presentation is complete 2958 break; 2959 } 2960 } 2961 if (track->isStopping_2()) { 2962 track->mState = TrackBase::STOPPED; 2963 } 2964 if (track->isStopped()) { 2965 // Can't reset directly, as fast mixer is still polling this track 2966 // track->reset(); 2967 // So instead mark this track as needing to be reset after push with ack 2968 resetMask |= 1 << i; 2969 } 2970 isActive = false; 2971 break; 2972 case TrackBase::IDLE: 2973 default: 2974 LOG_FATAL("unexpected track state %d", track->mState); 2975 } 2976 2977 if (isActive) { 2978 // was it previously inactive? 2979 if (!(state->mTrackMask & (1 << j))) { 2980 ExtendedAudioBufferProvider *eabp = track; 2981 VolumeProvider *vp = track; 2982 fastTrack->mBufferProvider = eabp; 2983 fastTrack->mVolumeProvider = vp; 2984 fastTrack->mSampleRate = track->mSampleRate; 2985 fastTrack->mChannelMask = track->mChannelMask; 2986 fastTrack->mGeneration++; 2987 state->mTrackMask |= 1 << j; 2988 didModify = true; 2989 // no acknowledgement required for newly active tracks 2990 } 2991 // cache the combined master volume and stream type volume for fast mixer; this 2992 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2993 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2994 ++fastTracks; 2995 } else { 2996 // was it previously active? 2997 if (state->mTrackMask & (1 << j)) { 2998 fastTrack->mBufferProvider = NULL; 2999 fastTrack->mGeneration++; 3000 state->mTrackMask &= ~(1 << j); 3001 didModify = true; 3002 // If any fast tracks were removed, we must wait for acknowledgement 3003 // because we're about to decrement the last sp<> on those tracks. 3004 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3005 } else { 3006 LOG_FATAL("fast track %d should have been active", j); 3007 } 3008 tracksToRemove->add(track); 3009 // Avoids a misleading display in dumpsys 3010 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3011 } 3012 continue; 3013 } 3014 3015 { // local variable scope to avoid goto warning 3016 3017 audio_track_cblk_t* cblk = track->cblk(); 3018 3019 // The first time a track is added we wait 3020 // for all its buffers to be filled before processing it 3021 int name = track->name(); 3022 // make sure that we have enough frames to mix one full buffer. 3023 // enforce this condition only once to enable draining the buffer in case the client 3024 // app does not call stop() and relies on underrun to stop: 3025 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3026 // during last round 3027 size_t desiredFrames; 3028 uint32_t sr = track->sampleRate(); 3029 if (sr == mSampleRate) { 3030 desiredFrames = mNormalFrameCount; 3031 } else { 3032 // +1 for rounding and +1 for additional sample needed for interpolation 3033 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3034 // add frames already consumed but not yet released by the resampler 3035 // because mAudioTrackServerProxy->framesReady() will include these frames 3036 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3037 // the minimum track buffer size is normally twice the number of frames necessary 3038 // to fill one buffer and the resampler should not leave more than one buffer worth 3039 // of unreleased frames after each pass, but just in case... 3040 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3041 } 3042 uint32_t minFrames = 1; 3043 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3044 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3045 minFrames = desiredFrames; 3046 } 3047 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 3048 size_t framesReady; 3049 if (track->sharedBuffer() == 0) { 3050 framesReady = track->framesReady(); 3051 } else if (track->isStopped()) { 3052 framesReady = 0; 3053 } else { 3054 framesReady = 1; 3055 } 3056 if ((framesReady >= minFrames) && track->isReady() && 3057 !track->isPaused() && !track->isTerminated()) 3058 { 3059 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3060 3061 mixedTracks++; 3062 3063 // track->mainBuffer() != mMixBuffer means there is an effect chain 3064 // connected to the track 3065 chain.clear(); 3066 if (track->mainBuffer() != mMixBuffer) { 3067 chain = getEffectChain_l(track->sessionId()); 3068 // Delegate volume control to effect in track effect chain if needed 3069 if (chain != 0) { 3070 tracksWithEffect++; 3071 } else { 3072 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3073 "session %d", 3074 name, track->sessionId()); 3075 } 3076 } 3077 3078 3079 int param = AudioMixer::VOLUME; 3080 if (track->mFillingUpStatus == Track::FS_FILLED) { 3081 // no ramp for the first volume setting 3082 track->mFillingUpStatus = Track::FS_ACTIVE; 3083 if (track->mState == TrackBase::RESUMING) { 3084 track->mState = TrackBase::ACTIVE; 3085 param = AudioMixer::RAMP_VOLUME; 3086 } 3087 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3088 // FIXME should not make a decision based on mServer 3089 } else if (cblk->mServer != 0) { 3090 // If the track is stopped before the first frame was mixed, 3091 // do not apply ramp 3092 param = AudioMixer::RAMP_VOLUME; 3093 } 3094 3095 // compute volume for this track 3096 uint32_t vl, vr, va; 3097 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3098 vl = vr = va = 0; 3099 if (track->isPausing()) { 3100 track->setPaused(); 3101 } 3102 } else { 3103 3104 // read original volumes with volume control 3105 float typeVolume = mStreamTypes[track->streamType()].volume; 3106 float v = masterVolume * typeVolume; 3107 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3108 uint32_t vlr = proxy->getVolumeLR(); 3109 vl = vlr & 0xFFFF; 3110 vr = vlr >> 16; 3111 // track volumes come from shared memory, so can't be trusted and must be clamped 3112 if (vl > MAX_GAIN_INT) { 3113 ALOGV("Track left volume out of range: %04X", vl); 3114 vl = MAX_GAIN_INT; 3115 } 3116 if (vr > MAX_GAIN_INT) { 3117 ALOGV("Track right volume out of range: %04X", vr); 3118 vr = MAX_GAIN_INT; 3119 } 3120 // now apply the master volume and stream type volume 3121 vl = (uint32_t)(v * vl) << 12; 3122 vr = (uint32_t)(v * vr) << 12; 3123 // assuming master volume and stream type volume each go up to 1.0, 3124 // vl and vr are now in 8.24 format 3125 3126 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3127 // send level comes from shared memory and so may be corrupt 3128 if (sendLevel > MAX_GAIN_INT) { 3129 ALOGV("Track send level out of range: %04X", sendLevel); 3130 sendLevel = MAX_GAIN_INT; 3131 } 3132 va = (uint32_t)(v * sendLevel); 3133 } 3134 3135 // Delegate volume control to effect in track effect chain if needed 3136 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3137 // Do not ramp volume if volume is controlled by effect 3138 param = AudioMixer::VOLUME; 3139 track->mHasVolumeController = true; 3140 } else { 3141 // force no volume ramp when volume controller was just disabled or removed 3142 // from effect chain to avoid volume spike 3143 if (track->mHasVolumeController) { 3144 param = AudioMixer::VOLUME; 3145 } 3146 track->mHasVolumeController = false; 3147 } 3148 3149 // Convert volumes from 8.24 to 4.12 format 3150 // This additional clamping is needed in case chain->setVolume_l() overshot 3151 vl = (vl + (1 << 11)) >> 12; 3152 if (vl > MAX_GAIN_INT) { 3153 vl = MAX_GAIN_INT; 3154 } 3155 vr = (vr + (1 << 11)) >> 12; 3156 if (vr > MAX_GAIN_INT) { 3157 vr = MAX_GAIN_INT; 3158 } 3159 3160 if (va > MAX_GAIN_INT) { 3161 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3162 } 3163 3164 // XXX: these things DON'T need to be done each time 3165 mAudioMixer->setBufferProvider(name, track); 3166 mAudioMixer->enable(name); 3167 3168 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3169 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3170 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3171 mAudioMixer->setParameter( 3172 name, 3173 AudioMixer::TRACK, 3174 AudioMixer::FORMAT, (void *)track->format()); 3175 mAudioMixer->setParameter( 3176 name, 3177 AudioMixer::TRACK, 3178 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3179 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3180 uint32_t maxSampleRate = mSampleRate * 2; 3181 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3182 if (reqSampleRate == 0) { 3183 reqSampleRate = mSampleRate; 3184 } else if (reqSampleRate > maxSampleRate) { 3185 reqSampleRate = maxSampleRate; 3186 } 3187 mAudioMixer->setParameter( 3188 name, 3189 AudioMixer::RESAMPLE, 3190 AudioMixer::SAMPLE_RATE, 3191 (void *)reqSampleRate); 3192 mAudioMixer->setParameter( 3193 name, 3194 AudioMixer::TRACK, 3195 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3196 mAudioMixer->setParameter( 3197 name, 3198 AudioMixer::TRACK, 3199 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3200 3201 // reset retry count 3202 track->mRetryCount = kMaxTrackRetries; 3203 3204 // If one track is ready, set the mixer ready if: 3205 // - the mixer was not ready during previous round OR 3206 // - no other track is not ready 3207 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3208 mixerStatus != MIXER_TRACKS_ENABLED) { 3209 mixerStatus = MIXER_TRACKS_READY; 3210 } 3211 } else { 3212 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3213 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3214 } 3215 // clear effect chain input buffer if an active track underruns to avoid sending 3216 // previous audio buffer again to effects 3217 chain = getEffectChain_l(track->sessionId()); 3218 if (chain != 0) { 3219 chain->clearInputBuffer(); 3220 } 3221 3222 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3223 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3224 track->isStopped() || track->isPaused()) { 3225 // We have consumed all the buffers of this track. 3226 // Remove it from the list of active tracks. 3227 // TODO: use actual buffer filling status instead of latency when available from 3228 // audio HAL 3229 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3230 size_t framesWritten = mBytesWritten / mFrameSize; 3231 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3232 if (track->isStopped()) { 3233 track->reset(); 3234 } 3235 tracksToRemove->add(track); 3236 } 3237 } else { 3238 // No buffers for this track. Give it a few chances to 3239 // fill a buffer, then remove it from active list. 3240 if (--(track->mRetryCount) <= 0) { 3241 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3242 tracksToRemove->add(track); 3243 // indicate to client process that the track was disabled because of underrun; 3244 // it will then automatically call start() when data is available 3245 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3246 // If one track is not ready, mark the mixer also not ready if: 3247 // - the mixer was ready during previous round OR 3248 // - no other track is ready 3249 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3250 mixerStatus != MIXER_TRACKS_READY) { 3251 mixerStatus = MIXER_TRACKS_ENABLED; 3252 } 3253 } 3254 mAudioMixer->disable(name); 3255 } 3256 3257 } // local variable scope to avoid goto warning 3258track_is_ready: ; 3259 3260 } 3261 3262 // Push the new FastMixer state if necessary 3263 bool pauseAudioWatchdog = false; 3264 if (didModify) { 3265 state->mFastTracksGen++; 3266 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3267 if (kUseFastMixer == FastMixer_Dynamic && 3268 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3269 state->mCommand = FastMixerState::COLD_IDLE; 3270 state->mColdFutexAddr = &mFastMixerFutex; 3271 state->mColdGen++; 3272 mFastMixerFutex = 0; 3273 if (kUseFastMixer == FastMixer_Dynamic) { 3274 mNormalSink = mOutputSink; 3275 } 3276 // If we go into cold idle, need to wait for acknowledgement 3277 // so that fast mixer stops doing I/O. 3278 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3279 pauseAudioWatchdog = true; 3280 } 3281 } 3282 if (sq != NULL) { 3283 sq->end(didModify); 3284 sq->push(block); 3285 } 3286#ifdef AUDIO_WATCHDOG 3287 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3288 mAudioWatchdog->pause(); 3289 } 3290#endif 3291 3292 // Now perform the deferred reset on fast tracks that have stopped 3293 while (resetMask != 0) { 3294 size_t i = __builtin_ctz(resetMask); 3295 ALOG_ASSERT(i < count); 3296 resetMask &= ~(1 << i); 3297 sp<Track> t = mActiveTracks[i].promote(); 3298 if (t == 0) { 3299 continue; 3300 } 3301 Track* track = t.get(); 3302 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3303 track->reset(); 3304 } 3305 3306 // remove all the tracks that need to be... 3307 removeTracks_l(*tracksToRemove); 3308 3309 // mix buffer must be cleared if all tracks are connected to an 3310 // effect chain as in this case the mixer will not write to 3311 // mix buffer and track effects will accumulate into it 3312 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3313 (mixedTracks == 0 && fastTracks > 0))) { 3314 // FIXME as a performance optimization, should remember previous zero status 3315 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3316 } 3317 3318 // if any fast tracks, then status is ready 3319 mMixerStatusIgnoringFastTracks = mixerStatus; 3320 if (fastTracks > 0) { 3321 mixerStatus = MIXER_TRACKS_READY; 3322 } 3323 return mixerStatus; 3324} 3325 3326// getTrackName_l() must be called with ThreadBase::mLock held 3327int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3328{ 3329 return mAudioMixer->getTrackName(channelMask, sessionId); 3330} 3331 3332// deleteTrackName_l() must be called with ThreadBase::mLock held 3333void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3334{ 3335 ALOGV("remove track (%d) and delete from mixer", name); 3336 mAudioMixer->deleteTrackName(name); 3337} 3338 3339// checkForNewParameters_l() must be called with ThreadBase::mLock held 3340bool AudioFlinger::MixerThread::checkForNewParameters_l() 3341{ 3342 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3343 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3344 bool reconfig = false; 3345 3346 while (!mNewParameters.isEmpty()) { 3347 3348 if (mFastMixer != NULL) { 3349 FastMixerStateQueue *sq = mFastMixer->sq(); 3350 FastMixerState *state = sq->begin(); 3351 if (!(state->mCommand & FastMixerState::IDLE)) { 3352 previousCommand = state->mCommand; 3353 state->mCommand = FastMixerState::HOT_IDLE; 3354 sq->end(); 3355 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3356 } else { 3357 sq->end(false /*didModify*/); 3358 } 3359 } 3360 3361 status_t status = NO_ERROR; 3362 String8 keyValuePair = mNewParameters[0]; 3363 AudioParameter param = AudioParameter(keyValuePair); 3364 int value; 3365 3366 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3367 reconfig = true; 3368 } 3369 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3370 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3371 status = BAD_VALUE; 3372 } else { 3373 // no need to save value, since it's constant 3374 reconfig = true; 3375 } 3376 } 3377 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3378 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3379 status = BAD_VALUE; 3380 } else { 3381 // no need to save value, since it's constant 3382 reconfig = true; 3383 } 3384 } 3385 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3386 // do not accept frame count changes if tracks are open as the track buffer 3387 // size depends on frame count and correct behavior would not be guaranteed 3388 // if frame count is changed after track creation 3389 if (!mTracks.isEmpty()) { 3390 status = INVALID_OPERATION; 3391 } else { 3392 reconfig = true; 3393 } 3394 } 3395 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3396#ifdef ADD_BATTERY_DATA 3397 // when changing the audio output device, call addBatteryData to notify 3398 // the change 3399 if (mOutDevice != value) { 3400 uint32_t params = 0; 3401 // check whether speaker is on 3402 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3403 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3404 } 3405 3406 audio_devices_t deviceWithoutSpeaker 3407 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3408 // check if any other device (except speaker) is on 3409 if (value & deviceWithoutSpeaker ) { 3410 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3411 } 3412 3413 if (params != 0) { 3414 addBatteryData(params); 3415 } 3416 } 3417#endif 3418 3419 // forward device change to effects that have requested to be 3420 // aware of attached audio device. 3421 if (value != AUDIO_DEVICE_NONE) { 3422 mOutDevice = value; 3423 for (size_t i = 0; i < mEffectChains.size(); i++) { 3424 mEffectChains[i]->setDevice_l(mOutDevice); 3425 } 3426 } 3427 } 3428 3429 if (status == NO_ERROR) { 3430 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3431 keyValuePair.string()); 3432 if (!mStandby && status == INVALID_OPERATION) { 3433 mOutput->stream->common.standby(&mOutput->stream->common); 3434 mStandby = true; 3435 mBytesWritten = 0; 3436 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3437 keyValuePair.string()); 3438 } 3439 if (status == NO_ERROR && reconfig) { 3440 readOutputParameters(); 3441 delete mAudioMixer; 3442 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3443 for (size_t i = 0; i < mTracks.size() ; i++) { 3444 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3445 if (name < 0) { 3446 break; 3447 } 3448 mTracks[i]->mName = name; 3449 } 3450 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3451 } 3452 } 3453 3454 mNewParameters.removeAt(0); 3455 3456 mParamStatus = status; 3457 mParamCond.signal(); 3458 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3459 // already timed out waiting for the status and will never signal the condition. 3460 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3461 } 3462 3463 if (!(previousCommand & FastMixerState::IDLE)) { 3464 ALOG_ASSERT(mFastMixer != NULL); 3465 FastMixerStateQueue *sq = mFastMixer->sq(); 3466 FastMixerState *state = sq->begin(); 3467 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3468 state->mCommand = previousCommand; 3469 sq->end(); 3470 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3471 } 3472 3473 return reconfig; 3474} 3475 3476 3477void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3478{ 3479 const size_t SIZE = 256; 3480 char buffer[SIZE]; 3481 String8 result; 3482 3483 PlaybackThread::dumpInternals(fd, args); 3484 3485 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3486 result.append(buffer); 3487 write(fd, result.string(), result.size()); 3488 3489 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3490 const FastMixerDumpState copy(mFastMixerDumpState); 3491 copy.dump(fd); 3492 3493#ifdef STATE_QUEUE_DUMP 3494 // Similar for state queue 3495 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3496 observerCopy.dump(fd); 3497 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3498 mutatorCopy.dump(fd); 3499#endif 3500 3501#ifdef TEE_SINK 3502 // Write the tee output to a .wav file 3503 dumpTee(fd, mTeeSource, mId); 3504#endif 3505 3506#ifdef AUDIO_WATCHDOG 3507 if (mAudioWatchdog != 0) { 3508 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3509 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3510 wdCopy.dump(fd); 3511 } 3512#endif 3513} 3514 3515uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3516{ 3517 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3518} 3519 3520uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3521{ 3522 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3523} 3524 3525void AudioFlinger::MixerThread::cacheParameters_l() 3526{ 3527 PlaybackThread::cacheParameters_l(); 3528 3529 // FIXME: Relaxed timing because of a certain device that can't meet latency 3530 // Should be reduced to 2x after the vendor fixes the driver issue 3531 // increase threshold again due to low power audio mode. The way this warning 3532 // threshold is calculated and its usefulness should be reconsidered anyway. 3533 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3534} 3535 3536// ---------------------------------------------------------------------------- 3537 3538AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3539 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3540 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3541 // mLeftVolFloat, mRightVolFloat 3542{ 3543} 3544 3545AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3546 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3547 ThreadBase::type_t type) 3548 : PlaybackThread(audioFlinger, output, id, device, type) 3549 // mLeftVolFloat, mRightVolFloat 3550{ 3551} 3552 3553AudioFlinger::DirectOutputThread::~DirectOutputThread() 3554{ 3555} 3556 3557void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3558{ 3559 audio_track_cblk_t* cblk = track->cblk(); 3560 float left, right; 3561 3562 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3563 left = right = 0; 3564 } else { 3565 float typeVolume = mStreamTypes[track->streamType()].volume; 3566 float v = mMasterVolume * typeVolume; 3567 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3568 uint32_t vlr = proxy->getVolumeLR(); 3569 float v_clamped = v * (vlr & 0xFFFF); 3570 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3571 left = v_clamped/MAX_GAIN; 3572 v_clamped = v * (vlr >> 16); 3573 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3574 right = v_clamped/MAX_GAIN; 3575 } 3576 3577 if (lastTrack) { 3578 if (left != mLeftVolFloat || right != mRightVolFloat) { 3579 mLeftVolFloat = left; 3580 mRightVolFloat = right; 3581 3582 // Convert volumes from float to 8.24 3583 uint32_t vl = (uint32_t)(left * (1 << 24)); 3584 uint32_t vr = (uint32_t)(right * (1 << 24)); 3585 3586 // Delegate volume control to effect in track effect chain if needed 3587 // only one effect chain can be present on DirectOutputThread, so if 3588 // there is one, the track is connected to it 3589 if (!mEffectChains.isEmpty()) { 3590 mEffectChains[0]->setVolume_l(&vl, &vr); 3591 left = (float)vl / (1 << 24); 3592 right = (float)vr / (1 << 24); 3593 } 3594 if (mOutput->stream->set_volume) { 3595 mOutput->stream->set_volume(mOutput->stream, left, right); 3596 } 3597 } 3598 } 3599} 3600 3601 3602AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3603 Vector< sp<Track> > *tracksToRemove 3604) 3605{ 3606 size_t count = mActiveTracks.size(); 3607 mixer_state mixerStatus = MIXER_IDLE; 3608 3609 // find out which tracks need to be processed 3610 for (size_t i = 0; i < count; i++) { 3611 sp<Track> t = mActiveTracks[i].promote(); 3612 // The track died recently 3613 if (t == 0) { 3614 continue; 3615 } 3616 3617 Track* const track = t.get(); 3618 audio_track_cblk_t* cblk = track->cblk(); 3619 // Only consider last track started for volume and mixer state control. 3620 // In theory an older track could underrun and restart after the new one starts 3621 // but as we only care about the transition phase between two tracks on a 3622 // direct output, it is not a problem to ignore the underrun case. 3623 sp<Track> l = mLatestActiveTrack.promote(); 3624 bool last = l.get() == track; 3625 3626 // The first time a track is added we wait 3627 // for all its buffers to be filled before processing it 3628 uint32_t minFrames; 3629 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3630 minFrames = mNormalFrameCount; 3631 } else { 3632 minFrames = 1; 3633 } 3634 3635 if ((track->framesReady() >= minFrames) && track->isReady() && 3636 !track->isPaused() && !track->isTerminated()) 3637 { 3638 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3639 3640 if (track->mFillingUpStatus == Track::FS_FILLED) { 3641 track->mFillingUpStatus = Track::FS_ACTIVE; 3642 // make sure processVolume_l() will apply new volume even if 0 3643 mLeftVolFloat = mRightVolFloat = -1.0; 3644 if (track->mState == TrackBase::RESUMING) { 3645 track->mState = TrackBase::ACTIVE; 3646 } 3647 } 3648 3649 // compute volume for this track 3650 processVolume_l(track, last); 3651 if (last) { 3652 // reset retry count 3653 track->mRetryCount = kMaxTrackRetriesDirect; 3654 mActiveTrack = t; 3655 mixerStatus = MIXER_TRACKS_READY; 3656 } 3657 } else { 3658 // clear effect chain input buffer if the last active track started underruns 3659 // to avoid sending previous audio buffer again to effects 3660 if (!mEffectChains.isEmpty() && last) { 3661 mEffectChains[0]->clearInputBuffer(); 3662 } 3663 3664 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3665 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3666 track->isStopped() || track->isPaused()) { 3667 // We have consumed all the buffers of this track. 3668 // Remove it from the list of active tracks. 3669 // TODO: implement behavior for compressed audio 3670 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3671 size_t framesWritten = mBytesWritten / mFrameSize; 3672 if (mStandby || !last || 3673 track->presentationComplete(framesWritten, audioHALFrames)) { 3674 if (track->isStopped()) { 3675 track->reset(); 3676 } 3677 tracksToRemove->add(track); 3678 } 3679 } else { 3680 // No buffers for this track. Give it a few chances to 3681 // fill a buffer, then remove it from active list. 3682 // Only consider last track started for mixer state control 3683 if (--(track->mRetryCount) <= 0) { 3684 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3685 tracksToRemove->add(track); 3686 // indicate to client process that the track was disabled because of underrun; 3687 // it will then automatically call start() when data is available 3688 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3689 } else if (last) { 3690 mixerStatus = MIXER_TRACKS_ENABLED; 3691 } 3692 } 3693 } 3694 } 3695 3696 // remove all the tracks that need to be... 3697 removeTracks_l(*tracksToRemove); 3698 3699 return mixerStatus; 3700} 3701 3702void AudioFlinger::DirectOutputThread::threadLoop_mix() 3703{ 3704 size_t frameCount = mFrameCount; 3705 int8_t *curBuf = (int8_t *)mMixBuffer; 3706 // output audio to hardware 3707 while (frameCount) { 3708 AudioBufferProvider::Buffer buffer; 3709 buffer.frameCount = frameCount; 3710 mActiveTrack->getNextBuffer(&buffer); 3711 if (buffer.raw == NULL) { 3712 memset(curBuf, 0, frameCount * mFrameSize); 3713 break; 3714 } 3715 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3716 frameCount -= buffer.frameCount; 3717 curBuf += buffer.frameCount * mFrameSize; 3718 mActiveTrack->releaseBuffer(&buffer); 3719 } 3720 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3721 sleepTime = 0; 3722 standbyTime = systemTime() + standbyDelay; 3723 mActiveTrack.clear(); 3724} 3725 3726void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3727{ 3728 if (sleepTime == 0) { 3729 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3730 sleepTime = activeSleepTime; 3731 } else { 3732 sleepTime = idleSleepTime; 3733 } 3734 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3735 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3736 sleepTime = 0; 3737 } 3738} 3739 3740// getTrackName_l() must be called with ThreadBase::mLock held 3741int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3742 int sessionId) 3743{ 3744 return 0; 3745} 3746 3747// deleteTrackName_l() must be called with ThreadBase::mLock held 3748void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3749{ 3750} 3751 3752// checkForNewParameters_l() must be called with ThreadBase::mLock held 3753bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3754{ 3755 bool reconfig = false; 3756 3757 while (!mNewParameters.isEmpty()) { 3758 status_t status = NO_ERROR; 3759 String8 keyValuePair = mNewParameters[0]; 3760 AudioParameter param = AudioParameter(keyValuePair); 3761 int value; 3762 3763 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3764 // do not accept frame count changes if tracks are open as the track buffer 3765 // size depends on frame count and correct behavior would not be garantied 3766 // if frame count is changed after track creation 3767 if (!mTracks.isEmpty()) { 3768 status = INVALID_OPERATION; 3769 } else { 3770 reconfig = true; 3771 } 3772 } 3773 if (status == NO_ERROR) { 3774 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3775 keyValuePair.string()); 3776 if (!mStandby && status == INVALID_OPERATION) { 3777 mOutput->stream->common.standby(&mOutput->stream->common); 3778 mStandby = true; 3779 mBytesWritten = 0; 3780 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3781 keyValuePair.string()); 3782 } 3783 if (status == NO_ERROR && reconfig) { 3784 readOutputParameters(); 3785 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3786 } 3787 } 3788 3789 mNewParameters.removeAt(0); 3790 3791 mParamStatus = status; 3792 mParamCond.signal(); 3793 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3794 // already timed out waiting for the status and will never signal the condition. 3795 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3796 } 3797 return reconfig; 3798} 3799 3800uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3801{ 3802 uint32_t time; 3803 if (audio_is_linear_pcm(mFormat)) { 3804 time = PlaybackThread::activeSleepTimeUs(); 3805 } else { 3806 time = 10000; 3807 } 3808 return time; 3809} 3810 3811uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3812{ 3813 uint32_t time; 3814 if (audio_is_linear_pcm(mFormat)) { 3815 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3816 } else { 3817 time = 10000; 3818 } 3819 return time; 3820} 3821 3822uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3823{ 3824 uint32_t time; 3825 if (audio_is_linear_pcm(mFormat)) { 3826 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3827 } else { 3828 time = 10000; 3829 } 3830 return time; 3831} 3832 3833void AudioFlinger::DirectOutputThread::cacheParameters_l() 3834{ 3835 PlaybackThread::cacheParameters_l(); 3836 3837 // use shorter standby delay as on normal output to release 3838 // hardware resources as soon as possible 3839 if (audio_is_linear_pcm(mFormat)) { 3840 standbyDelay = microseconds(activeSleepTime*2); 3841 } else { 3842 standbyDelay = kOffloadStandbyDelayNs; 3843 } 3844} 3845 3846// ---------------------------------------------------------------------------- 3847 3848AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3849 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3850 : Thread(false /*canCallJava*/), 3851 mPlaybackThread(playbackThread), 3852 mWriteAckSequence(0), 3853 mDrainSequence(0) 3854{ 3855} 3856 3857AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3858{ 3859} 3860 3861void AudioFlinger::AsyncCallbackThread::onFirstRef() 3862{ 3863 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3864} 3865 3866bool AudioFlinger::AsyncCallbackThread::threadLoop() 3867{ 3868 while (!exitPending()) { 3869 uint32_t writeAckSequence; 3870 uint32_t drainSequence; 3871 3872 { 3873 Mutex::Autolock _l(mLock); 3874 mWaitWorkCV.wait(mLock); 3875 if (exitPending()) { 3876 break; 3877 } 3878 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3879 mWriteAckSequence, mDrainSequence); 3880 writeAckSequence = mWriteAckSequence; 3881 mWriteAckSequence &= ~1; 3882 drainSequence = mDrainSequence; 3883 mDrainSequence &= ~1; 3884 } 3885 { 3886 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3887 if (playbackThread != 0) { 3888 if (writeAckSequence & 1) { 3889 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3890 } 3891 if (drainSequence & 1) { 3892 playbackThread->resetDraining(drainSequence >> 1); 3893 } 3894 } 3895 } 3896 } 3897 return false; 3898} 3899 3900void AudioFlinger::AsyncCallbackThread::exit() 3901{ 3902 ALOGV("AsyncCallbackThread::exit"); 3903 Mutex::Autolock _l(mLock); 3904 requestExit(); 3905 mWaitWorkCV.broadcast(); 3906} 3907 3908void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3909{ 3910 Mutex::Autolock _l(mLock); 3911 // bit 0 is cleared 3912 mWriteAckSequence = sequence << 1; 3913} 3914 3915void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3916{ 3917 Mutex::Autolock _l(mLock); 3918 // ignore unexpected callbacks 3919 if (mWriteAckSequence & 2) { 3920 mWriteAckSequence |= 1; 3921 mWaitWorkCV.signal(); 3922 } 3923} 3924 3925void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3926{ 3927 Mutex::Autolock _l(mLock); 3928 // bit 0 is cleared 3929 mDrainSequence = sequence << 1; 3930} 3931 3932void AudioFlinger::AsyncCallbackThread::resetDraining() 3933{ 3934 Mutex::Autolock _l(mLock); 3935 // ignore unexpected callbacks 3936 if (mDrainSequence & 2) { 3937 mDrainSequence |= 1; 3938 mWaitWorkCV.signal(); 3939 } 3940} 3941 3942 3943// ---------------------------------------------------------------------------- 3944AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3945 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3946 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3947 mHwPaused(false), 3948 mFlushPending(false), 3949 mPausedBytesRemaining(0), 3950 mPreviousTrack(NULL) 3951{ 3952 //FIXME: mStandby should be set to true by ThreadBase constructor 3953 mStandby = true; 3954} 3955 3956void AudioFlinger::OffloadThread::threadLoop_exit() 3957{ 3958 if (mFlushPending || mHwPaused) { 3959 // If a flush is pending or track was paused, just discard buffered data 3960 flushHw_l(); 3961 } else { 3962 mMixerStatus = MIXER_DRAIN_ALL; 3963 threadLoop_drain(); 3964 } 3965 mCallbackThread->exit(); 3966 PlaybackThread::threadLoop_exit(); 3967} 3968 3969AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3970 Vector< sp<Track> > *tracksToRemove 3971) 3972{ 3973 size_t count = mActiveTracks.size(); 3974 3975 mixer_state mixerStatus = MIXER_IDLE; 3976 bool doHwPause = false; 3977 bool doHwResume = false; 3978 3979 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3980 3981 // find out which tracks need to be processed 3982 for (size_t i = 0; i < count; i++) { 3983 sp<Track> t = mActiveTracks[i].promote(); 3984 // The track died recently 3985 if (t == 0) { 3986 continue; 3987 } 3988 Track* const track = t.get(); 3989 audio_track_cblk_t* cblk = track->cblk(); 3990 // Only consider last track started for volume and mixer state control. 3991 // In theory an older track could underrun and restart after the new one starts 3992 // but as we only care about the transition phase between two tracks on a 3993 // direct output, it is not a problem to ignore the underrun case. 3994 sp<Track> l = mLatestActiveTrack.promote(); 3995 bool last = l.get() == track; 3996 3997 if (mPreviousTrack != NULL) { 3998 if (track != mPreviousTrack) { 3999 // Flush any data still being written from last track 4000 mBytesRemaining = 0; 4001 if (mPausedBytesRemaining) { 4002 // Last track was paused so we also need to flush saved 4003 // mixbuffer state and invalidate track so that it will 4004 // re-submit that unwritten data when it is next resumed 4005 mPausedBytesRemaining = 0; 4006 // Invalidate is a bit drastic - would be more efficient 4007 // to have a flag to tell client that some of the 4008 // previously written data was lost 4009 mPreviousTrack->invalidate(); 4010 } 4011 } 4012 } 4013 mPreviousTrack = track; 4014 if (track->isPausing()) { 4015 track->setPaused(); 4016 if (last) { 4017 if (!mHwPaused) { 4018 doHwPause = true; 4019 mHwPaused = true; 4020 } 4021 // If we were part way through writing the mixbuffer to 4022 // the HAL we must save this until we resume 4023 // BUG - this will be wrong if a different track is made active, 4024 // in that case we want to discard the pending data in the 4025 // mixbuffer and tell the client to present it again when the 4026 // track is resumed 4027 mPausedWriteLength = mCurrentWriteLength; 4028 mPausedBytesRemaining = mBytesRemaining; 4029 mBytesRemaining = 0; // stop writing 4030 } 4031 tracksToRemove->add(track); 4032 } else if (track->framesReady() && track->isReady() && 4033 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4034 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4035 if (track->mFillingUpStatus == Track::FS_FILLED) { 4036 track->mFillingUpStatus = Track::FS_ACTIVE; 4037 // make sure processVolume_l() will apply new volume even if 0 4038 mLeftVolFloat = mRightVolFloat = -1.0; 4039 if (track->mState == TrackBase::RESUMING) { 4040 track->mState = TrackBase::ACTIVE; 4041 if (last) { 4042 if (mPausedBytesRemaining) { 4043 // Need to continue write that was interrupted 4044 mCurrentWriteLength = mPausedWriteLength; 4045 mBytesRemaining = mPausedBytesRemaining; 4046 mPausedBytesRemaining = 0; 4047 } 4048 if (mHwPaused) { 4049 doHwResume = true; 4050 mHwPaused = false; 4051 // threadLoop_mix() will handle the case that we need to 4052 // resume an interrupted write 4053 } 4054 // enable write to audio HAL 4055 sleepTime = 0; 4056 } 4057 } 4058 } 4059 4060 if (last) { 4061 // reset retry count 4062 track->mRetryCount = kMaxTrackRetriesOffload; 4063 mActiveTrack = t; 4064 mixerStatus = MIXER_TRACKS_READY; 4065 } 4066 } else { 4067 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4068 if (track->isStopping_1()) { 4069 // Hardware buffer can hold a large amount of audio so we must 4070 // wait for all current track's data to drain before we say 4071 // that the track is stopped. 4072 if (mBytesRemaining == 0) { 4073 // Only start draining when all data in mixbuffer 4074 // has been written 4075 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4076 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4077 // do not drain if no data was ever sent to HAL (mStandby == true) 4078 if (last && !mStandby) { 4079 sleepTime = 0; 4080 standbyTime = systemTime() + standbyDelay; 4081 mixerStatus = MIXER_DRAIN_TRACK; 4082 mDrainSequence += 2; 4083 if (mHwPaused) { 4084 // It is possible to move from PAUSED to STOPPING_1 without 4085 // a resume so we must ensure hardware is running 4086 mOutput->stream->resume(mOutput->stream); 4087 mHwPaused = false; 4088 } 4089 } 4090 } 4091 } else if (track->isStopping_2()) { 4092 // Drain has completed or we are in standby, signal presentation complete 4093 if (!(mDrainSequence & 1) || !last || mStandby) { 4094 track->mState = TrackBase::STOPPED; 4095 size_t audioHALFrames = 4096 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4097 size_t framesWritten = 4098 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4099 track->presentationComplete(framesWritten, audioHALFrames); 4100 track->reset(); 4101 tracksToRemove->add(track); 4102 } 4103 } else { 4104 // No buffers for this track. Give it a few chances to 4105 // fill a buffer, then remove it from active list. 4106 if (--(track->mRetryCount) <= 0) { 4107 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4108 track->name()); 4109 tracksToRemove->add(track); 4110 // indicate to client process that the track was disabled because of underrun; 4111 // it will then automatically call start() when data is available 4112 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4113 } else if (last){ 4114 mixerStatus = MIXER_TRACKS_ENABLED; 4115 } 4116 } 4117 } 4118 // compute volume for this track 4119 processVolume_l(track, last); 4120 } 4121 4122 // make sure the pause/flush/resume sequence is executed in the right order. 4123 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4124 // before flush and then resume HW. This can happen in case of pause/flush/resume 4125 // if resume is received before pause is executed. 4126 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4127 mOutput->stream->pause(mOutput->stream); 4128 if (!doHwPause) { 4129 doHwResume = true; 4130 } 4131 } 4132 if (mFlushPending) { 4133 flushHw_l(); 4134 mFlushPending = false; 4135 } 4136 if (!mStandby && doHwResume) { 4137 mOutput->stream->resume(mOutput->stream); 4138 } 4139 4140 // remove all the tracks that need to be... 4141 removeTracks_l(*tracksToRemove); 4142 4143 return mixerStatus; 4144} 4145 4146void AudioFlinger::OffloadThread::flushOutput_l() 4147{ 4148 mFlushPending = true; 4149} 4150 4151// must be called with thread mutex locked 4152bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4153{ 4154 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4155 mWriteAckSequence, mDrainSequence); 4156 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4157 return true; 4158 } 4159 return false; 4160} 4161 4162// must be called with thread mutex locked 4163bool AudioFlinger::OffloadThread::shouldStandby_l() 4164{ 4165 bool trackPaused = false; 4166 4167 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4168 // after a timeout and we will enter standby then. 4169 if (mTracks.size() > 0) { 4170 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4171 } 4172 4173 return !mStandby && !trackPaused; 4174} 4175 4176 4177bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4178{ 4179 Mutex::Autolock _l(mLock); 4180 return waitingAsyncCallback_l(); 4181} 4182 4183void AudioFlinger::OffloadThread::flushHw_l() 4184{ 4185 mOutput->stream->flush(mOutput->stream); 4186 // Flush anything still waiting in the mixbuffer 4187 mCurrentWriteLength = 0; 4188 mBytesRemaining = 0; 4189 mPausedWriteLength = 0; 4190 mPausedBytesRemaining = 0; 4191 if (mUseAsyncWrite) { 4192 // discard any pending drain or write ack by incrementing sequence 4193 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4194 mDrainSequence = (mDrainSequence + 2) & ~1; 4195 ALOG_ASSERT(mCallbackThread != 0); 4196 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4197 mCallbackThread->setDraining(mDrainSequence); 4198 } 4199} 4200 4201// ---------------------------------------------------------------------------- 4202 4203AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4204 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4205 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4206 DUPLICATING), 4207 mWaitTimeMs(UINT_MAX) 4208{ 4209 addOutputTrack(mainThread); 4210} 4211 4212AudioFlinger::DuplicatingThread::~DuplicatingThread() 4213{ 4214 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4215 mOutputTracks[i]->destroy(); 4216 } 4217} 4218 4219void AudioFlinger::DuplicatingThread::threadLoop_mix() 4220{ 4221 // mix buffers... 4222 if (outputsReady(outputTracks)) { 4223 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4224 } else { 4225 memset(mMixBuffer, 0, mixBufferSize); 4226 } 4227 sleepTime = 0; 4228 writeFrames = mNormalFrameCount; 4229 mCurrentWriteLength = mixBufferSize; 4230 standbyTime = systemTime() + standbyDelay; 4231} 4232 4233void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4234{ 4235 if (sleepTime == 0) { 4236 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4237 sleepTime = activeSleepTime; 4238 } else { 4239 sleepTime = idleSleepTime; 4240 } 4241 } else if (mBytesWritten != 0) { 4242 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4243 writeFrames = mNormalFrameCount; 4244 memset(mMixBuffer, 0, mixBufferSize); 4245 } else { 4246 // flush remaining overflow buffers in output tracks 4247 writeFrames = 0; 4248 } 4249 sleepTime = 0; 4250 } 4251} 4252 4253ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4254{ 4255 for (size_t i = 0; i < outputTracks.size(); i++) { 4256 outputTracks[i]->write(mMixBuffer, writeFrames); 4257 } 4258 mStandby = false; 4259 return (ssize_t)mixBufferSize; 4260} 4261 4262void AudioFlinger::DuplicatingThread::threadLoop_standby() 4263{ 4264 // DuplicatingThread implements standby by stopping all tracks 4265 for (size_t i = 0; i < outputTracks.size(); i++) { 4266 outputTracks[i]->stop(); 4267 } 4268} 4269 4270void AudioFlinger::DuplicatingThread::saveOutputTracks() 4271{ 4272 outputTracks = mOutputTracks; 4273} 4274 4275void AudioFlinger::DuplicatingThread::clearOutputTracks() 4276{ 4277 outputTracks.clear(); 4278} 4279 4280void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4281{ 4282 Mutex::Autolock _l(mLock); 4283 // FIXME explain this formula 4284 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4285 OutputTrack *outputTrack = new OutputTrack(thread, 4286 this, 4287 mSampleRate, 4288 mFormat, 4289 mChannelMask, 4290 frameCount, 4291 IPCThreadState::self()->getCallingUid()); 4292 if (outputTrack->cblk() != NULL) { 4293 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4294 mOutputTracks.add(outputTrack); 4295 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4296 updateWaitTime_l(); 4297 } 4298} 4299 4300void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4301{ 4302 Mutex::Autolock _l(mLock); 4303 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4304 if (mOutputTracks[i]->thread() == thread) { 4305 mOutputTracks[i]->destroy(); 4306 mOutputTracks.removeAt(i); 4307 updateWaitTime_l(); 4308 return; 4309 } 4310 } 4311 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4312} 4313 4314// caller must hold mLock 4315void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4316{ 4317 mWaitTimeMs = UINT_MAX; 4318 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4319 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4320 if (strong != 0) { 4321 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4322 if (waitTimeMs < mWaitTimeMs) { 4323 mWaitTimeMs = waitTimeMs; 4324 } 4325 } 4326 } 4327} 4328 4329 4330bool AudioFlinger::DuplicatingThread::outputsReady( 4331 const SortedVector< sp<OutputTrack> > &outputTracks) 4332{ 4333 for (size_t i = 0; i < outputTracks.size(); i++) { 4334 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4335 if (thread == 0) { 4336 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4337 outputTracks[i].get()); 4338 return false; 4339 } 4340 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4341 // see note at standby() declaration 4342 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4343 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4344 thread.get()); 4345 return false; 4346 } 4347 } 4348 return true; 4349} 4350 4351uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4352{ 4353 return (mWaitTimeMs * 1000) / 2; 4354} 4355 4356void AudioFlinger::DuplicatingThread::cacheParameters_l() 4357{ 4358 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4359 updateWaitTime_l(); 4360 4361 MixerThread::cacheParameters_l(); 4362} 4363 4364// ---------------------------------------------------------------------------- 4365// Record 4366// ---------------------------------------------------------------------------- 4367 4368AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4369 AudioStreamIn *input, 4370 uint32_t sampleRate, 4371 audio_channel_mask_t channelMask, 4372 audio_io_handle_t id, 4373 audio_devices_t outDevice, 4374 audio_devices_t inDevice 4375#ifdef TEE_SINK 4376 , const sp<NBAIO_Sink>& teeSink 4377#endif 4378 ) : 4379 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4380 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4381 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4382 // are set by readInputParameters() 4383 // mRsmpInIndex LEGACY 4384 mReqChannelCount(popcount(channelMask)), 4385 mReqSampleRate(sampleRate) 4386 // mBytesRead is only meaningful while active, and so is cleared in start() 4387 // (but might be better to also clear here for dump?) 4388#ifdef TEE_SINK 4389 , mTeeSink(teeSink) 4390#endif 4391{ 4392 snprintf(mName, kNameLength, "AudioIn_%X", id); 4393 4394 readInputParameters(); 4395} 4396 4397 4398AudioFlinger::RecordThread::~RecordThread() 4399{ 4400 delete[] mRsmpInBuffer; 4401 delete mResampler; 4402 delete[] mRsmpOutBuffer; 4403} 4404 4405void AudioFlinger::RecordThread::onFirstRef() 4406{ 4407 run(mName, PRIORITY_URGENT_AUDIO); 4408} 4409 4410bool AudioFlinger::RecordThread::threadLoop() 4411{ 4412 nsecs_t lastWarning = 0; 4413 4414 inputStandBy(); 4415 sp<RecordTrack> activeTrack; 4416 { 4417 Mutex::Autolock _l(mLock); 4418 activeTrack = mActiveTrack; 4419 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4420 } 4421 4422 // used to verify we've read at least once before evaluating how many bytes were read 4423 bool readOnce = false; 4424 4425 // used to request a deferred sleep, to be executed later while mutex is unlocked 4426 bool doSleep = false; 4427 4428 // start recording 4429 for (;;) { 4430 TrackBase::track_state activeTrackState; 4431 Vector< sp<EffectChain> > effectChains; 4432 4433 // sleep with mutex unlocked 4434 if (doSleep) { 4435 doSleep = false; 4436 usleep(kRecordThreadSleepUs); 4437 } 4438 4439 { // scope for mLock 4440 Mutex::Autolock _l(mLock); 4441 if (exitPending()) { 4442 break; 4443 } 4444 processConfigEvents_l(); 4445 // return value 'reconfig' is currently unused 4446 bool reconfig = checkForNewParameters_l(); 4447 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4448 SortedVector<int> tmp; 4449 tmp.add(mActiveTrack->uid()); 4450 updateWakeLockUids_l(tmp); 4451 } 4452 // make a stable copy of mActiveTrack 4453 activeTrack = mActiveTrack; 4454 if (activeTrack == 0) { 4455 standbyIfNotAlreadyInStandby(); 4456 // exitPending() can't become true here 4457 releaseWakeLock_l(); 4458 ALOGV("RecordThread: loop stopping"); 4459 // go to sleep 4460 mWaitWorkCV.wait(mLock); 4461 ALOGV("RecordThread: loop starting"); 4462 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4463 continue; 4464 } 4465 4466 if (activeTrack->isTerminated()) { 4467 removeTrack_l(activeTrack); 4468 mActiveTrack.clear(); 4469 continue; 4470 } 4471 4472 activeTrackState = activeTrack->mState; 4473 switch (activeTrackState) { 4474 case TrackBase::PAUSING: 4475 standbyIfNotAlreadyInStandby(); 4476 mActiveTrack.clear(); 4477 mStartStopCond.broadcast(); 4478 doSleep = true; 4479 continue; 4480 4481 case TrackBase::RESUMING: 4482 mStandby = false; 4483 if (mReqChannelCount != activeTrack->channelCount()) { 4484 mActiveTrack.clear(); 4485 mStartStopCond.broadcast(); 4486 continue; 4487 } 4488 if (readOnce) { 4489 mStartStopCond.broadcast(); 4490 // record start succeeds only if first read from audio input succeeds 4491 if (mBytesRead < 0) { 4492 mActiveTrack.clear(); 4493 continue; 4494 } 4495 activeTrack->mState = TrackBase::ACTIVE; 4496 } 4497 break; 4498 4499 case TrackBase::ACTIVE: 4500 break; 4501 4502 case TrackBase::IDLE: 4503 doSleep = true; 4504 continue; 4505 4506 default: 4507 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4508 } 4509 4510 lockEffectChains_l(effectChains); 4511 } 4512 4513 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4514 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4515 4516 for (size_t i = 0; i < effectChains.size(); i ++) { 4517 // thread mutex is not locked, but effect chain is locked 4518 effectChains[i]->process_l(); 4519 } 4520 4521 AudioBufferProvider::Buffer buffer; 4522 buffer.frameCount = mFrameCount; 4523 status_t status = activeTrack->getNextBuffer(&buffer); 4524 if (status == NO_ERROR) { 4525 readOnce = true; 4526 size_t framesOut = buffer.frameCount; 4527 if (mResampler == NULL) { 4528 // no resampling 4529 while (framesOut) { 4530 size_t framesIn = mFrameCount - mRsmpInIndex; 4531 if (framesIn > 0) { 4532 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4533 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4534 activeTrack->mFrameSize; 4535 if (framesIn > framesOut) { 4536 framesIn = framesOut; 4537 } 4538 mRsmpInIndex += framesIn; 4539 framesOut -= framesIn; 4540 if (mChannelCount == mReqChannelCount) { 4541 memcpy(dst, src, framesIn * mFrameSize); 4542 } else { 4543 if (mChannelCount == 1) { 4544 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4545 (int16_t *)src, framesIn); 4546 } else { 4547 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4548 (int16_t *)src, framesIn); 4549 } 4550 } 4551 } 4552 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4553 void *readInto; 4554 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4555 readInto = buffer.raw; 4556 framesOut = 0; 4557 } else { 4558 readInto = mRsmpInBuffer; 4559 mRsmpInIndex = 0; 4560 } 4561 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4562 mBufferSize); 4563 if (mBytesRead <= 0) { 4564 // TODO: verify that it's benign to use a stale track state 4565 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4566 { 4567 ALOGE("Error reading audio input"); 4568 // Force input into standby so that it tries to 4569 // recover at next read attempt 4570 inputStandBy(); 4571 doSleep = true; 4572 } 4573 mRsmpInIndex = mFrameCount; 4574 framesOut = 0; 4575 buffer.frameCount = 0; 4576 } 4577#ifdef TEE_SINK 4578 else if (mTeeSink != 0) { 4579 (void) mTeeSink->write(readInto, 4580 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4581 } 4582#endif 4583 } 4584 } 4585 } else { 4586 // resampling 4587 4588 // avoid busy-waiting if client doesn't keep up 4589 bool madeProgress = false; 4590 4591 // keep mRsmpInBuffer full so resampler always has sufficient input 4592 for (;;) { 4593 int32_t rear = mRsmpInRear; 4594 ssize_t filled = rear - mRsmpInFront; 4595 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4596 // exit once there is enough data in buffer for resampler 4597 if ((size_t) filled >= mRsmpInFrames) { 4598 break; 4599 } 4600 size_t avail = mRsmpInFramesP2 - filled; 4601 // Only try to read full HAL buffers. 4602 // But if the HAL read returns a partial buffer, use it. 4603 if (avail < mFrameCount) { 4604 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4605 avail, mFrameCount); 4606 break; 4607 } 4608 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4609 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4610 rear &= mRsmpInFramesP2 - 1; 4611 mBytesRead = mInput->stream->read(mInput->stream, 4612 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4613 if (mBytesRead <= 0) { 4614 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4615 break; 4616 } 4617 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4618 size_t framesRead = mBytesRead / mFrameSize; 4619 ALOG_ASSERT(framesRead > 0); 4620 madeProgress = true; 4621 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4622 size_t part1 = mRsmpInFramesP2 - rear; 4623 if (framesRead > part1) { 4624 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4625 (framesRead - part1) * mFrameSize); 4626 } 4627 mRsmpInRear += framesRead; 4628 } 4629 4630 if (!madeProgress) { 4631 ALOGV("Did not make progress"); 4632 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4633 } 4634 4635 // resampler accumulates, but we only have one source track 4636 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4637 mResampler->resample(mRsmpOutBuffer, framesOut, 4638 this /* AudioBufferProvider* */); 4639 // ditherAndClamp() works as long as all buffers returned by 4640 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4641 if (mReqChannelCount == 1) { 4642 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4643 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4644 // the resampler always outputs stereo samples: 4645 // do post stereo to mono conversion 4646 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4647 framesOut); 4648 } else { 4649 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4650 } 4651 // now done with mRsmpOutBuffer 4652 4653 } 4654 if (mFramestoDrop == 0) { 4655 activeTrack->releaseBuffer(&buffer); 4656 } else { 4657 if (mFramestoDrop > 0) { 4658 mFramestoDrop -= buffer.frameCount; 4659 if (mFramestoDrop <= 0) { 4660 clearSyncStartEvent(); 4661 } 4662 } else { 4663 mFramestoDrop += buffer.frameCount; 4664 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4665 mSyncStartEvent->isCancelled()) { 4666 ALOGW("Synced record %s, session %d, trigger session %d", 4667 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4668 activeTrack->sessionId(), 4669 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4670 clearSyncStartEvent(); 4671 } 4672 } 4673 } 4674 activeTrack->clearOverflow(); 4675 } 4676 // client isn't retrieving buffers fast enough 4677 else { 4678 if (!activeTrack->setOverflow()) { 4679 nsecs_t now = systemTime(); 4680 if ((now - lastWarning) > kWarningThrottleNs) { 4681 ALOGW("RecordThread: buffer overflow"); 4682 lastWarning = now; 4683 } 4684 } 4685 // Release the processor for a while before asking for a new buffer. 4686 // This will give the application more chance to read from the buffer and 4687 // clear the overflow. 4688 doSleep = true; 4689 } 4690 4691 // enable changes in effect chain 4692 unlockEffectChains(effectChains); 4693 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4694 } 4695 4696 standbyIfNotAlreadyInStandby(); 4697 4698 { 4699 Mutex::Autolock _l(mLock); 4700 for (size_t i = 0; i < mTracks.size(); i++) { 4701 sp<RecordTrack> track = mTracks[i]; 4702 track->invalidate(); 4703 } 4704 mActiveTrack.clear(); 4705 mStartStopCond.broadcast(); 4706 } 4707 4708 releaseWakeLock(); 4709 4710 ALOGV("RecordThread %p exiting", this); 4711 return false; 4712} 4713 4714void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4715{ 4716 if (!mStandby) { 4717 inputStandBy(); 4718 mStandby = true; 4719 } 4720} 4721 4722void AudioFlinger::RecordThread::inputStandBy() 4723{ 4724 mInput->stream->common.standby(&mInput->stream->common); 4725} 4726 4727sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4728 const sp<AudioFlinger::Client>& client, 4729 uint32_t sampleRate, 4730 audio_format_t format, 4731 audio_channel_mask_t channelMask, 4732 size_t frameCount, 4733 int sessionId, 4734 int uid, 4735 IAudioFlinger::track_flags_t *flags, 4736 pid_t tid, 4737 status_t *status) 4738{ 4739 sp<RecordTrack> track; 4740 status_t lStatus; 4741 4742 lStatus = initCheck(); 4743 if (lStatus != NO_ERROR) { 4744 ALOGE("createRecordTrack_l() audio driver not initialized"); 4745 goto Exit; 4746 } 4747 // client expresses a preference for FAST, but we get the final say 4748 if (*flags & IAudioFlinger::TRACK_FAST) { 4749 if ( 4750 // use case: callback handler and frame count is default or at least as large as HAL 4751 ( 4752 (tid != -1) && 4753 ((frameCount == 0) || 4754 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4755 ) && 4756 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4757 // mono or stereo 4758 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4759 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4760 // hardware sample rate 4761 (sampleRate == mSampleRate) && 4762 // record thread has an associated fast recorder 4763 hasFastRecorder() 4764 // FIXME test that RecordThread for this fast track has a capable output HAL 4765 // FIXME add a permission test also? 4766 ) { 4767 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4768 if (frameCount == 0) { 4769 frameCount = mFrameCount * kFastTrackMultiplier; 4770 } 4771 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4772 frameCount, mFrameCount); 4773 } else { 4774 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4775 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4776 "hasFastRecorder=%d tid=%d", 4777 frameCount, mFrameCount, format, 4778 audio_is_linear_pcm(format), 4779 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4780 *flags &= ~IAudioFlinger::TRACK_FAST; 4781 // For compatibility with AudioRecord calculation, buffer depth is forced 4782 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4783 // This is probably too conservative, but legacy application code may depend on it. 4784 // If you change this calculation, also review the start threshold which is related. 4785 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4786 size_t mNormalFrameCount = 2048; // FIXME 4787 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4788 if (minBufCount < 2) { 4789 minBufCount = 2; 4790 } 4791 size_t minFrameCount = mNormalFrameCount * minBufCount; 4792 if (frameCount < minFrameCount) { 4793 frameCount = minFrameCount; 4794 } 4795 } 4796 } 4797 4798 // FIXME use flags and tid similar to createTrack_l() 4799 4800 { // scope for mLock 4801 Mutex::Autolock _l(mLock); 4802 4803 track = new RecordTrack(this, client, sampleRate, 4804 format, channelMask, frameCount, sessionId, uid); 4805 4806 lStatus = track->initCheck(); 4807 if (lStatus != NO_ERROR) { 4808 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4809 track.clear(); 4810 goto Exit; 4811 } 4812 mTracks.add(track); 4813 4814 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4815 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4816 mAudioFlinger->btNrecIsOff(); 4817 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4818 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4819 4820 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4821 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4822 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4823 // so ask activity manager to do this on our behalf 4824 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4825 } 4826 } 4827 lStatus = NO_ERROR; 4828 4829Exit: 4830 *status = lStatus; 4831 return track; 4832} 4833 4834status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4835 AudioSystem::sync_event_t event, 4836 int triggerSession) 4837{ 4838 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4839 sp<ThreadBase> strongMe = this; 4840 status_t status = NO_ERROR; 4841 4842 if (event == AudioSystem::SYNC_EVENT_NONE) { 4843 clearSyncStartEvent(); 4844 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4845 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4846 triggerSession, 4847 recordTrack->sessionId(), 4848 syncStartEventCallback, 4849 this); 4850 // Sync event can be cancelled by the trigger session if the track is not in a 4851 // compatible state in which case we start record immediately 4852 if (mSyncStartEvent->isCancelled()) { 4853 clearSyncStartEvent(); 4854 } else { 4855 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4856 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4857 } 4858 } 4859 4860 { 4861 // This section is a rendezvous between binder thread executing start() and RecordThread 4862 AutoMutex lock(mLock); 4863 if (mActiveTrack != 0) { 4864 if (recordTrack != mActiveTrack.get()) { 4865 status = -EBUSY; 4866 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4867 mActiveTrack->mState = TrackBase::ACTIVE; 4868 } 4869 return status; 4870 } 4871 4872 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4873 recordTrack->mState = TrackBase::IDLE; 4874 mActiveTrack = recordTrack; 4875 mLock.unlock(); 4876 status_t status = AudioSystem::startInput(mId); 4877 mLock.lock(); 4878 // FIXME should verify that mActiveTrack is still == recordTrack 4879 if (status != NO_ERROR) { 4880 mActiveTrack.clear(); 4881 clearSyncStartEvent(); 4882 return status; 4883 } 4884 // FIXME LEGACY 4885 mRsmpInIndex = mFrameCount; 4886 mRsmpInFront = 0; 4887 mRsmpInRear = 0; 4888 mRsmpInUnrel = 0; 4889 mBytesRead = 0; 4890 if (mResampler != NULL) { 4891 mResampler->reset(); 4892 } 4893 // FIXME hijacking a playback track state name which was intended for start after pause; 4894 // here 'STARTING_2' would be more accurate 4895 mActiveTrack->mState = TrackBase::RESUMING; 4896 // signal thread to start 4897 ALOGV("Signal record thread"); 4898 mWaitWorkCV.broadcast(); 4899 // do not wait for mStartStopCond if exiting 4900 if (exitPending()) { 4901 mActiveTrack.clear(); 4902 status = INVALID_OPERATION; 4903 goto startError; 4904 } 4905 // FIXME incorrect usage of wait: no explicit predicate or loop 4906 mStartStopCond.wait(mLock); 4907 if (mActiveTrack == 0) { 4908 ALOGV("Record failed to start"); 4909 status = BAD_VALUE; 4910 goto startError; 4911 } 4912 ALOGV("Record started OK"); 4913 return status; 4914 } 4915 4916startError: 4917 AudioSystem::stopInput(mId); 4918 clearSyncStartEvent(); 4919 return status; 4920} 4921 4922void AudioFlinger::RecordThread::clearSyncStartEvent() 4923{ 4924 if (mSyncStartEvent != 0) { 4925 mSyncStartEvent->cancel(); 4926 } 4927 mSyncStartEvent.clear(); 4928 mFramestoDrop = 0; 4929} 4930 4931void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4932{ 4933 sp<SyncEvent> strongEvent = event.promote(); 4934 4935 if (strongEvent != 0) { 4936 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4937 me->handleSyncStartEvent(strongEvent); 4938 } 4939} 4940 4941void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4942{ 4943 if (event == mSyncStartEvent) { 4944 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4945 // from audio HAL 4946 mFramestoDrop = mFrameCount * 2; 4947 } 4948} 4949 4950bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4951 ALOGV("RecordThread::stop"); 4952 AutoMutex _l(mLock); 4953 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4954 return false; 4955 } 4956 // note that threadLoop may still be processing the track at this point [without lock] 4957 recordTrack->mState = TrackBase::PAUSING; 4958 // do not wait for mStartStopCond if exiting 4959 if (exitPending()) { 4960 return true; 4961 } 4962 // FIXME incorrect usage of wait: no explicit predicate or loop 4963 mStartStopCond.wait(mLock); 4964 // if we have been restarted, recordTrack == mActiveTrack.get() here 4965 if (exitPending() || recordTrack != mActiveTrack.get()) { 4966 ALOGV("Record stopped OK"); 4967 return true; 4968 } 4969 return false; 4970} 4971 4972bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4973{ 4974 return false; 4975} 4976 4977status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4978{ 4979#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4980 if (!isValidSyncEvent(event)) { 4981 return BAD_VALUE; 4982 } 4983 4984 int eventSession = event->triggerSession(); 4985 status_t ret = NAME_NOT_FOUND; 4986 4987 Mutex::Autolock _l(mLock); 4988 4989 for (size_t i = 0; i < mTracks.size(); i++) { 4990 sp<RecordTrack> track = mTracks[i]; 4991 if (eventSession == track->sessionId()) { 4992 (void) track->setSyncEvent(event); 4993 ret = NO_ERROR; 4994 } 4995 } 4996 return ret; 4997#else 4998 return BAD_VALUE; 4999#endif 5000} 5001 5002// destroyTrack_l() must be called with ThreadBase::mLock held 5003void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5004{ 5005 track->terminate(); 5006 track->mState = TrackBase::STOPPED; 5007 // active tracks are removed by threadLoop() 5008 if (mActiveTrack != track) { 5009 removeTrack_l(track); 5010 } 5011} 5012 5013void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5014{ 5015 mTracks.remove(track); 5016 // need anything related to effects here? 5017} 5018 5019void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5020{ 5021 dumpInternals(fd, args); 5022 dumpTracks(fd, args); 5023 dumpEffectChains(fd, args); 5024} 5025 5026void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5027{ 5028 const size_t SIZE = 256; 5029 char buffer[SIZE]; 5030 String8 result; 5031 5032 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5033 result.append(buffer); 5034 5035 if (mActiveTrack != 0) { 5036 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5037 result.append(buffer); 5038 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5039 result.append(buffer); 5040 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5041 result.append(buffer); 5042 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5043 result.append(buffer); 5044 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5045 result.append(buffer); 5046 } else { 5047 result.append("No active record client\n"); 5048 } 5049 5050 write(fd, result.string(), result.size()); 5051 5052 dumpBase(fd, args); 5053} 5054 5055void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 5056{ 5057 const size_t SIZE = 256; 5058 char buffer[SIZE]; 5059 String8 result; 5060 5061 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5062 result.append(buffer); 5063 RecordTrack::appendDumpHeader(result); 5064 for (size_t i = 0; i < mTracks.size(); ++i) { 5065 sp<RecordTrack> track = mTracks[i]; 5066 if (track != 0) { 5067 track->dump(buffer, SIZE); 5068 result.append(buffer); 5069 } 5070 } 5071 5072 if (mActiveTrack != 0) { 5073 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5074 result.append(buffer); 5075 RecordTrack::appendDumpHeader(result); 5076 mActiveTrack->dump(buffer, SIZE); 5077 result.append(buffer); 5078 5079 } 5080 write(fd, result.string(), result.size()); 5081} 5082 5083// AudioBufferProvider interface 5084status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5085{ 5086 int32_t rear = mRsmpInRear; 5087 int32_t front = mRsmpInFront; 5088 ssize_t filled = rear - front; 5089 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5090 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5091 front &= mRsmpInFramesP2 - 1; 5092 size_t part1 = mRsmpInFramesP2 - front; 5093 if (part1 > (size_t) filled) { 5094 part1 = filled; 5095 } 5096 size_t ask = buffer->frameCount; 5097 ALOG_ASSERT(ask > 0); 5098 if (part1 > ask) { 5099 part1 = ask; 5100 } 5101 if (part1 == 0) { 5102 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5103 ALOGE("RecordThread::getNextBuffer() starved"); 5104 buffer->raw = NULL; 5105 buffer->frameCount = 0; 5106 mRsmpInUnrel = 0; 5107 return NOT_ENOUGH_DATA; 5108 } 5109 5110 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5111 buffer->frameCount = part1; 5112 mRsmpInUnrel = part1; 5113 return NO_ERROR; 5114} 5115 5116// AudioBufferProvider interface 5117void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5118{ 5119 size_t stepCount = buffer->frameCount; 5120 if (stepCount == 0) { 5121 return; 5122 } 5123 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5124 mRsmpInUnrel -= stepCount; 5125 mRsmpInFront += stepCount; 5126 buffer->raw = NULL; 5127 buffer->frameCount = 0; 5128} 5129 5130bool AudioFlinger::RecordThread::checkForNewParameters_l() 5131{ 5132 bool reconfig = false; 5133 5134 while (!mNewParameters.isEmpty()) { 5135 status_t status = NO_ERROR; 5136 String8 keyValuePair = mNewParameters[0]; 5137 AudioParameter param = AudioParameter(keyValuePair); 5138 int value; 5139 audio_format_t reqFormat = mFormat; 5140 uint32_t reqSamplingRate = mReqSampleRate; 5141 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5142 5143 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5144 reqSamplingRate = value; 5145 reconfig = true; 5146 } 5147 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5148 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5149 status = BAD_VALUE; 5150 } else { 5151 reqFormat = (audio_format_t) value; 5152 reconfig = true; 5153 } 5154 } 5155 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5156 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5157 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5158 status = BAD_VALUE; 5159 } else { 5160 reqChannelMask = mask; 5161 reconfig = true; 5162 } 5163 } 5164 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5165 // do not accept frame count changes if tracks are open as the track buffer 5166 // size depends on frame count and correct behavior would not be guaranteed 5167 // if frame count is changed after track creation 5168 if (mActiveTrack != 0) { 5169 status = INVALID_OPERATION; 5170 } else { 5171 reconfig = true; 5172 } 5173 } 5174 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5175 // forward device change to effects that have requested to be 5176 // aware of attached audio device. 5177 for (size_t i = 0; i < mEffectChains.size(); i++) { 5178 mEffectChains[i]->setDevice_l(value); 5179 } 5180 5181 // store input device and output device but do not forward output device to audio HAL. 5182 // Note that status is ignored by the caller for output device 5183 // (see AudioFlinger::setParameters() 5184 if (audio_is_output_devices(value)) { 5185 mOutDevice = value; 5186 status = BAD_VALUE; 5187 } else { 5188 mInDevice = value; 5189 // disable AEC and NS if the device is a BT SCO headset supporting those 5190 // pre processings 5191 if (mTracks.size() > 0) { 5192 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5193 mAudioFlinger->btNrecIsOff(); 5194 for (size_t i = 0; i < mTracks.size(); i++) { 5195 sp<RecordTrack> track = mTracks[i]; 5196 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5197 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5198 } 5199 } 5200 } 5201 } 5202 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5203 mAudioSource != (audio_source_t)value) { 5204 // forward device change to effects that have requested to be 5205 // aware of attached audio device. 5206 for (size_t i = 0; i < mEffectChains.size(); i++) { 5207 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5208 } 5209 mAudioSource = (audio_source_t)value; 5210 } 5211 5212 if (status == NO_ERROR) { 5213 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5214 keyValuePair.string()); 5215 if (status == INVALID_OPERATION) { 5216 inputStandBy(); 5217 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5218 keyValuePair.string()); 5219 } 5220 if (reconfig) { 5221 if (status == BAD_VALUE && 5222 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5223 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5224 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5225 <= (2 * reqSamplingRate)) && 5226 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5227 <= FCC_2 && 5228 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5229 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5230 status = NO_ERROR; 5231 } 5232 if (status == NO_ERROR) { 5233 readInputParameters(); 5234 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5235 } 5236 } 5237 } 5238 5239 mNewParameters.removeAt(0); 5240 5241 mParamStatus = status; 5242 mParamCond.signal(); 5243 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5244 // already timed out waiting for the status and will never signal the condition. 5245 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5246 } 5247 return reconfig; 5248} 5249 5250String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5251{ 5252 Mutex::Autolock _l(mLock); 5253 if (initCheck() != NO_ERROR) { 5254 return String8(); 5255 } 5256 5257 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5258 const String8 out_s8(s); 5259 free(s); 5260 return out_s8; 5261} 5262 5263void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5264 AudioSystem::OutputDescriptor desc; 5265 const void *param2 = NULL; 5266 5267 switch (event) { 5268 case AudioSystem::INPUT_OPENED: 5269 case AudioSystem::INPUT_CONFIG_CHANGED: 5270 desc.channelMask = mChannelMask; 5271 desc.samplingRate = mSampleRate; 5272 desc.format = mFormat; 5273 desc.frameCount = mFrameCount; 5274 desc.latency = 0; 5275 param2 = &desc; 5276 break; 5277 5278 case AudioSystem::INPUT_CLOSED: 5279 default: 5280 break; 5281 } 5282 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5283} 5284 5285void AudioFlinger::RecordThread::readInputParameters() 5286{ 5287 delete[] mRsmpInBuffer; 5288 // mRsmpInBuffer is always assigned a new[] below 5289 delete[] mRsmpOutBuffer; 5290 mRsmpOutBuffer = NULL; 5291 delete mResampler; 5292 mResampler = NULL; 5293 5294 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5295 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5296 mChannelCount = popcount(mChannelMask); 5297 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5298 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5299 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5300 } 5301 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5302 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5303 mFrameCount = mBufferSize / mFrameSize; 5304 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5305 // 1 full output buffer, regardless of the alignment of the available input. 5306 mRsmpInFrames = mFrameCount * 3; 5307 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5308 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5309 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5310 mRsmpInFront = 0; 5311 mRsmpInRear = 0; 5312 mRsmpInUnrel = 0; 5313 5314 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5315 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5316 mResampler->setSampleRate(mSampleRate); 5317 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5318 // resampler always outputs stereo 5319 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5320 } 5321 mRsmpInIndex = mFrameCount; 5322} 5323 5324unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5325{ 5326 Mutex::Autolock _l(mLock); 5327 if (initCheck() != NO_ERROR) { 5328 return 0; 5329 } 5330 5331 return mInput->stream->get_input_frames_lost(mInput->stream); 5332} 5333 5334uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5335{ 5336 Mutex::Autolock _l(mLock); 5337 uint32_t result = 0; 5338 if (getEffectChain_l(sessionId) != 0) { 5339 result = EFFECT_SESSION; 5340 } 5341 5342 for (size_t i = 0; i < mTracks.size(); ++i) { 5343 if (sessionId == mTracks[i]->sessionId()) { 5344 result |= TRACK_SESSION; 5345 break; 5346 } 5347 } 5348 5349 return result; 5350} 5351 5352KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5353{ 5354 KeyedVector<int, bool> ids; 5355 Mutex::Autolock _l(mLock); 5356 for (size_t j = 0; j < mTracks.size(); ++j) { 5357 sp<RecordThread::RecordTrack> track = mTracks[j]; 5358 int sessionId = track->sessionId(); 5359 if (ids.indexOfKey(sessionId) < 0) { 5360 ids.add(sessionId, true); 5361 } 5362 } 5363 return ids; 5364} 5365 5366AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5367{ 5368 Mutex::Autolock _l(mLock); 5369 AudioStreamIn *input = mInput; 5370 mInput = NULL; 5371 return input; 5372} 5373 5374// this method must always be called either with ThreadBase mLock held or inside the thread loop 5375audio_stream_t* AudioFlinger::RecordThread::stream() const 5376{ 5377 if (mInput == NULL) { 5378 return NULL; 5379 } 5380 return &mInput->stream->common; 5381} 5382 5383status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5384{ 5385 // only one chain per input thread 5386 if (mEffectChains.size() != 0) { 5387 return INVALID_OPERATION; 5388 } 5389 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5390 5391 chain->setInBuffer(NULL); 5392 chain->setOutBuffer(NULL); 5393 5394 checkSuspendOnAddEffectChain_l(chain); 5395 5396 mEffectChains.add(chain); 5397 5398 return NO_ERROR; 5399} 5400 5401size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5402{ 5403 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5404 ALOGW_IF(mEffectChains.size() != 1, 5405 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5406 chain.get(), mEffectChains.size(), this); 5407 if (mEffectChains.size() == 1) { 5408 mEffectChains.removeAt(0); 5409 } 5410 return 0; 5411} 5412 5413}; // namespace android 5414